| Rfc | 7826 |
|---|
| Title | Real-Time Streaming Protocol Version 2.0 |
|---|
| Author | H. Schulzrinne, A. Rao, R.
Lanphier, M. Westerlund, M. Stiemerling, Ed. |
|---|
| Date | December 2016 |
|---|
| Format: | TXT, HTML |
|---|
| Obsoletes | RFC2326 |
|---|
| Status: | PROPOSED STANDARD |
|---|
|
Internet Engineering Task Force (IETF) H. Schulzrinne
Request for Comments: 7826 Columbia University
Obsoletes: 2326 A. Rao
Category: Standards Track Cisco
ISSN: 2070-1721 R. Lanphier
M. Westerlund
Ericsson
M. Stiemerling, Ed.
University of Applied Sciences Darmstadt
December 2016
Real-Time Streaming Protocol Version 2.0
Abstract
This memorandum defines the Real-Time Streaming Protocol (RTSP)
version 2.0, which obsoletes RTSP version 1.0 defined in RFC 2326.
RTSP is an application-layer protocol for the setup and control of
the delivery of data with real-time properties. RTSP provides an
extensible framework to enable controlled, on-demand delivery of
real-time data, such as audio and video. Sources of data can include
both live data feeds and stored clips. This protocol is intended to
control multiple data delivery sessions; provide a means for choosing
delivery channels such as UDP, multicast UDP, and TCP; and provide a
means for choosing delivery mechanisms based upon RTP (RFC 3550).
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc7826.
Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the
document authors. All rights reserved.
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Table of Contents
1. Introduction ...................................................10
2. Protocol Overview ..............................................11
2.1. Presentation Description ..................................12
2.2. Session Establishment .....................................12
2.3. Media Delivery Control ....................................14
2.4. Session Parameter Manipulations ...........................15
2.5. Media Delivery ............................................16
2.5.1. Media Delivery Manipulations .......................16
2.6. Session Maintenance and Termination .......................19
2.7. Extending RTSP ............................................20
3. Document Conventions ...........................................21
3.1. Notational Conventions ....................................21
3.2. Terminology ...............................................21
4. Protocol Parameters ............................................25
4.1. RTSP Version ..............................................25
4.2. RTSP IRI and URI ..........................................25
4.3. Session Identifiers .......................................28
4.4. Media-Time Formats ........................................28
4.4.1. SMPTE-Relative Timestamps ..........................28
4.4.2. Normal Play Time ...................................29
4.4.3. Absolute Time ......................................30
4.5. Feature Tags ..............................................31
4.6. Message Body Tags .........................................32
4.7. Media Properties ..........................................32
4.7.1. Random Access and Seeking ..........................33
4.7.2. Retention ..........................................34
4.7.3. Content Modifications ..............................34
4.7.4. Supported Scale Factors ............................34
4.7.5. Mapping to the Attributes ..........................35
5. RTSP Message ...................................................35
5.1. Message Types .............................................36
5.2. Message Headers ...........................................36
5.3. Message Body ..............................................37
5.4. Message Length ............................................37
6. General-Header Fields ..........................................37
7. Request ........................................................39
7.1. Request Line ..............................................40
7.2. Request-Header Fields .....................................42
8. Response .......................................................43
8.1. Status-Line ...............................................43
8.1.1. Status Code and Reason Phrase ......................43
8.2. Response Headers ..........................................47
9. Message Body ...................................................47
9.1. Message Body Header Fields ................................48
9.2. Message Body ..............................................49
9.3. Message Body Format Negotiation ...........................49
10. Connections ...................................................50
10.1. Reliability and Acknowledgements .........................50
10.2. Using Connections ........................................51
10.3. Closing Connections ......................................54
10.4. Timing Out Connections and RTSP Messages .................56
10.5. Showing Liveness .........................................57
10.6. Use of IPv6 ..............................................58
10.7. Overload Control .........................................58
11. Capability Handling ...........................................60
11.1. Feature Tag: play.basic ..................................62
12. Pipelining Support ............................................62
13. Method Definitions ............................................63
13.1. OPTIONS ..................................................65
13.2. DESCRIBE .................................................66
13.3. SETUP ....................................................68
13.3.1. Changing Transport Parameters .....................71
13.4. PLAY .....................................................72
13.4.1. General Usage .....................................72
13.4.2. Aggregated Sessions ...............................77
13.4.3. Updating Current PLAY Requests ....................78
13.4.4. Playing On-Demand Media ...........................81
13.4.5. Playing Dynamic On-Demand Media ...................81
13.4.6. Playing Live Media ................................81
13.4.7. Playing Live with Recording .......................82
13.4.8. Playing Live with Time-Shift ......................83
13.5. PLAY_NOTIFY ..............................................83
13.5.1. End-of-Stream .....................................84
13.5.2. Media-Properties-Update ...........................86
13.5.3. Scale-Change ......................................87
13.6. PAUSE ....................................................89
13.7. TEARDOWN .................................................92
13.7.1. Client to Server ..................................92
13.7.2. Server to Client ..................................93
13.8. GET_PARAMETER ............................................94
13.9. SET_PARAMETER ............................................96
13.10. REDIRECT ................................................98
14. Embedded (Interleaved) Binary Data ...........................101
15. Proxies ......................................................103
15.1. Proxies and Protocol Extensions .........................104
15.2. Multiplexing and Demultiplexing of Messages .............105
16. Caching ......................................................106
16.1. Validation Model ........................................107
16.1.1. Last-Modified Dates ..............................108
16.1.2. Message Body Tag Cache Validators ................108
16.1.3. Weak and Strong Validators .......................108
16.1.4. Rules for When to Use Message Body Tags
and Last-Modified Dates ..........................110
16.1.5. Non-validating Conditionals ......................112
16.2. Invalidation after Updates or Deletions .................112
17. Status Code Definitions ......................................113
17.1. Informational 1xx .......................................113
17.1.1. 100 Continue .....................................113
17.2. Success 2xx .............................................113
17.2.1. 200 OK ...........................................113
17.3. Redirection 3xx .........................................113
17.3.1. 300 ..............................................114
17.3.2. 301 Moved Permanently ............................114
17.3.3. 302 Found ........................................114
17.3.4. 303 See Other ....................................115
17.3.5. 304 Not Modified .................................115
17.3.6. 305 Use Proxy ....................................115
17.4. Client Error 4xx ........................................116
17.4.1. 400 Bad Request ..................................116
17.4.2. 401 Unauthorized .................................116
17.4.3. 402 Payment Required .............................116
17.4.4. 403 Forbidden ....................................116
17.4.5. 404 Not Found ....................................116
18.11. Cache-Control ..........................................140
18.12. Connection .............................................143
18.13. Connection-Credentials .................................143
18.14. Content-Base ...........................................144
18.15. Content-Encoding .......................................145
18.16. Content-Language .......................................145
18.17. Content-Length .........................................146
18.18. Content-Location .......................................146
18.19. Content-Type ...........................................148
18.20. CSeq ...................................................148
18.21. Date ...................................................150
18.22. Expires ................................................151
18.23. From ...................................................151
18.24. If-Match ...............................................152
18.25. If-Modified-Since ......................................152
18.26. If-None-Match ..........................................153
18.27. Last-Modified ..........................................154
18.28. Location ...............................................154
18.29. Media-Properties .......................................154
18.30. Media-Range ............................................156
18.31. MTag ...................................................157
18.32. Notify-Reason ..........................................158
18.33. Pipelined-Requests .....................................158
18.34. Proxy-Authenticate .....................................159
18.35. Proxy-Authentication-Info ..............................159
18.36. Proxy-Authorization ....................................159
18.37. Proxy-Require ..........................................160
18.38. Proxy-Supported ........................................160
18.39. Public .................................................161
18.40. Range ..................................................162
18.41. Referrer ...............................................164
18.42. Request-Status .........................................164
18.43. Require ................................................165
18.44. Retry-After ............................................166
18.45. RTP-Info ...............................................167
18.46. Scale ..................................................169
18.47. Seek-Style .............................................170
18.48. Server .................................................171
18.49. Session ................................................172
18.50. Speed ..................................................173
18.51. Supported ..............................................174
18.52. Terminate-Reason .......................................175
18.53. Timestamp ..............................................175
18.54. Transport ..............................................176
18.55. Unsupported ............................................183
18.56. User-Agent .............................................184
18.57. Via ....................................................184
18.58. WWW-Authenticate .......................................185
19. Security Framework ...........................................185
19.1. RTSP and HTTP Authentication ............................185
19.1.1. Digest Authentication ............................186
19.2. RTSP over TLS ...........................................187
19.3. Security and Proxies ....................................188
19.3.1. Accept-Credentials ...............................189
19.3.2. User-Approved TLS Procedure ......................190
20. Syntax .......................................................192
20.1. Base Syntax .............................................193
20.2. RTSP Protocol Definition ................................195
20.2.1. Generic Protocol Elements ........................195
20.2.2. Message Syntax ...................................198
20.2.3. Header Syntax ....................................201
20.3. SDP Extension Syntax ....................................209
21. Security Considerations ......................................209
21.1. Signaling Protocol Threats ..............................210
21.2. Media Stream Delivery Threats ...........................213
21.2.1. Remote DoS Attack ................................215
21.2.2. RTP Security Analysis ............................216
22. IANA Considerations ..........................................217
22.1. Feature Tags ............................................218
22.1.1. Description ......................................218
22.1.2. Registering New Feature Tags with IANA ...........218
22.1.3. Registered Entries ...............................219
22.2. RTSP Methods ............................................219
22.2.1. Description ......................................219
22.2.2. Registering New Methods with IANA ................219
22.2.3. Registered Entries ...............................220
22.3. RTSP Status Codes .......................................220
22.3.1. Description ......................................220
22.3.2. Registering New Status Codes with IANA ...........220
22.3.3. Registered Entries ...............................221
22.4. RTSP Headers ............................................221
22.4.1. Description ......................................221
22.4.2. Registering New Headers with IANA ................221
22.4.3. Registered Entries ...............................222
22.5. Accept-Credentials ......................................223
22.5.1. Accept-Credentials Policies ......................223
22.5.2. Accept-Credentials Hash Algorithms ...............224
22.6. Cache-Control Cache Directive Extensions ................224
22.7. Media Properties ........................................225
22.7.1. Description ......................................225
22.7.2. Registration Rules ...............................226
22.7.3. Registered Values ................................226
22.8. Notify-Reason Values ....................................226
22.8.1. Description ......................................226
22.8.2. Registration Rules ...............................226
22.8.3. Registered Values ................................227
22.9. Range Header Formats ....................................227
22.9.1. Description ......................................227
22.9.2. Registration Rules ...............................227
22.9.3. Registered Values ................................228
22.10. Terminate-Reason Header ................................228
22.10.1. Redirect Reasons ................................228
22.10.2. Terminate-Reason Header Parameters ..............229
22.11. RTP-Info Header Parameters .............................229
22.11.1. Description .....................................229
22.11.2. Registration Rules ..............................229
22.11.3. Registered Values ...............................230
22.12. Seek-Style Policies ....................................230
22.12.1. Description .....................................230
22.12.2. Registration Rules ..............................230
22.12.3. Registered Values ...............................230
22.13. Transport Header Registries ............................231
22.13.1. Transport Protocol Identifier ...................231
22.13.2. Transport Modes .................................233
22.13.3. Transport Parameters ............................233
22.14. URI Schemes ............................................234
22.14.1. The "rtsp" URI Scheme ...........................234
22.14.2. The "rtsps" URI Scheme ..........................235
22.14.3. The "rtspu" URI Scheme ..........................237
22.15. SDP Attributes .........................................238
22.16. Media Type Registration for text/parameters ............238
23. References ...................................................240
23.1. Normative References ....................................240
23.2. Informative References ..................................245
Appendix A. Examples .............................................248
A.1. Media on Demand (Unicast) ................................248
A.2. Media on Demand Using Pipelining .........................251
A.3. Secured Media Session for On-Demand Content ..............254
A.4. Media on Demand (Unicast) ................................257
A.5. Single-Stream Container Files ............................260
A.6. Live Media Presentation Using Multicast ..................263
A.7. Capability Negotiation ...................................264
Appendix B. RTSP Protocol State Machine ..........................265
B.1. States ...................................................266
B.2. State Variables ..........................................266
B.3. Abbreviations ............................................266
B.4. State Tables .............................................267
Appendix C. Media-Transport Alternatives .........................272
C.1. RTP ......................................................272
C.1.1. AVP ..................................................272
C.1.2. AVP/UDP ..............................................273
C.1.3. AVPF/UDP .............................................274
C.1.4. SAVP/UDP .............................................275
C.1.5. SAVPF/UDP ............................................277
C.1.6. RTCP Usage with RTSP .................................278
C.2. RTP over TCP .............................................279
C.2.1. Interleaved RTP over TCP .............................280
C.2.2. RTP over Independent TCP .............................280
C.3. Handling Media-Clock Time Jumps in the RTP Media Layer ...284
C.4. Handling RTP Timestamps after PAUSE ......................287
C.5. RTSP/RTP Integration ....................................290
C.6. Scaling with RTP .........................................290
C.7. Maintaining NPT Synchronization with RTP Timestamps ......290
C.8. Continuous Audio .........................................290
C.9. Multiple Sources in an RTP Session .......................290
C.10. Usage of SSRCs and the RTCP BYE Message during an RTSP
Session .................................................290
C.11. Future Additions ........................................291
Appendix D. Use of SDP for RTSP Session Descriptions .............292
D.1. Definitions .............................................292
D.1.1. Control URI ..........................................292
D.1.2. Media Streams ........................................294
D.1.3. Payload Type(s) ......................................294
D.1.4. Format-Specific Parameters ...........................294
D.1.5. Directionality of Media Stream .......................295
D.1.6. Range of Presentation ................................295
D.1.7. Time of Availability .................................296
D.1.8. Connection Information ...............................297
D.1.9. Message Body Tag .....................................297
D.2. Aggregate Control Not Available ..........................298
D.3. Aggregate Control Available ..............................298
D.4. Grouping of Media Lines in SDP ...........................299
D.5. RTSP External SDP Delivery ...............................300
Appendix E. RTSP Use Cases .......................................300
E.1. On-Demand Playback of Stored Content .....................300
E.2. Unicast Distribution of Live Content .....................302
E.3. On-Demand Playback Using Multicast .......................303
E.4. Inviting an RTSP Server into a Conference ................303
E.5. Live Content Using Multicast .............................304
Appendix F. Text Format for Parameters ...........................305
Appendix G. Requirements for Unreliable Transport of RTSP ........305
Appendix H. Backwards-Compatibility Considerations ...............306
H.1. Play Request in Play State ...............................307
H.2. Using Persistent Connections .............................307
Appendix I. Changes ..............................................307
I.1. Brief Overview ...........................................308
I.2. Detailed List of Changes .................................309
Acknowledgements .................................................316
Contributors ....................................................317
Authors' Addresses ...............................................318
1. Introduction
This memo defines version 2.0 of the Real-Time Streaming Protocol
(RTSP 2.0). RTSP 2.0 is an application-layer protocol for the setup
and control over the delivery of data with real-time properties,
typically streaming media. Streaming media is, for instance, video
on demand or audio live streaming. Put simply, RTSP acts as a
"network remote control" for multimedia servers.
The protocol operates between RTSP 2.0 clients and servers, but it
also supports the use of proxies placed between clients and servers.
Clients can request information about streaming media from servers by
asking for a description of the media or use media description
provided externally. The media delivery protocol is used to
establish the media streams described by the media description.
Clients can then request to play out the media, pause it, or stop it
completely. The requested media can consist of multiple audio and
video streams that are delivered as time-synchronized streams from
servers to clients.
RTSP 2.0 is a replacement of RTSP 1.0 [RFC2326] and this document
obsoletes that specification. This protocol is based on RTSP 1.0 but
is not backwards compatible other than in the basic version
negotiation mechanism. The changes between the two documents are
listed in Appendix I. There are many reasons why RTSP 2.0 can't be
backwards compatible with RTSP 1.0; some of the main ones are as
follows:
o Most headers that needed to be extensible did not define the
allowed syntax, preventing safe deployment of extensions;
o the changed behavior of the PLAY method when received in Play
state;
o the changed behavior of the extensibility model and its mechanism;
and
o the change of syntax for some headers.
There are so many small updates that changing versions became
necessary to enable clarification and consistent behavior. Anyone
implementing RTSP for a new use case in which they have not installed
RTSP 1.0 should only implement RTSP 2.0 to avoid having to deal with
RTSP 1.0 inconsistencies.
This document is structured as follows. It begins with an overview
of the protocol operations and its functions in an informal way.
Then, a set of definitions of terms used and document conventions is
introduced. These are followed by the actual RTSP 2.0 core protocol
specification. The appendices describe and define some
functionalities that are not part of the core RTSP specification, but
which are still important to enable some usages. Among them, the RTP
usage is defined in Appendix C, the Session Description Protocol
(SDP) usage with RTSP is defined in Appendix D, and the "text/
parameters" file format Appendix F, are three normative specification
appendices. Other appendices include a number of informational parts
discussing the changes, use cases, different considerations or
motivations.
2. Protocol Overview
This section provides an informative overview of the different
mechanisms in the RTSP 2.0 protocol to give the reader a high-level
understanding before getting into all the specific details. In case
of conflict with this description and the later sections, the later
sections take precedence. For more information about use cases
considered for RTSP, see Appendix E.
RTSP 2.0 is a bidirectional request and response protocol that first
establishes a context including content resources (the media) and
then controls the delivery of these content resources from the
provider to the consumer. RTSP has three fundamental parts: Session
Establishment, Media Delivery Control, and an extensibility model
described below. The protocol is based on some assumptions about
existing functionality to provide a complete solution for client-
controlled real-time media delivery.
RTSP uses text-based messages, requests and responses, that may
contain a binary message body. An RTSP request starts with a method
line that identifies the method, the protocol, and version and the
resource on which to act. The resource is identified by a URI and
the hostname part of the URI is used by RTSP client to resolve the
IPv4 or IPv6 address of the RTSP server. Following the method line
are a number of RTSP headers. These lines are ended by two
consecutive carriage return line feed (CRLF) character pairs. The
message body, if present, follows the two CRLF character pairs, and
the body's length is described by a message header. RTSP responses
are similar, but they start with a response line with the protocol
and version followed by a status code and a reason phrase. RTSP
messages are sent over a reliable transport protocol between the
client and server. RTSP 2.0 requires clients and servers to
implement TCP and TLS over TCP as mandatory transports for RTSP
messages.
2.1. Presentation Description
RTSP exists to provide access to multimedia presentations and content
but tries to be agnostic about the media type or the actual media
delivery protocol that is used. To enable a client to implement a
complete system, an RTSP-external mechanism for describing the
presentation and the delivery protocol(s) is used. RTSP assumes that
this description is either delivered completely out of band or as a
data object in the response to a client's request using the DESCRIBE
method (Section 13.2).
Parameters that commonly have to be included in the presentation
description are the following:
o The number of media streams;
o the resource identifier for each media stream/resource that is to
be controlled by RTSP;
o the protocol that will be used to deliver each media stream;
o the transport protocol parameters that are not negotiated or vary
with each client;
o the media-encoding information enabling a client to correctly
decode the media upon reception; and
o an aggregate control resource identifier.
RTSP uses its own URI schemes ("rtsp" and "rtsps") to reference media
resources and aggregates under common control (see Section 4.2).
This specification describes in Appendix D how one uses SDP [RFC4566]
for describing the presentation.
2.2. Session Establishment
The RTSP client can request the establishment of an RTSP session
after having used the presentation description to determine which
media streams are available, which media delivery protocol is used,
and the resource identifiers of the media streams. The RTSP session
is a common context between the client and the server that consists
of one or more media resources that are to be under common media
delivery control.
The client creates an RTSP session by sending a request using the
SETUP method (Section 13.3) to the server. In the Transport header
(Section 18.54) of the SETUP request, the client also includes all
the transport parameters necessary to enable the media delivery
protocol to function. This includes parameters that are
preestablished by the presentation description but necessary for any
middlebox to correctly handle the media delivery protocols. The
Transport header in a request may contain multiple alternatives for
media delivery in a prioritized list, which the server can select
from. These alternatives are typically based on information in the
presentation description.
When receiving a SETUP request, the server determines if the media
resource is available and if one or more of the of the transport
parameter specifications are acceptable. If that is successful, an
RTSP session context is created and the relevant parameters and state
is stored. An identifier is created for the RTSP session and
included in the response in the Session header (Section 18.49). The
SETUP response includes a Transport header that specifies which of
the alternatives has been selected and relevant parameters.
A SETUP request that references an existing RTSP session but
identifies a new media resource is a request to add that media
resource under common control with the already-present media
resources in an aggregated session. A client can expect this to work
for all media resources under RTSP control within a multimedia
content container. However, a server will likely refuse to aggregate
resources from different content containers. Even if an RTSP session
contains only a single media stream, the RTSP session can be
referenced by the aggregate control URI.
To avoid an extra round trip in the session establishment of
aggregated RTSP sessions, RTSP 2.0 supports pipelined requests; i.e.,
the client can send multiple requests back-to-back without waiting
first for the completion of any of them. The client uses a client-
selected identifier in the Pipelined-Requests header (Section 18.33)
to instruct the server to bind multiple requests together as if they
included the session identifier.
The SETUP response also provides additional information about the
established sessions in a couple of different headers. The Media-
Properties header (Section 18.29) includes a number of properties
that apply for the aggregate that is valuable when doing media
delivery control and configuring user interface. The Accept-Ranges
header (Section 18.5) informs the client about range formats that the
server supports for these media resources. The Media-Range header
(Section 18.30) informs the client about the time range of the media
currently available.
2.3. Media Delivery Control
After having established an RTSP session, the client can start
controlling the media delivery. The basic operations are "begin
playback", using the PLAY method (Section 13.4) and "suspend (pause)
playback" by using the PAUSE method (Section 13.6). PLAY also allows
for choosing the starting media position from which the server should
deliver the media. The positioning is done by using the Range header
(Section 18.40) that supports several different time formats: Normal
Play Time (NPT) (Section 4.4.2), Society of Motion Picture and
Television Engineers (SMPTE) Timestamps (Section 4.4.1), and absolute
time (Section 4.4.3). The Range header also allows the client to
specify a position where delivery should end, thus allowing a
specific interval to be delivered.
The support for positioning/searching within media content depends on
the content's media properties. Content exists in a number of
different types, such as on-demand, live, and live with simultaneous
recording. Even within these categories, there are differences in
how the content is generated and distributed, which affect how it can
be accessed for playback. The properties applicable for the RTSP
session are provided by the server in the SETUP response using the
Media-Properties header (Section 18.29). These are expressed using
one or several independent attributes. A first attribute is Random-
Access, which indicates whether positioning is possible, and with
what granularity. Another aspect is whether the content will change
during the lifetime of the session. While on-demand content will be
provided in full from the beginning, a live stream being recorded
results in the length of the accessible content growing as the
session goes on. There also exists content that is dynamically built
by a protocol other than RTSP and, thus, also changes in steps during
the session, but maybe not continuously. Furthermore, when content
is recorded, there are cases where the complete content is not
maintained, but, for example, only the last hour. All of these
properties result in the need for mechanisms that will be discussed
below.
When the client accesses on-demand content that allows random access,
the client can issue the PLAY request for any point in the content
between the start and the end. The server will deliver media from
the closest random access point prior to the requested point and
indicate that in its PLAY response. If the client issues a PAUSE,
the delivery will be halted and the point at which the server stopped
will be reported back in the response. The client can later resume
by sending a PLAY request without a Range header. When the server is
about to complete the PLAY request by delivering the end of the
content or the requested range, the server will send a PLAY_NOTIFY
request (Section 13.5) indicating this.
When playing live content with no extra functions, such as recording,
the client will receive the live media from the server after having
sent a PLAY request. Seeking in such content is not possible as the
server does not store it, but only forwards it from the source of the
session. Thus, delivery continues until the client sends a PAUSE
request, tears down the session, or the content ends.
For live sessions that are being recorded, the client will need to
keep track of how the recording progresses. Upon session
establishment, the client will learn the current duration of the
recording from the Media-Range header. Because the recording is
ongoing, the content grows in direct relation to the time passed.
Therefore, each server's response to a PLAY request will contain the
current Media-Range header. The server should also regularly send
(approximately every 5 minutes) the current media range in a
PLAY_NOTIFY request (Section 13.5.2). If the live transmission ends,
the server must send a PLAY_NOTIFY request with the updated Media-
Properties indicating that the content stopped being a recorded live
session and instead became on-demand content; the request also
contains the final media range. While the live delivery continues,
the client can request to play the current live point by using the
NPT timescale symbol "now", or it can request a specific point in the
available content by an explicit range request for that point. If
the requested point is outside of the available interval, the server
will adjust the position to the closest available point, i.e., either
at the beginning or the end.
A special case of recording is that where the recording is not
retained longer than a specific time period; thus, as the live
delivery continues, the client can access any media within a moving
window that covers, for example, "now" to "now" minus 1 hour. A
client that pauses on a specific point within the content may not be
able to retrieve the content anymore. If the client waits too long
before resuming the pause point, the content may no longer be
available. In this case, the pause point will be adjusted to the
closest point in the available media.
2.4. Session Parameter Manipulations
A session may have additional state or functionality that affects how
the server or client treats the session or content, how it functions,
or feedback on how well the session works. Such extensions are not
defined in this specification, but they may be covered in various
extensions. RTSP has two methods for retrieving and setting
parameter values on either the client or the server: GET_PARAMETER
(Section 13.8) and SET_PARAMETER (Section 13.9). These methods carry
the parameters in a message body of the appropriate format. One can
also use headers to query state with the GET_PARAMETER method. As an
example, clients needing to know the current media range for a time-
progressing session can use the GET_PARAMETER method and include the
media range. Furthermore, synchronization information can be
requested by using a combination of RTP-Info (Section 18.45) and
Range (Section 18.40).
RTSP 2.0 does not have a strong mechanism for negotiating the headers
or parameters and their formats. However, responses will indicate
request-headers or parameters that are not supported. A priori
determination of what features are available needs to be done through
out-of-band mechanisms, like the session description, or through the
usage of feature tags (Section 4.5).
2.5. Media Delivery
This document specifies how media is delivered with RTP [RFC3550]
over UDP [RFC768], TCP [RFC793], or the RTSP connection. Additional
protocols may be specified in the future as needed.
The usage of RTP as a media delivery protocol requires some
additional information to function well. The PLAY response contains
information to enable reliable and timely delivery of how a client
should synchronize different sources in the different RTP sessions.
It also provides a mapping between RTP timestamps and the content-
time scale. When the server wants to notify the client about the
completion of the media delivery, it sends a PLAY_NOTIFY request to
the client. The PLAY_NOTIFY request includes information about the
stream end, including the last RTP sequence number for each stream,
thus enabling the client to empty the buffer smoothly.
2.5.1. Media Delivery Manipulations
The basic playback functionality of RTSP enables delivery of a range
of requested content to the client at the pace intended by the
content's creator. However, RTSP can also manipulate the delivery to
the client in two ways.
Scale: The ratio of media-content time delivered per unit of
playback time.
Speed: The ratio of playback time delivered per unit of wallclock
time.
Both affect the media delivery per time unit. However, they
manipulate two independent timescales and the effects are possible to
combine.
Scale (Section 18.46) is used for fast-forward or slow-motion control
as it changes the amount of content timescale that should be played
back per time unit. Scale > 1.0, means fast forward, e.g., scale =
2.0 results in that 2 seconds of content being played back every
second of playback. Scale = 1.0 is the default value that is used if
no scale is specified, i.e., playback at the content's original rate.
Scale values between 0 and 1.0 provide for slow motion. Scale can be
negative to allow for reverse playback in either regular pace
(scale = -1.0), fast backwards (scale < -1.0), or slow-motion
backwards (-1.0 < scale < 0). Scale = 0 would be equal to pause and
is not allowed.
In most cases, the realization of scale means server-side
manipulation of the media to ensure that the client can actually play
it back. The nature of these media manipulations and when they are
needed is highly media-type dependent. Let's consider two common
media types, audio and video.
It is very difficult to modify the playback rate of audio.
Typically, no more than a factor of two is possible while maintaining
intelligibility by changing the pitch and rate of speech. Music goes
out of tune if one tries to manipulate the playback rate by
resampling it. This is a well-known problem, and audio is commonly
muted or played back in short segments with skips to keep up with the
current playback point.
For video, it is possible to manipulate the frame rate, although the
rendering capabilities are often limited to certain frame rates.
Also, the allowed bitrates in decoding, the structure used in the
encoding, and the dependency between frames and other capabilities of
the rendering device limits the possible manipulations. Therefore,
the basic fast-forward capabilities often are implemented by
selecting certain subsets of frames.
Due to the media restrictions, the possible scale values are commonly
restricted to the set of realizable scale ratios. To enable the
clients to select from the possible scale values, RTSP can signal the
supported scale ratios for the content. To support aggregated or
dynamic content, where this may change during the ongoing session and
dependent on the location within the content, a mechanism for
updating the media properties and the scale factor currently in use,
exists.
Speed (Section 18.50) affects how much of the playback timeline is
delivered in a given wallclock period. The default is Speed = 1
which means to deliver at the same rate the media is consumed.
Speed > 1 means that the receiver will get content faster than it
regularly would consume it. Speed < 1 means that delivery is slower
than the regular media rate. Speed values of 0 or lower have no
meaning and are not allowed. This mechanism enables two general
functionalities. One is client-side scale operations, i.e., the
client receives all the frames and makes the adjustment to the
playback locally. The second is delivery control for the buffering
of media. By specifying a speed over 1.0, the client can build up
the amount of playback time it has present in its buffers to a level
that is sufficient for its needs.
A naive implementation of Speed would only affect the transmission
schedule of the media and has a clear impact on the needed bandwidth.
This would result in the data rate being proportional to the speed
factor. Speed = 1.5, i.e., 50% faster than normal delivery, would
result in a 50% increase in the data-transport rate. Whether or not
that can be supported depends solely on the underlying network path.
Scale may also have some impact on the required bandwidth due to the
manipulation of the content in the new playback schedule. An example
is fast forward where only the independently decodable intra-frames
are included in the media stream. This usage of solely intra-frames
increases the data rate significantly compared to a normal sequence
with the same number of frames, where most frames are encoded using
prediction.
This potential increase of the data rate needs to be handled by the
media sender. The client has requested that the media be delivered
in a specific way, which should be honored. However, the media
sender cannot ignore if the network path between the sender and the
receiver can't handle the resulting media stream. In that case, the
media stream needs to be adapted to fit the available resources of
the path. This can result in a reduced media quality.
The need for bitrate adaptation becomes especially problematic in
connection with the Speed semantics. If the goal is to fill up the
buffer, the client may not want to do that at the cost of reduced
quality. If the client wants to make local playout changes, then it
may actually require that the requested speed be honored. To resolve
this issue, Speed uses a range so that both cases can be supported.
The server is requested to use the highest possible speed value
within the range, which is compatible with the available bandwidth.
As long as the server can maintain a speed value within the range, it
shall not change the media quality, but instead modify the actual
delivery rate in response to available bandwidth and reflect this in
the Speed value in the response. However, if this is not possible,
the server should instead modify the media quality to respect the
lowest speed value and the available bandwidth.
This functionality enables the local scaling implementation to use a
tight range, or even a range where the lower bound equals the upper
bound, to identify that it requires the server to deliver the
requested amount of media time per delivery time, independent of how
much it needs to adapt the media quality to fit within the available
path bandwidth. For buffer filling, it is suitable to use a range
with a reasonable span and with a lower bound at the nominal media
rate 1.0, such as 1.0 - 2.5. If the client wants to reduce the
buffer, it can specify an upper bound that is below 1.0 to force the
server to deliver slower than the nominal media rate.
2.6. Session Maintenance and Termination
The session context that has been established is kept alive by having
the client show liveness. This is done in two main ways:
o Media-transport protocol keep-alive. RTP Control Protocol (RTCP)
may be used when using RTP.
o Any RTSP request referencing the session context.
Section 10.5 discusses the methods for showing liveness in more
depth. If the client fails to show liveness for more than the
established session timeout value (normally 60 seconds), the server
may terminate the context. Other values may be selected by the
server through the inclusion of the timeout parameter in the session
header.
The session context is normally terminated by the client sending a
TEARDOWN request (Section 13.7) to the server referencing the
aggregated control URI. An individual media resource can be removed
from a session context by a TEARDOWN request referencing that
particular media resource. If all media resources are removed from a
session context, the session context is terminated.
A client may keep the session alive indefinitely if allowed by the
server; however, a client is advised to release the session context
when an extended period of time without media delivery activity has
passed. The client can re-establish the session context if required
later. What constitutes an extended period of time is dependent on
the client, server, and their usage. It is recommended that the
client terminate the session before ten times the session timeout
value has passed. A server may terminate the session after one
session timeout period without any client activity beyond keep-alive.
When a server terminates the session context, it does so by sending a
TEARDOWN request indicating the reason.
A server can also request that the client tear down the session and
re-establish it at an alternative server, as may be needed for
maintenance. This is done by using the REDIRECT method
(Section 13.10). The Terminate-Reason header (Section 18.52) is used
to indicate when and why. The Location header indicates where it
should connect if there is an alternative server available. When the
deadline expires, the server simply stops providing the service. To
achieve a clean closure, the client needs to initiate session
termination prior to the deadline. In case the server has no other
server to redirect to, and it wants to close the session for
maintenance, it shall use the TEARDOWN method with a Terminate-Reason
header.
2.7. Extending RTSP
RTSP is quite a versatile protocol that supports extensions in many
different directions. Even this core specification contains several
blocks of functionality that are optional to implement. The use case
and need for the protocol deployment should determine what parts are
implemented. Allowing for extensions makes it possible for RTSP to
address additional use cases. However, extensions will affect the
interoperability of the protocol; therefore, it is important that
they can be added in a structured way.
The client can learn the capability of a server by using the OPTIONS
method (Section 13.1) and the Supported header (Section 18.51). It
can also try and possibly fail using new methods or require that
particular features be supported using the Require (Section 18.43) or
Proxy-Require (Section 18.37) header.
The RTSP, in itself, can be extended in three ways, listed here in
increasing order of the magnitude of changes supported:
o Existing methods can be extended with new parameters, for example,
headers, as long as these parameters can be safely ignored by the
recipient. If the client needs negative acknowledgment when a
method extension is not supported, a tag corresponding to the
extension may be added in the field of the Require or Proxy-
Require headers.
o New methods can be added. If the recipient of the message does
not understand the request, it must respond with error code 501
(Not Implemented) so that the sender can avoid using this method
again. A client may also use the OPTIONS method to inquire about
methods supported by the server. The server must list the methods
it supports using the Public response-header.
o A new version of the protocol can be defined, allowing almost all
aspects (except the position of the protocol version number) to
change. A new version of the protocol must be registered through
a Standards Track document.
The basic capability discovery mechanism can be used to both discover
support for a certain feature and to ensure that a feature is
available when performing a request. For a detailed explanation of
this, see Section 11.
New media delivery protocols may be added and negotiated at session
establishment, in addition to extensions to the core protocol.
Certain types of protocol manipulations can be done through parameter
formats using SET_PARAMETER and GET_PARAMETER.
3. Document Conventions
3.1. Notational Conventions
All the mechanisms specified in this document are described in both
prose and the Augmented Backus-Naur form (ABNF) described in detail
in [RFC5234].
Indented paragraphs are used to provide informative background and
motivation. This is intended to give readers who were not involved
with the formulation of the specification an understanding of why
things are the way they are in RTSP.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
[RFC2119].
The word, "unspecified" is used to indicate functionality or features
that are not defined in this specification. Such functionality
cannot be used in a standardized manner without further definition in
an extension specification to RTSP.
3.2. Terminology
Aggregate control: The concept of controlling multiple streams using
a single timeline, generally one maintained by the server. A
client, for example, uses aggregate control when it issues a
single play or pause message to simultaneously control both the
audio and video in a movie. A session that is under aggregate
control is referred to as an "aggregated session".
Aggregate control URI: The URI used in an RTSP request to refer to
and control an aggregated session. It normally, but not always,
corresponds to the presentation URI specified in the session
description. See Section 13.3 for more information.
Client: The client is the requester of media service from the media
server.
Connection: A transport-layer virtual circuit established between
two programs for the purpose of communication.
Container file: A file that may contain multiple media streams that
often constitute a presentation when played together. The concept
of a container file is not embedded in the protocol. However,
RTSP servers may offer aggregate control on the media streams
within these files.
Continuous media: Data where there is a timing relationship between
source and sink; that is, the sink needs to reproduce the timing
relationship that existed at the source. The most common examples
of continuous media are audio and motion video. Continuous media
can be real time (interactive or conversational), where there is a
"tight" timing relationship between source and sink or it can be
streaming where the relationship is less strict.
Feature tag: A tag representing a certain set of functionality,
i.e., a feature.
IRI: An Internationalized Resource Identifier is similar to a URI
but allows characters from the whole Universal Character Set
(Unicode/ISO 10646), rather than the US-ASCII only. See [RFC3987]
for more information.
Live: A live presentation or session originates media from an event
taking place at the same time as the media delivery. Live
sessions often have an unbound or only loosely defined duration
and seek operations may not be possible.
Media initialization: The datatype- or codec-specific
initialization. This includes such things as clock rates, color
tables, etc. Any transport-independent information that is
required by a client for playback of a media stream occurs in the
media initialization phase of stream setup.
Media parameter: A parameter specific to a media type that may be
changed before or during stream delivery.
Media server: The server providing media-delivery services for one
or more media streams. Different media streams within a
presentation may originate from different media servers. A media
server may reside on the same host or on a different host from
which the presentation is invoked.
(Media) Stream: A single media instance, e.g., an audio stream or a
video stream as well as a single whiteboard or shared application
group. When using RTP, a stream consists of all RTP and RTCP
packets created by a media source within an RTP session.
Message: The basic unit of RTSP communication, consisting of a
structured sequence of octets matching the syntax defined in
Section 20 and transmitted over a transport between RTSP agents.
A message is either a request or a response.
Message body: The information transferred as the payload of a
message (request or response). A message body consists of meta-
information in the form of message body headers and content in the
form of an arbitrary number of data octets, as described in
Section 9.
Non-aggregated control: Control of a single media stream.
Presentation: A set of one or more streams presented to the client
as a complete media feed and described by a presentation
description as defined below. Presentations with more than one
media stream are often handled in RTSP under aggregate control.
Presentation description: A presentation description contains
information about one or more media streams within a presentation,
such as the set of encodings, network addresses, and information
about the content. Other IETF protocols, such as SDP ([RFC4566]),
use the term "session" for a presentation. The presentation
description may take several different formats, including but not
limited to SDP format.
Response: An RTSP response to a request. One type of RTSP message.
If an HTTP response is meant, it is indicated explicitly.
Request: An RTSP request. One type of RTSP message. If an HTTP
request is meant, it is indicated explicitly.
Request-URI: The URI used in a request to indicate the resource on
which the request is to be performed.
RTSP agent: Either an RTSP client, an RTSP server, or an RTSP proxy.
In this specification, there are many capabilities that are common
to these three entities such as the capability to send requests or
receive responses. This term will be used when describing
functionality that is applicable to all three of these entities.
RTSP session: A stateful abstraction upon which the main control
methods of RTSP operate. An RTSP session is a common context; it
is created and maintained on a client's request and can be
destroyed by either the client or server. It is established by an
RTSP server upon the completion of a successful SETUP request
(when a 200 OK response is sent) and is labeled with a session
identifier at that time. The session exists until timed out by
the server or explicitly removed by a TEARDOWN request. An RTSP
session is a stateful entity; an RTSP server maintains an explicit
session state machine (see Appendix B) where most state
transitions are triggered by client requests. The existence of a
session implies the existence of state about the session's media
streams and their respective transport mechanisms. A given
session can have one or more media streams associated with it. An
RTSP server uses the session to aggregate control over multiple
media streams.
Origin server: The server on which a given resource resides.
Seeking: Requesting playback from a particular point in the content
time line.
Transport initialization: The negotiation of transport information
(e.g., port numbers, transport protocols) between the client and
the server.
URI: A Universal Resource Identifier; see [RFC3986]. The URIs used
in RTSP are generally URLs as they give a location for the
resource. As URLs are a subset of URIs, they will be referred to
as URIs to cover also the cases when an RTSP URI would not be a
URL.
URL: A Universal Resource Locator is a URI that identifies the
resource through its primary access mechanism rather than
identifying the resource by name or by some other attribute(s) of
that resource.
4. Protocol Parameters
4.1. RTSP Version
This specification defines version 2.0 of RTSP.
RTSP uses a "<major>.<minor>" numbering scheme to indicate versions
of the protocol. The protocol versioning policy is intended to allow
the sender to indicate the format of a message and its capacity for
understanding further RTSP communication rather than the features
obtained via that communication. No change is made to the version
number for the addition of message components that do not affect
communication behavior or that only add to extensible field values.
The <minor> number is incremented when the changes made to the
protocol add features that do not change the general message parsing
algorithm but that may add to the message semantics and imply
additional capabilities of the sender. The <major> number is
incremented when the format of a message within the protocol is
changed. The version of an RTSP message is indicated by an RTSP-
Version field in the first line of the message. Note that the major
and minor numbers MUST be treated as separate integers and that each
MAY be incremented higher than a single digit. Thus, RTSP/2.4 is a
lower version than RTSP/2.13, which, in turn, is lower than
RTSP/12.3. Leading zeros SHALL NOT be sent and MUST be ignored by
recipients.
4.2. RTSP IRI and URI
RTSP 2.0 defines and registers or updates three URI schemes "rtsp",
"rtsps", and "rtspu". The usage of the last, "rtspu", is unspecified
in RTSP 2.0 and is defined here to register the URI scheme that was
defined in RTSP 1.0. The "rtspu" scheme indicates unspecified
transport of the RTSP messages over unreliable transport means (UDP
in RTSP 1.0). An RTSP server MUST respond with an error code
indicating the "rtspu" scheme is not implemented (501) to a request
that carries a "rtspu" URI scheme.
The details of the syntax of "rtsp" and "rtsps" URIs have been
changed from RTSP 1.0. These changes include the addition of:
o Support for an IPv6 literal in the host part and future IP
literals through a mechanism defined in [RFC3986].
o A new relative format to use in the RTSP elements that is not
required to start with "/".
Neither should have any significant impact on interoperability. If
IPv6 literals are needed in the RTSP URI, then that RTSP server must
be IPv6 capable, and RTSP 1.0 is not a fully IPv6 capable protocol.
If an RTSP 1.0 client attempts to process the URI, the URI will not
match the allowed syntax, it will be considered invalid, and
processing will be stopped. This is clearly a failure to reach the
resource; however, it is not a signification issue as RTSP 2.0
support was needed anyway in both server and client. Thus, failure
will only occur in a later step when there is an RTSP version
mismatch between client and server. The second change will only
occur inside RTSP message headers, as the Request-URI must be an
absolute URI. Thus, such usages will only occur after an agent has
accepted and started processing RTSP 2.0 messages, and an agent using
RTSP 1.0 only will not be required to parse such types of relative
URIs.
This specification also defines the format of RTSP IRIs [RFC3987]
that can be used as RTSP resource identifiers and locators on web
pages, user interfaces, on paper, etc. However, the RTSP request
message format only allows usage of the absolute URI format. The
RTSP IRI format MUST use the rules and transformation for IRIs to
URIs, as defined in [RFC3987]. This allows a URI that matches the
RTSP 2.0 specification, and so is suitable for use in a request, to
be created from an RTSP IRI.
The RTSP IRI and URI are both syntax restricted compared to the
generic syntax defined in [RFC3986] and [RFC3987]:
o An absolute URI requires the authority part; i.e., a host identity
MUST be provided.
o Parameters in the path element are prefixed with the reserved
separator ";".
The "scheme" and "host" parts of all URIs [RFC3986] and IRIs
[RFC3987] are case insensitive. All other parts of RTSP URIs and
IRIs are case sensitive, and they MUST NOT be case mapped.
The fragment identifier is used as defined in Sections 3.5 and 4.3 of
[RFC3986], i.e., the fragment is to be stripped from the IRI by the
requester and not included in the Request-URI. The user agent needs
to interpret the value of the fragment based on the media type the
request relates to; i.e., the media type indicated in Content-Type
header in the response to a DESCRIBE request.
The syntax of any URI query string is unspecified and responder
(usually the server) specific. The query is, from the requester's
perspective, an opaque string and needs to be handled as such.
Please note that relative URIs with queries are difficult to handle
due to the relative URI handling rules of RFC 3986. Any change of
the path element using a relative URI results in the stripping of the
query, which means the relative part needs to contain the query.
The URI scheme "rtsp" requires that commands be issued via a reliable
protocol (within the Internet, TCP), while the scheme "rtsps"
identifies a reliable transport using secure transport (TLS
[RFC5246]); see Section 19.
For the scheme "rtsp", if no port number is provided in the authority
part of the URI, the port number 554 MUST be used. For the scheme
"rtsps", if no port number is provided in the authority part of the
URI port number, the TCP port 322 MUST be used.
A presentation or a stream is identified by a textual media
identifier, using the character set and escape conventions of URIs
[RFC3986]. URIs may refer to a stream or an aggregate of streams;
i.e., a presentation. Accordingly, requests described in Section 13
can apply to either the whole presentation or an individual stream
within the presentation. Note that some request methods can only be
applied to streams, not presentations, and vice versa.
For example, the RTSP URI:
rtsp://media.example.com:554/twister/audiotrack
may identify the audio stream within the presentation "twister",
which can be controlled via RTSP requests issued over a TCP
connection to port 554 of host media.example.com.
Also, the RTSP URI:
rtsp://media.example.com:554/twister
identifies the presentation "twister", which may be composed of audio
and video streams, but could also be something else, such as a random
media redirector.
This does not imply a standard way to reference streams in URIs.
The presentation description defines the hierarchical
relationships in the presentation and the URIs for the individual
streams. A presentation description may name a stream "a.mov" and
the whole presentation "b.mov".
The path components of the RTSP URI are opaque to the client and do
not imply any particular file system structure for the server.
This decoupling also allows presentation descriptions to be used
with non-RTSP media control protocols simply by replacing the
scheme in the URI.
4.3. Session Identifiers
Session identifiers are strings of a length between 8-128 characters.
A session identifier MUST be generated using methods that make it
cryptographically random (see [RFC4086]). It is RECOMMENDED that a
session identifier contain 128 bits of entropy, i.e., approximately
22 characters from a high-quality generator (see Section 21).
However, note that the session identifier does not provide any
security against session hijacking unless it is kept confidential by
the client, server, and trusted proxies.
4.4. Media-Time Formats
RTSP currently supports three different media-time formats defined
below. Additional time formats may be specified in the future.
These time formats can be used with the Range header (Section 18.40)
to request playback and specify at which media position protocol
requests actually will or have taken place. They are also used in
description of the media's properties using the Media-Range header
(Section 18.30). The unqualified format identifier is used on its
own in Accept-Ranges header (Section 18.5) to declare supported time
formats and also in the Range header (Section 18.40) to request the
time format used in the response.
4.4.1. SMPTE-Relative Timestamps
A timestamp may use a format derived from a Society of Motion Picture
and Television Engineers (SMPTE) specification and expresses time
offsets anchored at the start of the media clip. Relative timestamps
are expressed as SMPTE time codes [SMPTE-TC] for frame-level access
accuracy. The time code has the format:
hours:minutes:seconds:frames.subframes
with the origin at the start of the clip. The default SMPTE format
is "SMPTE 30 drop" format, with a frame rate of 29.97 frames per
second. Other SMPTE codes MAY be supported (such as "SMPTE 25")
through the use of "smpte-type". For SMPTE 30, the "frames" field in
the time value can assume the values 0 through 29. The difference
between 30 and 29.97 frames per second is handled by dropping the
first two frame indices (values 00 and 01) of every minute, except
every tenth minute. If the frame and the subframe values are zero,
they may be omitted. Subframes are measured in hundredths of a
frame.
Examples:
smpte=10:12:33:20-
smpte=10:07:33-
smpte=10:07:00-10:07:33:05.01
smpte-25=10:07:00-10:07:33:05.01
4.4.2. Normal Play Time
Normal Play Time (NPT) indicates the stream-absolute position
relative to the beginning of the presentation. The timestamp
consists of two parts: The mandatory first part may be expressed in
either seconds only or in hours, minutes, and seconds. The optional
second part consists of a decimal point and decimal figures and
indicates fractions of a second.
The beginning of a presentation corresponds to 0.0 seconds. Negative
values are not defined.
The special constant "now" is defined as the current instant of a
live event. It MAY only be used for live events and MUST NOT be used
for on-demand (i.e., non-live) content.
NPT is defined as in Digital Storage Media Command and Control
(DSMb;CC) [ISO.13818-6.1995]:
Intuitively, NPT is the clock the viewer associates with a
program. It is often digitally displayed on a DVD player. NPT
advances normally when in normal play mode (scale = 1), advances
at a faster rate when in fast-scan forward (high positive scale
ratio), decrements when in scan reverse (negative scale ratio) and
is fixed in pause mode. NPT is (logically) equivalent to SMPTE
time codes.
Examples:
npt=123.45-125
npt=12:05:35.3-
npt=now-
The syntax is based on ISO 8601 [ISO.8601.2000] and expresses the
time elapsed since presentation start, with two different notations
allowed:
o The npt-hhmmss notation uses an ISO 8601 extended complete
representation of the time of the day format (Section 5.3.1.1 of
[ISO.8601.2000] ) using colons (":") as separators between hours,
minutes, and seconds (hh:mm:ss). The hour counter is not limited
to 0-24 hours; up to nineteen (19) hour digits are allowed.
* In accordance with the requirements of the ISO 8601 time
format, the hours, minutes, and seconds MUST all be present,
with two digits used for minutes and for seconds and with at
least two digits for hours. An NPT of 7 minutes and 0 seconds
is represented as "00:07:00", and an NPT of 392 hours, 0
minutes, and 6 seconds is represented as "392:00:06".
* RTSP 1.0 allowed NPT in the npt-hhmmss notation without any
leading zeros to ensure that implementations don't fail; for
backward compatibility, all RTSP 2.0 implementations are
REQUIRED to support receiving NPT values, hours, minutes, or
seconds, without leading zeros.
o The npt-sec notation expresses the time in seconds, using between
one and nineteen (19) digits.
Both notations allow decimal fractions of seconds as specified in
Section 5.3.1.3 of [ISO.8601.2000], using at most nine digits, and
allowing only "." (full stop) as the decimal separator.
The npt-sec notation is optimized for automatic generation; the npt-
hhmmss notation is optimized for consumption by human readers. The
"now" constant allows clients to request to receive the live feed
rather than the stored or time-delayed version. This is needed since
neither absolute time nor zero time are appropriate for this case.
4.4.3. Absolute Time
Absolute time is expressed using a timestamp based on ISO 8601
[ISO.8601.2000]. The date is a complete representation of the
calendar date in basic format (YYYYMMDD) without separators (per
Section 5.2.1.1 of [ISO.8601.2000]). The time of day is provided in
the complete representation basic format (hhmmss) as specified in
Section 5.3.1.1 of [ISO.8601.2000], allowing decimal fractions of
seconds following Section 5.3.1.3 requiring "." (full stop) as
decimal separator and limiting the number of digits to no more than
nine. The time expressed MUST use UTC (GMT), i.e., no time zone
offsets are allowed. The full date and time specification is the
eight-digit date followed by a "T" followed by the six-digit time
value, optionally followed by a full stop followed by one to nine
fractions of a second and ended by "Z", e.g., YYYYMMDDThhmmss.ssZ.
The reasons for this time format rather than using "Date and Time
on the Internet: Timestamps" [RFC3339] are historic. We continue
to use the format specified in RTSP 1.0. The motivations raised
in RFC 3339 apply to why a selection from ISO 8601 was made;
however, a different and even more restrictive selection was
applied in this case.
Below are three examples of media time formats, first, a request for
a clock format range request for a starting time of November 8, 1996
at 14 h 37 min and 20 1/4 seconds UTC playing for 10 min and 5
seconds, followed by a Media-Properties header's "Time-Limited" UTC
property for the 24th of December 2014 at 15 hours and 00 minutes,
and finally a Terminate-Reason header "time" property for the 18th of
June 2013 at 16 hours, 12 minutes, and 56 seconds:
clock=19961108T143720.25Z-19961108T144725.25Z
Time-Limited=20141224T1500Z
time=20130618T161256Z
4.5. Feature Tags
Feature tags are unique identifiers used to designate features in
RTSP. These tags are used in Require (Section 18.43), Proxy-Require
(Section 18.37), Proxy-Supported (Section 18.38), Supported
(Section 18.51), and Unsupported (Section 18.55) header fields.
A feature tag definition MUST indicate which combination of clients,
servers, or proxies to which it applies.
The creator of a new RTSP feature tag should either prefix the
feature tag with a reverse domain name (e.g.,
"com.example.mynewfeature" is an apt name for a feature whose
inventor can be reached at "example.com") or register the new feature
tag with the Internet Assigned Numbers Authority (IANA). (See
Section 22, "IANA Considerations".)
The usage of feature tags is further described in Section 11, which
deals with capability handling.
4.6. Message Body Tags
Message body tags are opaque strings that are used to compare two
message bodies from the same resource, for example, in caches or to
optimize setup after a redirect. Message body tags can be carried in
the MTag header (see Section 18.31) or in SDP (see Appendix D.1.9).
MTag is similar to ETag in HTTP/1.1 (see Section 3.11 of [RFC2068]).
A message body tag MUST be unique across all versions of all message
bodies associated with a particular resource. A given message body
tag value MAY be used for message bodies obtained by requests on
different URIs. The use of the same message body tag value in
conjunction with message bodies obtained by requests on different
URIs does not imply the equivalence of those message bodies.
Message body tags are used in RTSP to make some methods conditional.
The methods are made conditional through the inclusion of headers;
see Section 18.24 and Section 18.26 for information on the If-Match
and If-None-Match headers, respectively. Note that RTSP message body
tags apply to the complete presentation, i.e., both the presentation
description and the individual media streams. Thus, message body
tags can be used to verify at setup time after a redirect that the
same session description applies to the media at the new location
using the If-Match header.
4.7. Media Properties
When an RTSP server handles media, it is important to consider the
different properties a media instance for delivery and playback can
have. This specification considers the media properties listed below
in its protocol operations. They are derived from the differences
between a number of supported usages.
On-demand: Media that has a fixed (given) duration that doesn't
change during the lifetime of the RTSP session and is known at the
time of the creation of the session. It is expected that the
content of the media will not change, even if the representation,
such as encoding, or quality, may change. Generally, one can
seek, i.e., request any range, within the media.
Dynamic On-demand: This is a variation of the on-demand case where
external methods are used to manipulate the actual content of the
media setup for the RTSP session. The main example is content
defined by a playlist.
Live: Live media represents a progressing content stream (such as
broadcast TV) where the duration may or may not be known. It is
not seekable, only the content presently being delivered can be
accessed.
Live with Recording: A live stream that is combined with a server-
side capability to store and retain the content of the live
session and allow for random access delivery within the part of
the already-recorded content. The actual behavior of the media
stream is very much dependent on the retention policy for the
media stream; either the server will be able to capture the
complete media stream or it will have a limitation in how much
will be retained. The media range will dynamically change as the
session progress. For servers with a limited amount of storage
available for recording, there will typically be a sliding window
that moves forward while new data is made available and older data
is discarded.
To cover the above usages, the following media properties with
appropriate values are specified.
4.7.1. Random Access and Seeking
Random access is the ability to specify and get media delivered
starting from any time (instant) within the content, an operation
called "seeking". The Media-Properties header will indicate the
general capability for a media resource to perform random access.
Random-Access: The media is seekable to any out of a large number of
points within the media. Due to media-encoding limitations, a
particular point may not be reachable, but seeking to a point
close by is enabled. A floating-point number of seconds may be
provided to express the worst-case distance between random access
points.
Beginning-Only: Seeking is only possible to the beginning of the
content.
No-Seeking: Seeking is not possible at all.
If random access is possible, as indicated by the Media-Properties
header, the actual behavior policy when seeking can be controlled
using the Seek-Style header (Section 18.47).
4.7.2. Retention
The following retention policies are used by media to limit possible
protocol operations:
Unlimited: The media will not be removed as long as the RTSP session
is in existence.
Time-Limited: The media will not be removed before the given
wallclock time. After that time, it may or may not be available
anymore.
Time-Duration: The media (on fragment or unit basis) will be
retained for the specified duration.
4.7.3. Content Modifications
The media content and its timeline can be of different types, e.g.
pre-produced content on demand, a live source that is being generated
as time progresses, or something that is dynamically altered or
recomposed during playback. Therefore, a media property for content
modifications is needed and the following initial values are defined:
Immutable: The content of the media will not change, even if the
representation, such as encoding or quality changes.
Dynamic: The content can change due to external methods or triggers,
such as playlists, but this will be announced by explicit updates.
Time-Progressing: As time progresses, new content will become
available. If the content is also retained, it will become longer
as everything between the start point and the point currently
being made available can be accessed. If the media server uses a
sliding-window policy for retention, the start point will also
change as time progresses.
4.7.4. Supported Scale Factors
A particular media content item often supports only a limited set or
range of scales when delivering the media. To enable the client to
know what values or ranges of scale operations that the whole content
or the current position supports, a media properties attribute for
this is defined that contains a list with the values or ranges that
are supported. The attribute is named "Scales". The "Scales"
attribute may be updated at any point in the content due to content
consisting of spliced pieces or content being dynamically updated by
out-of-band mechanisms.
4.7.5. Mapping to the Attributes
This section shows examples of how one would map the above usages to
the properties and their values.
Example of On-Demand:
Random Access: Random-Access=5.0, Content Modifications:
Immutable, Retention: Unlimited or Time-Limited.
Example of Dynamic On-Demand:
Random Access: Random-Access=3.0, Content Modifications: Dynamic,
Retention: Unlimited or Time-Limited.
Example of Live:
Random Access: No-Seeking, Content Modifications: Time-
Progressing, Retention: Time-Duration=0.0
Example of Live with Recording:
Random Access: Random-Access=3.0, Content Modifications: Time-
Progressing, Retention: Time-Duration=7200.0
5. RTSP Message
RTSP is a text-based protocol that uses the ISO 10646 character set
in UTF-8 encoding per RFC 3629 [RFC3629]. Lines MUST be terminated
by a CRLF.
Text-based protocols make it easier to add optional parameters in
a self-describing manner. Since the number of parameters and the
frequency of commands is low, processing efficiency is not a
concern. Text-based protocols, if used carefully, also allow easy
implementation of research prototypes in scripting languages such
as Python, PHP, Perl and TCL.
The ISO 10646 character set avoids character-set switching, but is
invisible to the application as long as US-ASCII is being used. This
is also the encoding used for text fields in RTCP [RFC3550].
A request contains a method, the object the method is operating upon,
and parameters to further describe the method. Methods are
idempotent unless otherwise noted. Methods are also designed to
require little or no state maintenance at the media server.
5.1. Message Types
RTSP messages are either requests from client to server or from
server to client, and responses in the reverse direction. Request
(Section 7) and response (Section 8) messages use a format based on
the generic message format of RFC 5322 [RFC5322] for transferring
bodies (the payload of the message). Both types of messages consist
of a start-line, zero or more header fields (also known as
"headers"), an empty line (i.e., a line with nothing preceding the
CRLF) indicating the end of the headers, and possibly the data of the
message body. The ABNF [RFC5234] below is for illustration only; the
formal message specification is presented in Section 20.2.2.
generic-message = start-line
*(rtsp-header CRLF)
CRLF
[ message-body-data ]
start-line = Request-Line / Status-Line
In the interest of robustness, agents MUST ignore any empty line(s)
received where a Request-Line or Status-Line is expected. In other
words, if the agent is reading the protocol stream at the beginning
of a message and receives any number of CRLFs first, it MUST ignore
all of the CRLFs.
5.2. Message Headers
RTSP header fields (see Section 18) include general-header, request-
header, response-header, and message body header fields.
The order in which header fields with differing field names are
received is not significant. However, it is "good practice" to send
general-header fields first, followed by a request-header or
response-header field, and ending with the message body header
fields.
Multiple header fields with the same field-name MAY be present in a
message if and only if the entire field-value for that header field
is defined as a comma-separated list. It MUST be possible to combine
the multiple header fields into one "field-name: field-value" pair,
without changing the semantics of the message, by appending each
subsequent field-value to the first, each separated by a comma. The
order in which header fields with the same field-name are received is
therefore significant to the interpretation of the combined field
value; thus, a proxy MUST NOT change the order of these field-values
when a message is forwarded.
Unknown message headers MUST be ignored (skipping over the header to
the next protocol element, and not causing an error) by an RTSP
server or client. An RTSP proxy MUST forward unknown message
headers. Message headers defined outside of this specification that
are required to be interpreted by the RTSP agent will need to use
feature tags (Section 4.5) and include them in the appropriate
Require (Section 18.43) or Proxy-Require (Section 18.37) header.
5.3. Message Body
The message body (if any) of an RTSP message is used to carry further
information for a particular resource associated with the request or
response. An example of a message body is an SDP message.
The presence of a message body in either a request or a response MUST
be signaled by the inclusion of a Content-Length header (see
Section 18.17) and Content-Type header (see Section 18.19). A
message body MUST NOT be included in a request or response if the
specification of the particular method (see Method Definitions
(Section 13)) does not allow sending a message body. In case a
message body is received in a message when not expected, the message
body data SHOULD be discarded. This is to allow future extensions to
define optional use of a message body.
5.4. Message Length
An RTSP message that does not contain any message body is terminated
by the first empty line after the header fields (note: an empty line
is a line with nothing preceding the CRLF.). In RTSP messages that
contain message bodies, the empty line is followed by the message
body. The length of that body is determined by the value of the
Content-Length header (Section 18.17). The value in the header
represents the length of the message body in octets. If this header
field is not present, a value of zero is assumed, i.e., no message
body present in the message. Unlike an HTTP message, an RTSP message
MUST contain a Content-Length header whenever it contains a message
body. Note that RTSP does not support the HTTP/1.1 "chunked"
transfer coding (see Section 4.1 of [RFC7230]).
Given the moderate length of presentation descriptions returned,
the server should always be able to determine its length, even if
it is generated dynamically, making the chunked transfer encoding
unnecessary.
6. General-Header Fields
General headers are headers that may be used in both requests and
responses. The general-headers are listed in Table 1:
+--------------------+----------------+
| Header Name | Defined in |
+--------------------+----------------+
| Accept-Ranges | Section 18.5 |
| | |
| Cache-Control | Section 18.11 |
| | |
| Connection | Section 18.12 |
| | |
| CSeq | Section 18.20 |
| | |
| Date | Section 18.21 |
| | |
| Media-Properties | Section 18.29 |
| | |
| Media-Range | Section 18.30 |
| | |
| Pipelined-Requests | Section 18.33 |
| | |
| Proxy-Supported | Section 18.38 |
| | |
| Range | Section 18.40 |
| | |
| RTP-Info | Section 18.45 |
| | |
| Scale | Section 18.46 |
| | |
| Seek-Style | Section 18.47 |
| | |
| Server | Section 18.48 |
| | |
| Session | Section 18.49 |
| | |
| Speed | Section 18.50 |
| | |
| Supported | Section 18.51 |
| | |
| Timestamp | Section 18.53 |
| | |
| Transport | Section 18.54 |
| | |
| User-Agent | Section 18.56 |
| | |
| Via | Section 18.57 |
+--------------------+----------------+
Table 1: The General Headers Used in RTSP
7. Request
A request message uses the format outlined below regardless of the
direction of a request, whether client to server or server to client:
o Request line, containing the method to be applied to the resource,
the identifier of the resource, and the protocol version in use;
o Zero or more Header lines, which can be of the following types:
general-headers (Section 6), request-headers (Section 7.2), or
message body headers (Section 9.1);
o One empty line (CRLF) to indicate the end of the header section;
o Optionally, a message body, consisting of one or more lines. The
length of the message body in octets is indicated by the Content-
Length message header.
7.1. Request Line
The request line provides the key information about the request: what
method, on what resources, and using which RTSP version. The methods
that are defined by this specification are listed in Table 2.
+---------------+----------------+
| Method | Defined in |
+---------------+----------------+
| DESCRIBE | Section 13.2 |
| | |
| GET_PARAMETER | Section 13.8 |
| | |
| OPTIONS | Section 13.1 |
| | |
| PAUSE | Section 13.6 |
| | |
| PLAY | Section 13.4 |
| | |
| PLAY_NOTIFY | Section 13.5 |
| | |
| REDIRECT | Section 13.10 |
| | |
| SETUP | Section 13.3 |
| | |
| SET_PARAMETER | Section 13.9 |
| | |
| TEARDOWN | Section 13.7 |
+---------------+----------------+
Table 2: The RTSP Methods
The syntax of the RTSP request line has the following:
<Method> SP <Request-URI> SP <RTSP-Version> CRLF
Note: This syntax cannot be freely changed in future versions of
RTSP. This line needs to remain parsable by older RTSP
implementations since it indicates the RTSP version of the message.
In contrast to HTTP/1.1 [RFC7230], RTSP requests identify the
resource through an absolute RTSP URI (including scheme, host, and
port) (see Section 4.2) rather than just the absolute path.
HTTP/1.1 requires servers to understand the absolute URI, but
clients are supposed to use the Host request-header. This is
purely needed for backward compatibility with HTTP/1.0 servers, a
consideration that does not apply to RTSP.
An asterisk "*" can be used instead of an absolute URI in the
Request-URI part to indicate that the request does not apply to a
particular resource but to the server or proxy itself, and is only
allowed when the request method does not necessarily apply to a
resource.
For example:
OPTIONS * RTSP/2.0
An OPTIONS in this form will determine the capabilities of the server
or the proxy that first receives the request. If the capability of
the specific server needs to be determined, without regard to the
capability of an intervening proxy, the server should be addressed
explicitly with an absolute URI that contains the server's address.
For example:
OPTIONS rtsp://example.com RTSP/2.0
7.2. Request-Header Fields
The RTSP headers in Table 3 can be included in a request, as request-
headers, to modify the specifics of the request.
+---------------------+----------------+
| Header | Defined in |
+---------------------+----------------+
| Accept | Section 18.1 |
| | |
| Accept-Credentials | Section 18.2 |
| | |
| Accept-Encoding | Section 18.3 |
| | |
| Accept-Language | Section 18.4 |
| | |
| Authorization | Section 18.8 |
| | |
| Bandwidth | Section 18.9 |
| | |
| Blocksize | Section 18.10 |
| | |
| From | Section 18.23 |
| | |
| If-Match | Section 18.24 |
| | |
| If-Modified-Since | Section 18.25 |
| | |
| If-None-Match | Section 18.26 |
| | |
| Notify-Reason | Section 18.32 |
| | |
| Proxy-Authorization | Section 18.36 |
| | |
| Proxy-Require | Section 18.37 |
| | |
| Referrer | Section 18.41 |
| | |
| Request-Status | Section 18.42 |
| | |
| Require | Section 18.43 |
| | |
| Terminate-Reason | Section 18.52 |
+---------------------+----------------+
Table 3: The RTSP Request-Headers
Detailed header definitions are provided in Section 18.
New request-headers may be defined. If the receiver of the request
is required to understand the request-header, the request MUST
include a corresponding feature tag in a Require or Proxy-Require
header to ensure the processing of the header.
8. Response
After receiving and interpreting a request message, the recipient
responds with an RTSP response message. Normally, there is only one,
final, response. Responses using the response code class 1xx is the
only class for which there MAY be sent one or more responses prior to
the final response message.
The valid response codes and the methods they can be used with are
listed in Table 4.
8.1. Status-Line
The first line of a response message is the Status-Line, consisting
of the protocol version followed by a numeric status code and the
textual phrase associated with the status code, with each element
separated by SP characters. No CR or LF is allowed except in the
final CRLF sequence.
<RTSP-Version> SP <Status-Code> SP <Reason Phrase> CRLF
8.1.1. Status Code and Reason Phrase
The Status-Code element is a 3-digit integer result code of the
attempt to understand and satisfy the request. These codes are fully
defined in Section 17. The reason phrase is intended to give a short
textual description of the Status-Code. The Status-Code is intended
for use by automata and the reason phrase is intended for the human
user. The client is not required to examine or display the reason
phrase.
The first digit of the Status-Code defines the class of response.
The last two digits do not have any categorization role. There are
five values for the first digit:
1xx: Informational - Request received, continuing process
2xx: Success - The action was successfully received, understood, and
accepted
3rr: Redirection - Further action needs to be taken in order to
complete the request (3rr rather than 3xx is used as 304 is
excluded; see Section 17.3)
4xx: Client Error - The request contains bad syntax or cannot be
fulfilled
5xx: Server Error - The server failed to fulfill an apparently valid
request
The individual values of the numeric status codes defined for RTSP
2.0, and an example set of corresponding reason phrases, are
presented in Table 4. The reason phrases listed here are only
recommended; they may be replaced by local equivalents without
affecting the protocol. Note that RTSP adopted most HTTP/1.1
[RFC2068] status codes and then added RTSP-specific status codes
starting at x50 to avoid conflicts with future HTTP status codes that
are desirable to import into RTSP. All these codes are RTSP specific
and RTSP has its own registry separate from HTTP for status codes.
RTSP status codes are extensible. RTSP applications are not required
to understand the meaning of all registered status codes, though such
understanding is obviously desirable. However, applications MUST
understand the class of any status code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the
x00 status code of that class, with an exception for unknown 3xx
codes, which MUST be treated as a 302 (Found). The reason for that
exception is that the status code 300 (Multiple Choices in HTTP) is
not defined for RTSP. A response with an unrecognized status code
MUST NOT be cached. For example, if an unrecognized status code of
431 is received by the client, it can safely assume that there was
something wrong with its request and treat the response as if it had
received a 400 status code. In such cases, user agents SHOULD
present to the user the message body returned with the response,
since that message body is likely to include human-readable
information that will explain the unusual status.
+------+---------------------------------+--------------------------+
| Code | Reason | Method |
+------+---------------------------------+--------------------------+
| 100 | Continue | all |
| | | |
| 200 | OK | all |
| | | |
| 301 | Moved Permanently | all |
| | | |
| 302 | Found | all |
| | | |
| 303 | See Other | n/a |
| | | |
| 304 | Not Modified | all |
| | | |
| 305 | Use Proxy | all |
| | | |
| 400 | Bad Request | all |
| | | |
| 401 | Unauthorized | all |
| | | |
| 402 | Payment Required | all |
| | | |
| 403 | Forbidden | all |
| | | |
| 404 | Not Found | all |
| | | |
| 405 | Method Not Allowed | all |
| | | |
| 406 | Not Acceptable | all |
| | | |
| 407 | Proxy Authentication Required | all |
| | | |
| 408 | Request Timeout | all |
| | | |
| 410 | Gone | all |
| | | |
| 412 | Precondition Failed | DESCRIBE, SETUP |
| | | |
| 413 | Request Message Body Too Large | all |
| | | |
| 414 | Request-URI Too Long | all |
| | | |
| 415 | Unsupported Media Type | all |
| | | |
| 451 | Parameter Not Understood | SET_PARAMETER, |
| | | GET_PARAMETER |
| | | |
| 452 | reserved | n/a |
| | | |
| 453 | Not Enough Bandwidth | SETUP |
| | | |
| 454 | Session Not Found | all |
| | | |
| 455 | Method Not Valid in This State | all |
| | | |
| 456 | Header Field Not Valid for | all |
| | Resource | |
| | | |
| 457 | Invalid Range | PLAY, PAUSE |
| | | |
| 458 | Parameter Is Read-Only | SET_PARAMETER |
| | | |
| 459 | Aggregate Operation Not Allowed | all |
| | | |
| 460 | Only Aggregate Operation | all |
| | Allowed | |
| | | |
| 461 | Unsupported Transport | all |
| | | |
| 462 | Destination Unreachable | all |
| | | |
| 463 | Destination Prohibited | SETUP |
| | | |
| 464 | Data Transport Not Ready Yet | PLAY |
| | | |
| 465 | Notification Reason Unknown | PLAY_NOTIFY |
| | | |
| 466 | Key Management Error | all |
| | | |
| 470 | Connection Authorization | all |
| | Required | |
| | | |
| 471 | Connection Credentials Not | all |
| | Accepted | |
| | | |
| 472 | Failure to Establish Secure | all |
| | Connection | |
| | | |
| 500 | Internal Server Error | all |
| | | |
| 501 | Not Implemented | all |
| | | |
| 502 | Bad Gateway | all |
| | | |
| 503 | Service Unavailable | all |
| | | |
| 504 | Gateway Timeout | all |
| | | |
| 505 | RTSP Version Not Supported | all |
| | | |
| 551 | Option Not Supported | all |
| | | |
| 553 | Proxy Unavailable | all |
+------+---------------------------------+--------------------------+
Table 4: Status Codes and Their Usage with RTSP Methods
8.2. Response Headers
The response-headers allow the request recipient to pass additional
information about the response that cannot be placed in the Status-
Line. This header gives information about the server and about
further access to the resource identified by the Request-URI. All
headers currently classified as response-headers are listed in
Table 5.
+------------------------+----------------+
| Header | Defined in |
+------------------------+----------------+
| Authentication-Info | Section 18.7 |
| | |
| Connection-Credentials | Section 18.13 |
| | |
| Location | Section 18.28 |
| | |
| MTag | Section 18.31 |
| | |
| Proxy-Authenticate | Section 18.34 |
| | |
| Public | Section 18.39 |
| | |
| Retry-After | Section 18.44 |
| | |
| Unsupported | Section 18.55 |
| | |
| WWW-Authenticate | Section 18.58 |
+------------------------+----------------+
Table 5: The RTSP Response Headers
Response-header names can be extended reliably only in combination
with a change in the protocol version. However, the usage of feature
tags in the request allows the responding party to learn the
capability of the receiver of the response. A new or experimental
header can be given the semantics of response-header if all parties
in the communication recognize them to be a response-header.
Unrecognized headers in responses MUST be ignored.
9. Message Body
Some request and response messages include a message body, if not
otherwise restricted by the request method or response status code.
The message body consists of the content data itself (see also
Section 5.3).
The SET_PARAMETER and GET_PARAMETER requests and responses, and the
DESCRIBE response as defined by this specification, can have a
message body; the purpose of the message body is defined in each
case. All 4xx and 5xx responses MAY also have a message body to
carry additional response information. Generally, a message body MAY
be attached to any RTSP 2.0 request or response, but the content of
the message body MAY be ignored by the receiver. Extensions to this
specification can specify the purpose and content of message bodies,
including requiring their inclusion.
In this section, both sender and recipient refer to either the client
or the server, depending on who sends and who receives the message
body.
9.1. Message Body Header Fields
Message body header fields define meta-information about the content
data in the message body. The message body header fields are listed
in Table 6.
+------------------+----------------+
| Header | Defined in |
+------------------+----------------+
| Allow | Section 18.6 |
| | |
| Content-Base | Section 18.14 |
| | |
| Content-Encoding | Section 18.15 |
| | |
| Content-Language | Section 18.16 |
| | |
| Content-Length | Section 18.17 |
| | |
| Content-Location | Section 18.18 |
| | |
| Content-Type | Section 18.19 |
| | |
| Expires | Section 18.22 |
| | |
| Last-Modified | Section 18.27 |
+------------------+----------------+
Table 6: The RTSP Message Body Headers
The extension-header mechanism allows additional message body header
fields to be defined without changing the protocol, but these fields
cannot be assumed to be recognizable by the recipient. Unrecognized
header fields MUST be ignored by the recipient and forwarded by
proxies.
9.2. Message Body
An RTSP message with a message body MUST include the Content-Type and
Content-Length headers. When a message body is included with a
message, the data type of that content data is determined via the
Content-Type and Content-Encoding header fields.
Content-Type specifies the media type of the underlying data. There
is no default media format and the actual format used in the body is
required to be explicitly stated in the Content-Type header. By
being explicit and always requiring the inclusion of the Content-Type
header with accurate information, one avoids the many pitfalls in a
heuristic-based interpretation of the body content. The user
experience of HTTP and email have suffered from relying on such
heuristics.
Content-Encoding may be used to indicate any additional content-
codings applied to the data, usually for the purpose of data
compression, that are a property of the requested resource. The
default encoding is 'identity', i.e. no transformation of the message
body.
The Content-Length of a message is the length of the content,
measured in octets.
9.3. Message Body Format Negotiation
The content format of the message body is provided using the Content-
Type header (Section 18.19). To enable the responder of a request to
determine which media type it should use, the requester may include
the Accept header (Section 18.1) in a request to identify supported
media types or media type ranges suitable to the response. In case
the responder is not supporting any of the specified formats, then
the request response will be a 406 (Not Acceptable) error code.
The media types that may be used on requests with message bodies need
to be determined through the use of feature tags, specification
requirement, or trial and error. Trial and error works because when
the responder does not support the media type of the message body, it
will respond with a 415 (Unsupported Media Type).
The formats supported and their negotiation is done individually on a
per method and direction (request or response body) direction.
Requirements on supporting particular media types for use as message
bodies in requests and response SHALL also be specified on a per-
method and per-direction basis.
10. Connections
RTSP messages are transferred between RTSP agents and proxies using a
transport connection. This transport connection uses TCP or TCP/TLS.
This transport connection is referred to as the "connection" or "RTSP
connection" within this document.
RTSP requests can be transmitted using the two different connection
scenarios listed below:
o persistent - a transport connection is used for several request/
response transactions;
o transient - a transport connection is used for each single
request/response transaction.
RFC 2326 attempted to specify an optional mechanism for transmitting
RTSP messages in connectionless mode over a transport protocol such
as UDP. However, it was not specified in sufficient detail to allow
for interoperable implementations. In an attempt to reduce
complexity and scope, and due to lack of interest, RTSP 2.0 does not
attempt to define a mechanism for supporting RTSP over UDP or other
connectionless transport protocols. A side effect of this is that
RTSP requests MUST NOT be sent to multicast groups since no
connection can be established with a specific receiver in multicast
environments.
Certain RTSP headers, such as the CSeq header (Section 18.20), which
may appear to be relevant only to connectionless transport scenarios,
are still retained and MUST be implemented according to this
specification. In the case of CSeq, it is quite useful for matching
responses to requests if the requests are pipelined (see Section 12).
It is also useful in proxies for keeping track of the different
requests when aggregating several client requests on a single TCP
connection.
10.1. Reliability and Acknowledgements
Since RTSP messages are transmitted using reliable transport
protocols, they MUST NOT be retransmitted at the RTSP level.
Instead, the implementation must rely on the underlying transport to
provide reliability. The RTSP implementation may use any indication
of reception acknowledgment of the message from the underlying
transport protocols to optimize the RTSP behavior.
If both the underlying reliable transport, such as TCP, and the
RTSP application retransmit requests, each packet loss or message
loss may result in two retransmissions. The receiver typically
cannot take advantage of the application-layer retransmission
since the transport stack will not deliver the application-layer
retransmission before the first attempt has reached the receiver.
If the packet loss is caused by congestion, multiple
retransmissions at different layers will exacerbate the
congestion.
Lack of acknowledgment of an RTSP request should be handled within
the constraints of the connection timeout considerations described
below (Section 10.4).
10.2. Using Connections
A TCP transport can be used for both persistent connections (for
several message exchanges) and transient connections (for a single
message exchange). Implementations of this specification MUST
support RTSP over TCP. The scheme of the RTSP URI (Section 4.2)
allows the client to specify the port it will contact the server on,
and defines the default port to use if one is not explicitly given.
In addition to the registered default ports, i.e., 554 (rtsp) and 322
(rtsps), there is an alternative port 8554 registered. This port may
provide some benefits over non-registered ports if an RTSP server is
unable to use the default ports. The benefits may include
preconfigured security policies as well as classifiers in network
monitoring tools.
An RTSP client opening a TCP connection to access a particular
resource as identified by a URI uses the IP address and port derived
from the host and port parts of the URI. The IP address is either
the explicit address provided in the URI or any of the addresses
provided when performing A and AAAA record DNS lookups of the
hostname in the URI.
A server MUST handle both persistent and transient connections.
Transient connections facilitate mechanisms for fault tolerance.
They also allow for application-layer mobility. A server-and-
client pair that supports transient connections can survive the
loss of a TCP connection; e.g., due to a NAT timeout. When the
client has discovered that the TCP connection has been lost, it
can set up a new one when there is need to communicate again.
A persistent connection is RECOMMENDED to be used for all
transactions between the server and client, including messages for
multiple RTSP sessions. However, a persistent connection MAY be
closed after a few message exchanges. For example, a client may use
a persistent connection for the initial SETUP and PLAY message
exchanges in a session and then close the connection. Later, when
the client wishes to send a new request, such as a PAUSE for the
session, a new connection would be opened. This connection may be
either transient or persistent.
An RTSP agent MAY use one connection to handle multiple RTSP sessions
on the same server. The RTSP agent SHALL NOT use more than one
connection per RTSP session at any given point.
Having only one connection in use at any time avoids confusion
regarding on which connection any server-to-client requests shall
be sent. Using a single connection for multiple RTSP sessions
also saves complexity by enabling the server to maintain less
state about its connection resources on the server. Not using
more than one connection at a time for a particular RTSP session
avoids wasting connection resources and allows the server to track
only the most recently used client-to-server connection for each
RTSP session as being the currently valid server-to-client
connection.
RTSP allows a server to send requests to a client. However, this can
be supported only if a client establishes a persistent connection
with the server. In cases where a persistent connection does not
exist between a server and its client, due to the lack of a signaling
channel, the server may be forced to silently discard RTSP messages,
and it may even drop an RTSP session without notifying the client.
An example of such a case is when the server desires to send a
REDIRECT request for an RTSP session to the client but is not able to
do so because it cannot reach the client. A server that attempts to
send a request to a client that has no connection currently to the
server SHALL discard the request.
Without a persistent connection between the client and the server,
the media server has no reliable way of reaching the client.
Because of the likely failure of server-to-client established
connections, the server will not even attempt establishing any
connection.
Queuing of server-to-client requests has been considered.
However, a security issue exists as to how it might be possible to
authorize a client establishing a new connection as being a
legitimate receiver of a request related to a particular RTSP
session, without the client first issuing requests related to the
pending request. Thus, it would be likely to make any such
requests even more delayed and less useful.
The sending of client and server requests can be asynchronous events.
To avoid deadlock situations, both client and server MUST be able to
send and receive requests simultaneously. As an RTSP response may be
queued up for transmission, reception or processing behind the peer
RTSP agent's own requests, all RTSP agents are required to have a
certain capability of handling outstanding messages. A potential
issue is that outstanding requests may time out despite being
processed by the peer; this can be due to the response being caught
in the queue behind a number of requests that the RTSP agent is
processing but that take some time to complete. To avoid this
problem, an RTSP agent should buffer incoming messages locally so
that any response messages can be processed immediately upon
reception. If responses are separated from requests and directly
forwarded for processing, not only can the result be used
immediately, the state associated with that outstanding request can
also be released. However, buffering a number of requests on the
receiving RTSP agent consumes resources and enables a resource
exhaustion attack on the agent. Therefore, this buffer should be
limited so that an unreasonable number of requests or total message
size is not allowed to consume the receiving agent's resources. In
most APIs, having the receiving agent stop reading from the TCP
socket will result in TCP's window being clamped, thus forcing the
buffering onto the sending agent when the load is larger than
expected. However, as both RTSP message sizes and frequency may be
changed in the future by protocol extensions, an agent should be
careful about taking harsher measurements against a potential attack.
When under attack, an RTSP agent can close TCP connections and
release state associated with that TCP connection.
To provide some guidance on what is reasonable, the following
guidelines are given. It is RECOMMENDED that:
o an RTSP agent should not have more than 10 outstanding requests
per RTSP session;
o an RTSP agent should not have more than 10 outstanding requests
that are not related to an RTSP session or that are requesting to
create an RTSP session.
In light of the above, it is RECOMMENDED that clients use persistent
connections whenever possible. A client that supports persistent
connections MAY "pipeline" its requests (see Section 12).
RTSP agents can send requests to multiple different destinations,
either server or client contexts over the same connection to a proxy.
Then, the proxy forks the message to the different destinations over
proxy-to-agent connections. In these cases when multiple requests
are outstanding, the requesting agent MUST be ready to receive the
responses out of order compared to the order they where sent on the
connection. The order between multiple messages for each destination
will be maintained; however, the order between response from
different destinations can be different.
The reason for this is to avoid a head-of-line blocking situation.
In a sequence of requests, an early outstanding request may take
time to be processed at one destination. Simultaneously, a
response from any other destination that was later in the sequence
of requests may have arrived at the proxy; thus, allowing out-of-
order responses avoids forcing the proxy to buffer this response
and instead deliver it as soon as possible. Note, this will not
affect the order in which the messages sent to each separate
destination were processed at the request destination.
This scenario can occur in two cases involving proxies. The first is
a client issuing requests for sessions on different servers using a
common client-to-proxy connection. The second is for server-to-
client requests, like REDIRECT being sent by the server over a common
transport connection the proxy created for its different connecting
clients.
10.3. Closing Connections
The client MAY close a connection at any point when no outstanding
request/response transactions exist for any RTSP session being
managed through the connection. The server, however, SHOULD NOT
close a connection until all RTSP sessions being managed through the
connection have been timed out (Section 18.49). A server SHOULD NOT
close a connection immediately after responding to a session-level
TEARDOWN request for the last RTSP session being controlled through
the connection. Instead, the server should wait for a reasonable
amount of time for the client to receive and act upon the TEARDOWN
response and then initiate the connection closing. The server SHOULD
wait at least 10 seconds after sending the TEARDOWN response before
closing the connection.
This is to ensure that the client has time to issue a SETUP for a
new session on the existing connection after having torn the last
one down. Ten seconds should give the client ample opportunity to
get its message to the server.
A server SHOULD NOT close the connection directly as a result of
responding to a request with an error code.
Certain error responses such as 460 (Only Aggregate Operation
Allowed) (Section 17.4.24) are used for negotiating capabilities
of a server with respect to content or other factors. In such
cases, it is inefficient for the server to close a connection on
an error response. Also, such behavior would prevent
implementation of advanced or special types of requests or result
in extra overhead for the client when testing for new features.
On the other hand, keeping connections open after sending an error
response poses a Denial-of-Service (DoS) security risk
(Section 21).
The server MAY close a connection if it receives an incomplete
message and if the message is not completed within a reasonable
amount of time. It is RECOMMENDED that the server wait at least 10
seconds for the completion of a message or for the next part of the
message to arrive (which is an indication that the transport and the
client are still alive). Servers believing they are under attack or
that are otherwise starved for resources during that event MAY
consider using a shorter timeout.
If a server closes a connection while the client is attempting to
send a new request, the client will have to close its current
connection, establish a new connection, and send its request over the
new connection.
An RTSP message SHOULD NOT be terminated by closing the connection.
Such a message MAY be considered to be incomplete by the receiver and
discarded. An RTSP message is properly terminated as defined in
Section 5.
10.4. Timing Out Connections and RTSP Messages
Receivers of a request (responders) SHOULD respond to requests in a
timely manner even when a reliable transport such as TCP is used.
Similarly, the sender of a request (requester) SHOULD wait for a
sufficient time for a response before concluding that the responder
will not be acting upon its request.
A responder SHOULD respond to all requests within 5 seconds. If the
responder recognizes that the processing of a request will take
longer than 5 seconds, it SHOULD send a 100 (Continue) response as
soon as possible. It SHOULD continue sending a 100 response every 5
seconds thereafter until it is ready to send the final response to
the requester. After sending a 100 response, the responder MUST send
a final response indicating the success or failure of the request.
A requester SHOULD wait at least 10 seconds for a response before
concluding that the responder will not be responding to its request.
After receiving a 100 response, the requester SHOULD continue waiting
for further responses. If more than 10 seconds elapse without
receiving any response, the requester MAY assume that the responder
is unresponsive and abort the connection by closing the TCP
connection.
In some cases, multiple RTSP sessions share the same transport
connection; abandoning a request and closing the connection may have
significant impact on those other sessions. First of all, other RTSP
requests may have become queued up due to the request taking a long
time to process. Secondly, those sessions also lose the possibility
to receive server-to-client requests. To mitigate that situation,
the RTSP client or server SHOULD establish a new connection and send
any requests that are queued up or that haven't received a response
on this new connection. Thirdly, to ensure that the RTSP server
knows which connection is valid for a particular RTSP session, the
RTSP agent SHOULD send a keep-alive request, if no other request will
be sent immediately for that RTSP session, for each RTSP session on
the old connection. The keep-alive request will normally be a
SET_PARAMETER with a session header to inform the server that this
agent cares about this RTSP session.
A requester SHOULD wait longer than 10 seconds for a response if it
is experiencing significant transport delays on its connection to the
responder. The requester is capable of determining the Round-Trip
Time (RTT) of the request/response cycle using the Timestamp header
(Section 18.53) in any RTSP request.
The 10-second wait was chosen for the following reasons. It gives
TCP time to perform a couple of retransmissions, even if operating
on default values. It is short enough that users may not abandon
the process themselves. However, it should be noted that 10
seconds can be aggressive on certain types of networks. The
5-second value for 1xx messages is half the timeout giving a
reasonable chance of successful delivery before timeout happens on
the requester side.
10.5. Showing Liveness
RTSP requires the client to periodically show its liveness to the
server or the server may terminate any session state. Several
different protocol mechanism include in their usage a liveness proof
from the client. These mechanisms are RTSP requests with a Session
header to the server; if RTP & RTCP is used for media data transport
and the transport is established, the RTCP message proves liveness;
or through any other used media-transport protocol capable of
indicating liveness of the RTSP client. It is RECOMMENDED that a
client not wait to the last second of the timeout before trying to
send a liveness message. The RTSP message may take some time to
arrive safely at the receiver, due to packet loss and TCP
retransmissions. To show liveness between RTSP requests being issued
to accomplish other things, the following mechanisms can be used, in
descending order of preference:
RTCP: If RTP is used for media transport, RTCP SHOULD be used. If
RTCP is used to report transport statistics, it will
necessarily also function as a keep-alive. The server can
determine the client by network address and port together with
the fact that the client is reporting on the server's RTP
sender sources (synchronization source (SSRCs)). A downside of
using RTCP is that it only gives statistical guarantees of
reaching the server. However, the probability of a false
client timeout is so low that it can be ignored in most cases.
For example, assume a session with a 60-second timeout and
enough bitrate assigned to RTCP messages to send a message from
client to server on average every 5 seconds. That client has,
for a network with 5% packet loss, a probability of failing to
confirm liveness within the timeout interval for that session
of 2.4*E-16. Sessions with shorter timeouts, much higher
packet loss, or small RTCP bandwidths SHOULD also implement one
or more of the mechanisms below.
SET_PARAMETER: When using SET_PARAMETER for keep-alives, a body
SHOULD NOT be included. This method is the RECOMMENDED RTSP
method to use for a request intended only to perform keep-
alives. RTSP servers MUST support the SET_PARAMETER method, so
that clients can always use this mechanism.
GET_PARAMETER: When using GET_PARAMETER for keep-alives, a body
SHOULD NOT be included, dependent on implementation support in
the server. Use the OPTIONS method to determine if there is
method support or simply try.
OPTIONS: This method is also usable, but it causes the server to
perform more unnecessary processing and results in bigger
responses than necessary for the task. The reason is that the
server needs to determine the capabilities associated with the
media resource to correctly populate the Public and Allow
headers.
The timeout parameter of the Session header (Section 18.49) MAY be
included in a SETUP response and MUST NOT be included in requests.
The server uses it to indicate to the client how long the server is
prepared to wait between RTSP commands or other signs of life before
closing the session due to lack of activity (see Appendix B). The
timeout is measured in seconds, with a default of 60 seconds. The
length of the session timeout MUST NOT be changed in an established
session.
10.6. Use of IPv6
Explicit IPv6 [RFC2460] support was not present in RTSP 1.0. RTSP
2.0 has been updated for explicit IPv6 support. Implementations of
RTSP 2.0 MUST understand literal IPv6 addresses in URIs and RTSP
headers. Although the general URI format envisages potential future
new versions of the literal IP address, usage of any such new version
would require other modifications to the RTSP specification (e.g.,
address fields in the Transport header (Section 18.54)).
10.7. Overload Control
Overload in RTSP can occur when servers and proxies have insufficient
resources to complete the processing of a request. An improper
handling of such an overload situation at proxies and servers can
impact the operation of the RTSP deployment, and probably worsen the
situation. RTSP defines the 503 (Service Unavailable) response
(Section 17.5.4) to let servers and proxies notify requesting proxies
and RTSP clients about an overload situation. In conjunction with
the Retry-After header (Section 18.44), the server or proxy can
indicate the time after which the requesting entity can send another
request to the proxy or server.
There are two scopes of such 503 answers. The first scope is for an
established RTSP session, where the request resulting in the 503
response as well as the response itself carries a Session header
identifying the session that is suffering overload. This response
only applies to this particular session. The other scope is the
general RTSP server as identified by the host in the Request-URI.
Such a 503 answer with any Retry-After header applies to all requests
that are not session specific to that server, including a SETUP
request intended to create a new RTSP session.
Another scope for overload situations exists: the RTSP proxy. To
enable an RTSP proxy to signal that it is overloaded, or otherwise
unavailable and unable to handle the request, a 553 response code has
been defined with the meaning "Proxy Unavailable". As with servers,
there is a separation in response scopes between requests associated
with existing RTSP sessions and requests to create new sessions or
general proxy requests.
Simply implementing and using the 503 (Service Unavailable) and 553
(Proxy Unavailable) response codes is not sufficient for properly
handling overload situations. For instance, a simplistic approach
would be to send the 503 response with a Retry-After header set to a
fixed value. However, this can cause a situation in which multiple
RTSP clients again send requests to a proxy or server at roughly the
same time, which may again cause an overload situation. Another
situation would be if the "old" overload situation is not yet
resolved, i.e., the length indicated in the Retry-After header was
too short for the overload situation to subside.
An RTSP server or proxy in an overload situation must select the
value of the Retry-After header carefully, bearing in mind its
current load situation. It is REQUIRED to increase the timeout
period in proportion to the current load on the server, i.e., an
increasing workload should result in an increased length of the
indicated unavailability. It is REQUIRED not to send the same value
in the Retry-After header to all requesting proxies and clients, but
to add a variation to the mean value of the Retry-After header.
A more complex case may arise when a load-balancing RTSP proxy is in
use. This is the case when an RTSP proxy is used to select amongst a
set of RTSP servers to handle the requests or when multiple server
addresses are available for a given server name. The proxy or client
may receive a 503 (Service Unavailable) or 553 (Proxy Unavailable)
response code from one of its RTSP servers or proxies, or a TCP
timeout (if the server is even unable to handle the request message).
The proxy or client simply retries the other addresses or configured
proxies, but it may also receive a 503 (Service Unavailable) or 553
(Proxy Unavailable) response or TCP timeouts from those addresses.
In such a situation, where none of the RTSP servers/proxies/addresses
can handle the request, the RTSP agent has to wait before it can send
any new requests to the RTSP server. Any additional request to a
specific address MUST be delayed according to the Retry-After headers
received. For addresses where no response was received or TCP
timeout occurred, an initial wait timer SHOULD be set to 5 seconds.
That timer MUST be doubled for each additional failure to connect or
receive response until the value exceeds 30 minutes when the timer's
mean value may be set to 30 minutes. It is REQUIRED not to set the
same value in the timer for each scheduling, but instead to add a
variation to the mean value, resulting in picking a random value
within the range of 0.5 to 1.5 times the mean value.
11. Capability Handling
This section describes the available capability-handling mechanism
that allows RTSP to be extended. Extensions to this version of the
protocol are basically done in two ways. Firstly, new headers can be
added. Secondly, new methods can be added. The capability-handling
mechanism is designed to handle both cases.
When a method is added, the involved parties can use the OPTIONS
method to discover whether it is supported. This is done by issuing
an OPTIONS request to the other party. Depending on the URI, it will
either apply in regard to a certain media resource, the whole server
in general, or simply the next hop. The OPTIONS response MUST
contain a Public header that declares all methods supported for the
indicated resource.
It is not necessary to use OPTIONS to discover support of a method,
as the client could simply try the method. If the receiver of the
request does not support the method, it will respond with an error
code indicating the method is either not implemented (501) or does
not apply for the resource (405). The choice between the two
discovery methods depends on the requirements of the service.
Feature tags are defined to handle functionality additions that are
not new methods. Each feature tag represents a certain block of
functionality. The amount of functionality that a feature tag
represents can vary significantly. For example, a feature tag can
represent the functionality a single RTSP header provides. Another
feature tag can represent much more functionality, such as the
"play.basic" feature tag (Section 11.1), which represents the minimal
media delivery for playback implementation.
Feature tags are used to determine whether the client, server, or
proxy supports the functionality that is necessary to achieve the
desired service. To determine support of a feature tag, several
different headers can be used, each explained below:
Supported: This header is used to determine the complete set of
functionality that both client and server have, in general, and
is not dependent on a specific resource. The intended usage is
to determine before one needs to use a functionality that it is
supported. It can be used in any method, but OPTIONS is the
most suitable as it simultaneously determines all methods that
are implemented. When sending a request, the requester
declares all its capabilities by including all supported
feature tags. This results in the receiver learning the
requester's feature support. The receiver then includes its
set of features in the response.
Proxy-Supported: This header is used in a similar fashion as the
Supported header, but instead of giving the supported
functionality of the client or server, it provides both the
requester and the responder a view of the common functionality
supported in general by all members of the proxy chain between
the client and server; it does not depend on the resource.
Proxies are required to add this header whenever the Supported
header is present, but proxies may also add it independently of
the requester.
Require: This header can be included in any request where the
endpoint, i.e., the client or server, is required to understand
the feature to correctly perform the request. This can, for
example, be a SETUP request, where the server is required to
understand a certain parameter to be able to set up the media
delivery correctly. Ignoring this parameter would not have the
desired effect and is not acceptable. Therefore, the endpoint
receiving a request containing a Require MUST negatively
acknowledge any feature that it does not understand and not
perform the request. The response in cases where features are
not supported is 551 (Option Not Supported). Also, the
features that are not supported are given in the Unsupported
header in the response.
Proxy-Require: This header has the same purpose and behavior as
Require except that it only applies to proxies and not the
endpoint. Features that need to be supported by both proxies
and endpoints need to be included in both the Require and
Proxy-Require header.
Unsupported: This header is used in a 551 (Option Not Supported)
error response, to indicate which features were not supported.
Such a response is only the result of the usage of the Require
or Proxy-Require headers where one or more features were not
supported. This information allows the requester to make the
best of situations as it knows which features are not
supported.
11.1. Feature Tag: play.basic
An implementation supporting all normative parts of this
specification for the setup and control of playback of media uses the
feature tag "play.basic" to indicate this support. The appendices
(starting with letters) are not part of the functionality included in
the feature tag unless the appendix is explicitly specified in a main
section as being a required appendix.
Note: This feature tag does not mandate any media delivery
protocol, such as RTP.
In RTSP 1.0, there was a minimal implementation section. However,
that was not consistent with the rest of the specification. So,
rather than making an attempt to explicitly enumerate the features
for play.basic, this specification has to be taken as a whole and
the necessary features normatively defined as being required are
included.
12. Pipelining Support
Pipelining is a general method to improve performance of request/
response protocols by allowing the requesting agent to have more than
one request outstanding and to send them over the same persistent
connection. For RTSP, where the relative order of requests will
matter, it is important to maintain the order of the requests.
Because of this, the responding agent MUST process the incoming
requests in their sending order. The sending order can be determined
by the CSeq header and its sequence number. For TCP, the delivery
order will be the same, between two agents, as the sending order.
The processing of the request MUST also have been finished before
processing the next request from the same agent. The responses MUST
be sent in the order the requests were processed.
RTSP 2.0 has extended support for pipelining beyond the capabilities
in RTSP 1.0. As a major improvement, all requests involved in
setting up and initiating media delivery can now be pipelined,
indicated by the Pipelined-Request header (see Section 18.33). This
header allows a client to request that two or more requests be
processed in the same RTSP session context that the first request
creates. In other words, a client can request that two or more media
streams be set up and then played without needing to wait for a
single response. This speeds up the initial start-up time for an
RTSP session by at least one RTT.
If a pipelined request builds on the successful completion of one or
more prior requests, the requester must verify that all requests were
executed as expected. A common example will be two SETUP requests
and a PLAY request. In case one of the SETUP requests fails
unexpectedly, the PLAY request can still be successfully executed.
However, the resulting presentation will not be as expected by the
requesting client, as only a single media instead of two will be
played. In this case, the client can send a PAUSE request, correct
the failing SETUP request, and then request it be played.
13. Method Definitions
The method indicates what is to be performed on the resource
identified by the Request-URI. The method name is case sensitive.
New methods may be defined in the future. Method names MUST NOT
start with a $ character (decimal 36) and MUST be a token as defined
by the ABNF [RFC5234] in Section 20. The methods are summarized in
Table 7.
+---------------+-----------+--------+-------------+-------------+
| method | direction | object | Server req. | Client req. |
+---------------+-----------+--------+-------------+-------------+
| DESCRIBE | C -> S | P,S | recommended | recommended |
| | | | | |
| GET_PARAMETER | C -> S | P,S | optional | optional |
| | | | | |
| | S -> C | P,S | optional | optional |
| | | | | |
| OPTIONS | C -> S | P,S | required | required |
| | | | | |
| | S -> C | P,S | optional | optional |
| | | | | |
| PAUSE | C -> S | P,S | required | required |
| | | | | |
| PLAY | C -> S | P,S | required | required |
| | | | | |
| PLAY_NOTIFY | S -> C | P,S | required | required |
| | | | | |
| REDIRECT | S -> C | P,S | optional | required |
| | | | | |
| SETUP | C -> S | S | required | required |
| | | | | |
| SET_PARAMETER | C -> S | P,S | required | optional |
| | | | | |
| | S -> C | P,S | optional | optional |
| | | | | |
| TEARDOWN | C -> S | P,S | required | required |
| | | | | |
| | S -> C | P | required | required |
+---------------+-----------+--------+-------------+-------------+
Table 7: Overview of RTSP Methods
Note on Table 7: This table covers RTSP methods, their direction,
and on what objects (P: presentation, S: stream) they operate.
Further, it indicates whether a server or a client implementation
is required (mandatory), recommended, or optional.
Further note on Table 7: the GET_PARAMETER is optional. For
example, a fully functional server can be built to deliver media
without any parameters. However, SET_PARAMETER is required, i.e.,
mandatory to implement for the server; this is due to its usage
for keep-alive. PAUSE is required because it is the only way of
leaving the Play state without terminating the whole session.
If an RTSP agent does not support a particular method, it MUST return
a 501 (Not Implemented) response code and the requesting RTSP agent,
in turn, SHOULD NOT try this method again for the given agent/
resource combination. An RTSP proxy whose main function is to log or
audit and not modify transport or media handling in any way MAY
forward RTSP messages with unknown methods. Note that the proxy
still needs to perform the minimal required processing, like adding
the Via header.
13.1. OPTIONS
The semantics of the RTSP OPTIONS method is similar to that of the
HTTP OPTIONS method described in Section 4.3.7 of [RFC7231].
However, in RTSP, OPTIONS is bidirectional in that a client can send
the request to a server and vice versa. A client MUST implement the
capability to send an OPTIONS request and a server or a proxy MUST
implement the capability to respond to an OPTIONS request. In
addition to this "MUST-implement" functionality, clients, servers and
proxies MAY provide support both for sending OPTIONS requests and for
generating responses to the requests.
An OPTIONS request may be issued at any time. Such a request does
not modify the session state. However, it may prolong the session
lifespan (see below). The URI in an OPTIONS request determines the
scope of the request and the corresponding response. If the Request-
URI refers to a specific media resource on a given host, the scope is
limited to the set of methods supported for that media resource by
the indicated RTSP agent. A Request-URI with only the host address
limits the scope to the specified RTSP agent's general capabilities
without regard to any specific media. If the Request-URI is an
asterisk ("*"), the scope is limited to the general capabilities of
the next hop (i.e., the RTSP agent in direct communication with the
request sender).
Regardless of the scope of the request, the Public header MUST always
be included in the OPTIONS response, listing the methods that are
supported by the responding RTSP agent. In addition, if the scope of
the request is limited to a media resource, the Allow header MUST be
included in the response to enumerate the set of methods that are
allowed for that resource unless the set of methods completely
matches the set in the Public header. If the given resource is not
available, the RTSP agent SHOULD return an appropriate response code,
such as 3rr or 4xx. The Supported header MAY be included in the
request to query the set of features that are supported by the
responding RTSP agent.
The OPTIONS method can be used to keep an RTSP session alive.
However, this is not the preferred way of session keep-alive
signaling; see Section 18.49. An OPTIONS request intended for
keeping alive an RTSP session MUST include the Session header with
the associated session identifier. Such a request SHOULD also use
the media or the aggregated control URI as the Request-URI.
Example:
C->S: OPTIONS rtsp://server.example.com RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
Proxy-Require: gzipped-messages
Supported: play.basic
S->C: RTSP/2.0 200 OK
CSeq: 1
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, OPTIONS
Supported: play.basic, setup.rtp.rtcp.mux, play.scale
Server: PhonyServer/1.1
Note that the "gzipped-messages" feature tag in the Proxy-Require is
a fictitious feature.
13.2. DESCRIBE
The DESCRIBE method is used to retrieve the description of a
presentation or media object from a server. The Request-URI of the
DESCRIBE request identifies the media resource of interest. The
client MAY include the Accept header in the request to list the
description formats that it understands. The server MUST respond
with a description of the requested resource and return the
description in the message body of the response, if the DESCRIBE
method request can be successfully fulfilled. The DESCRIBE reply-
response pair constitutes the media initialization phase of RTSP.
The DESCRIBE response SHOULD contain all media initialization
information for the resource(s) that it describes. Servers SHOULD
NOT use the DESCRIBE response as a means of media indirection by
having the description point at another server; instead, using the
3rr responses is RECOMMENDED.
By forcing a DESCRIBE response to contain all media initialization
information for the set of streams that it describes, and
discouraging the use of DESCRIBE for media indirection, any
looping problems can be avoided that might have resulted from
other approaches.
Example:
C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/2.0
CSeq: 312
User-Agent: PhonyClient/1.2
Accept: application/sdp, application/example
S->C: RTSP/2.0 200 OK
CSeq: 312
Date: Thu, 23 Jan 1997 15:35:06 GMT
Server: PhonyServer/1.1
Content-Base: rtsp://server.example.com/fizzle/foo/
Content-Type: application/sdp
Content-Length: 358
v=0
o=MNobody 2890844526 2890842807 IN IP4 192.0.2.46
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.example.com/lectures/sdp.ps
e=seminar@example.com (Seminar Management)
c=IN IP4 0.0.0.0
a=control:*
t=2873397496 2873404696
m=audio 3456 RTP/AVP 0
a=control:audio
m=video 2232 RTP/AVP 31
a=control:video
Media initialization is a requirement for any RTSP-based system, but
the RTSP specification does not dictate that this is required to be
done via the DESCRIBE method. There are three ways that an RTSP
client may receive initialization information:
o via an RTSP DESCRIBE request
o via some other protocol (HTTP, email attachment, etc.)
o via some form of user interface
If a client obtains a valid description from an alternate source, the
client MAY use this description for initialization purposes without
issuing a DESCRIBE request for the same media. The client should use
any MTag to either validate the presentation description or make the
session establishment conditional on being valid.
It is RECOMMENDED that minimal servers support the DESCRIBE method,
and highly recommended that minimal clients support the ability to
act as "helper applications" that accept a media initialization file
from a user interface, or other means that are appropriate to the
operating environment of the clients.
13.3. SETUP
The description below uses the following states in a protocol state
machine that is related to a specific session when that session has
been created. The state transitions are driven by protocol
interactions. For additional information about the state machine,
see Appendix B.
Init: Initial state. No session exists.
Ready: Session is ready to start playing.
Play: Session is playing, i.e., sending media-stream data in the
direction S->C.
The SETUP request for a URI specifies the transport mechanism to be
used for the streamed media. The SETUP method may be used in two
different cases, namely, creating an RTSP session and changing the
transport parameters of media streams that are already set up. SETUP
can be used in all three states, Init, Ready, and Play, to change the
transport parameters. Additionally, Init and Ready can also be used
for the creation of the RTSP session. The usage of the SETUP method
in the Play state to add a media resource to the session is
unspecified.
The Transport header, see Section 18.54, specifies the media-
transport parameters acceptable to the client for data transmission;
the response will contain the transport parameters selected by the
server. This allows the client to enumerate, in descending order of
preference, the transport mechanisms and parameters acceptable to it,
so the server can select the most appropriate. It is expected that
the session description format used will enable the client to select
a limited number of possible configurations that are offered as
choices to the server. All transport-related parameters SHALL be
included in the Transport header; the use of other headers for this
purpose is NOT RECOMMENDED due to middleboxes, such as firewalls or
NATs.
For the benefit of any intervening firewalls, a client MUST indicate
the known transport parameters, even if it has no influence over
these parameters, for example, where the server advertises a fixed-
multicast address as destination.
Since SETUP includes all transport initialization information,
firewalls and other intermediate network devices (which need this
information) are spared the more arduous task of parsing the
DESCRIBE response, which has been reserved for media
initialization.
The client MUST include the Accept-Ranges header in the request,
indicating all supported unit formats in the Range header. This
allows the server to know which formats it may use in future session-
related responses, such as a PLAY response without any range in the
request. If the client does not support a time format necessary for
the presentation, the server MUST respond using 456 (Header Field Not
Valid for Resource) and include the Accept-Ranges header with the
range unit formats supported for the resource.
In a SETUP response, the server MUST include the Accept-Ranges header
(see Section 18.5) to indicate which time formats are acceptable to
use for this media resource.
The SETUP 200 OK response MUST include the Media-Properties header
(see Section 18.29). The combination of the parameters of the Media-
Properties header indicates the nature of the content present in the
session (see also Section 4.7). For example, a live stream with time
shifting is indicated by
o Random access set to Random-Access,
o Content Modifications set to Time-Progressing, and
o Retention set to Time-Duration (with specific recording window
time value).
The SETUP 200 OK response MUST include the Media-Range header (see
Section 18.30) if the media is Time-Progressing.
A basic example for SETUP:
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0
CSeq: 302
Transport: RTP/AVP;unicast;dest_addr=":4588"/":4589",
RTP/AVP/TCP;unicast;interleaved=0-1
Accept-Ranges: npt, clock
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 302
Date: Thu, 23 Jan 1997 15:35:06 GMT
Server: PhonyServer/1.1
Session: QKyjN8nt2WqbWw4tIYof52;timeout=60
Transport: RTP/AVP;unicast;dest_addr="192.0.2.53:4588"/
"192.0.2.53:4589"; src_addr="198.51.100.241:6256"/
"198.51.100.241:6257"; ssrc=2A3F93ED
Accept-Ranges: npt
Media-Properties: Random-Access=3.2, Time-Progressing,
Time-Duration=3600.0
Media-Range: npt=0-2893.23
In the above example, the client wants to create an RTSP session
containing the media resource "rtsp://example.com/foo/bar/baz.rm".
The transport parameters acceptable to the client are either RTP/AVP/
UDP (UDP per default) to be received on client port 4588 and 4589 at
the address the RTSP setup connection comes from or RTP/AVP
interleaved on the RTSP control channel. The server selects the
RTP/AVP/UDP transport and adds the address and ports it will send and
receive RTP and RTCP from, and the RTP SSRC that will be used by the
server.
The server MUST generate a session identifier in response to a
successful SETUP request unless a SETUP request to a server includes
a session identifier or a Pipelined-Requests header referencing an
existing session context. In that latter case, the server MUST
bundle this SETUP request into the existing session (aggregated
session) or return a 459 (Aggregate Operation Not Allowed) error code
(see Section 17.4.23). An aggregate control URI MUST be used to
control an aggregated session. This URI MUST be different from the
stream control URIs of the individual media streams included in the
aggregate (see Section 13.4.2 for aggregated sessions and for the
particular URIs see Appendix D.1.1). The aggregate control URI is to
be specified by the session description if the server supports
aggregated control and aggregated control is desired for the session.
However, even if aggregated control is offered, the client MAY choose
not to set up the session in aggregated control. If an aggregate
control URI is not specified in the session description, it is
normally an indication that non-aggregated control should be used.
The SETUP of media streams in an aggregate that has not been given an
aggregated control URI is unspecified.
While the session ID sometimes carries enough information for
aggregate control of a session, the aggregate control URI is still
important for some methods such as SET_PARAMETER where the control
URI enables the resource in question to be easily identified. The
aggregate control URI is also useful for proxies, enabling them to
route the request to the appropriate server, and for logging,
where it is useful to note the actual resource on which a request
was operating.
A session will exist until it is either removed by a TEARDOWN request
or is timed out by the server. The server MAY remove a session that
has not demonstrated liveness signs from the client(s) within a
certain timeout period. The default timeout value is 60 seconds; the
server MAY set this to a different value and indicate so in the
timeout field of the Session header in the SETUP response. For
further discussion, see Section 18.49. Signs of liveness for an RTSP
session include any RTSP requests from a client that contain a
Session header with the ID for that session, as well as RTCP sender
or receiver reports if RTP is used to transport the underlying media
stream. RTCP sender reports may, for example, be received in session
where the server is invited into a conference session and are thus
valid as a liveness indicator.
If a SETUP request on a session fails for any reason, the session
state, as well as transport and other parameters for associated
streams, MUST remain unchanged from their values as if the SETUP
request had never been received by the server.
13.3.1. Changing Transport Parameters
A client MAY issue a SETUP request for a stream that is already set
up or playing in the session to change transport parameters, which a
server MAY allow. If it does not allow the changing of parameters,
it MUST respond with error 455 (Method Not Valid in This State). The
reasons to support changing transport parameters include allowing
application-layer mobility and flexibility to utilize the best
available transport as it becomes available. If a client receives a
455 error when trying to change transport parameters while the server
is in Play state, it MAY try to put the server in Ready state using
PAUSE before issuing the SETUP request again. If that also fails,
the changing of transport parameters will require that the client
perform a TEARDOWN of the affected media and then set it up again.
For an aggregated session, not tearing down all the media at the same
time will avoid the creation of a new session.
All transport parameters MAY be changed. However, the primary usage
expected is to either change the transport protocol completely, like
switching from Interleaved TCP mode to UDP or vice versa, or to
change the delivery address.
In a SETUP response for a request to change the transport parameters
while in Play state, the server MUST include the Range header to
indicate at what point the new transport parameters will be used.
Further, if RTP is used for delivery, the server MUST also include
the RTP-Info header to indicate at what timestamp and RTP sequence
number the change will take place. If both RTP-Info and Range are
included in the response, the "rtp_time" parameter and start point in
the Range header MUST be for the corresponding time, i.e., be used in
the same way as for PLAY to ensure the correct synchronization
information is available.
If the transport-parameters change that happened while in Play state
results in a change of synchronization-related information, for
example, changing RTP SSRC, the server MUST include the necessary
synchronization information in the SETUP response. However, the
server SHOULD avoid changing the synchronization information if
possible.
13.4. PLAY
This section describes the usage of the PLAY method in general, for
aggregated sessions, and in different usage scenarios.
13.4.1. General Usage
The PLAY method tells the server to start sending data via the
mechanism specified in SETUP and which part of the media should be
played out. PLAY requests are valid when the session is in Ready or
Play state. A PLAY request MUST include a Session header to indicate
to which session the request applies.
Upon receipt of the PLAY request, the server MUST position the normal
play time to the beginning of the range specified in the received
Range header, within the limits of the media resource and in
accordance with the Seek-Style header (Section 18.47). It MUST
deliver stream data until the end of the range if given, until a new
PLAY request is received, until a PAUSE request (Section 13.5) is
received, or until the end of the media is reached. If no Range
header is present in the PLAY request, the server SHALL play from
current pause point until the end of media. The pause point defaults
at session start to the beginning of the media. For media that is
time-progressing and has no retention, the pause point will always be
set equal to NPT "now", i.e., the current delivery point. The pause
point may also be set to a particular point in the media by the PAUSE
method; see Section 13.6. The pause point for media that is
currently playing is equal to the current media position. For time-
progressing media with time-limited retention, if the pause point
represents a position that is older than what is retained by the
server, the pause point will be moved to the oldest retained
position.
What range values are valid depends on the type of content. For
content that isn't time-progressing, the range value is valid if the
given range is part of any media within the aggregate. In other
words, the valid media range for the aggregate is the union of all of
the media components in the aggregate. If a given range value points
outside of the media, the response MUST be the 457 (Invalid Range)
error code and include the Media-Range header (Section 18.30) with
the valid range for the media. Except for time-progressing content
where the client requests a start point prior to what is retained,
the start point is adjusted to the oldest retained content. For a
start point that is beyond the media front edge, i.e., beyond the
current value for "now", the server SHALL adjust the start value to
the current front edge. The Range header's stop point value may
point beyond the current media edge. In that case, the server SHALL
deliver media from the requested (and possibly adjusted) start point
until the first of either the provided stop point or the end of the
media. Please note that if one simply wants to play from a
particular start point until the end of media, using a Range header
with an implicit stop point is RECOMMENDED.
If a client requests to start playing at the end of media, either
explicitly with a Range header or implicitly with a pause point that
is at the end of media, a 457 (Invalid Range) error MUST be sent and
include the Media-Range header (Section 18.30). It is specified
below that the Range header also must be included in the response and
that it will carry the pause point in the media, in the case of the
session being in Ready State. Note that this also applies if the
pause point or requested start point is at the beginning of the media
and a Scale header (Section 18.46) is included with a negative value
(playing backwards).
For media with random access properties, a client may indicate which
policy for start point selection the server should use. This is done
by including the Seek-Style header (Section 18.47) in the PLAY
request. The Seek-Style applied will affect the content of the Range
header as it will be adjusted to indicate from what point the media
actually is delivered.
A client desiring to play the media from the beginning MUST send a
PLAY request with a Range header pointing at the beginning, e.g.,
"npt=0-". If a PLAY request is received without a Range header and
media delivery has stopped at the end, the server SHOULD respond with
a 457 (Invalid Range) error response. In that response, the current
pause point MUST be included in a Range header.
All range specifiers in this specification allow for ranges with an
implicit start point (e.g., "npt=-30"). When used in a PLAY request,
the server treats this as a request to start or resume delivery from
the current pause point, ending at the end time specified in the
Range header. If the pause point is located later than the given end
value, a 457 (Invalid Range) response MUST be returned.
The example below will play seconds 10 through 25. It also requests
that the server deliver media from the first random access point
prior to the indicated start point.
C->S: PLAY rtsp://audio.example.com/audio RTSP/2.0
CSeq: 835
Session: ULExwZCXh2pd0xuFgkgZJW
Range: npt=10-25
Seek-Style: RAP
User-Agent: PhonyClient/1.2
Servers MUST include a Range header in any PLAY response, even if no
Range header was present in the request. The response MUST use the
same format as the request's Range header contained. If no Range
header was in the request, the format used in any previous PLAY
request within the session SHOULD be used. If no format has been
indicated in a previous request, the server MAY use any time format
supported by the media and indicated in the Accept-Ranges header in
the SETUP request. It is RECOMMENDED that NPT is used if supported
by the media.
For any error response to a PLAY request, the server's response
depends on the current session state. If the session is in Ready
state, the current pause point is returned using a Range header with
the pause point as the explicit start point and an implicit stop
point. For time-progressing content, where the pause-point moves
with real-time due to limited retention, the current pause point is
returned. For sessions in Play state, the current playout point and
the remaining parts of the range request are returned. For any media
with retention longer than 0 seconds, the currently valid Media-Range
header SHALL also be included in the response.
A PLAY response MAY include a header carrying synchronization
information. As the information necessary is dependent on the media-
transport format, further rules specifying the header and its usage
are needed. For RTP the RTP-Info header is specified, see
Section 18.45, and used in the following example.
Here is a simple example for a single audio stream where the client
requests the media starting from 3.52 seconds and to the end. The
server sends a 200 OK response with the actual play time, which is 10
ms prior (3.51), and the RTP-Info header that contains the necessary
parameters for the RTP stack.
C->S: PLAY rtsp://example.com/audio RTSP/2.0
CSeq: 836
Session: ULExwZCXh2pd0xuFgkgZJW
Range: npt=3.52-
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 836
Date: Thu, 23 Jan 1997 15:35:06 GMT
Server: PhonyServer/1.0
Range: npt=3.51-324.39
Seek-Style: First-Prior
Session: ULExwZCXh2pd0xuFgkgZJW
RTP-Info:url="rtsp://example.com/audio"
ssrc=0D12F123:seq=14783;rtptime=2345962545
S->C: RTP Packet TS=2345962545 => NPT=3.51
Media duration=0.16 seconds
The server replies with the actual start point that will be
delivered. This may differ from the requested range if alignment of
the requested range to valid frame boundaries is required for the
media source. Note that some media streams in an aggregate may need
to be delivered from even earlier points. Also, some media formats
have a very long duration per individual data unit; therefore, it
might be necessary for the client to parse the data unit, and select
where to start. The server SHALL also indicate which policy it uses
for selecting the actual start point by including a Seek-Style
header.
In the following example, the client receives the first media packet
that stretches all the way up and past the requested playtime. Thus,
it is the client's decision whether to render to the user the time
between 3.52 and 7.05 or to skip it. In most cases, it is probably
most suitable not to render that time period.
C->S: PLAY rtsp://example.com/audio RTSP/2.0
CSeq: 836
Session: ZGGyCJOs8xaLkdNK2dmxQO
Range: npt=7.05-
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 836
Date: Thu, 23 Jan 1997 15:35:06 GMT
Server: PhonyServer/1.0
Session: ZGGyCJOs8xaLkdNK2dmxQO
Range: npt=3.52-
Seek-Style: First-Prior
RTP-Info:url="rtsp://example.com/audio"
ssrc=0D12F123:seq=14783;rtptime=2345962545
S->C: RTP Packet TS=2345962545 => NPT=3.52
Duration=4.15 seconds
After playing the desired range, the presentation does NOT change to
the Ready state, media delivery simply stops. If it is necessary to
put the stream into the Ready state, a PAUSE request MUST be issued.
A PLAY request while the stream is still in the Play state is legal
and can be issued without an intervening PAUSE request. Such a
request MUST replace the current PLAY action with the new one
requested, i.e., being handled in the same way as if as the request
was received in Ready state. In the case that the range in the Range
header has an implicit start time ("-endtime"), the server MUST
continue to play from where it currently was until the specified
endpoint. This is useful to change the end to at another point than
in the previous request.
The following example plays the whole presentation starting at SMPTE
time code 0:10:20 until the end of the clip. Note: the RTP-Info
headers have been broken into several lines, where subsequent lines
start with whitespace as allowed by the syntax.
C->S: PLAY rtsp://audio.example.com/twister.en RTSP/2.0
CSeq: 833
Session: N465Wvsv0cjUy6tLqINkcf
Range: smpte=0:10:20-
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 833
Date: Thu, 23 Jan 1997 15:35:06 GMT
Session: N465Wvsv0cjUy6tLqINkcf
Server: PhonyServer/1.0
Range: smpte=0:10:22-0:15:45
Seek-Style: Next
RTP-Info:url="rtsp://example.com/twister.en"
ssrc=0D12F123:seq=14783;rtptime=2345962545
For playing back a recording of a live presentation, it may be
desirable to use clock units:
C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/2.0
CSeq: 835
Session: N465Wvsv0cjUy6tLqINkcf
Range: clock=19961108T142300Z-19961108T143520Z
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 835
Date: Thu, 23 Jan 1997 15:35:06 GMT
Session: N465Wvsv0cjUy6tLqINkcf
Server: PhonyServer/1.0
Range: clock=19961108T142300Z-19961108T143520Z
Seek-Style: Next
RTP-Info:url="rtsp://example.com/meeting.en"
ssrc=0D12F123:seq=53745;rtptime=484589019
13.4.2. Aggregated Sessions
PLAY requests can operate on sessions controlling a single media
stream and on aggregated sessions controlling multiple media streams.
In an aggregated session, the PLAY request MUST contain an aggregated
control URI. A server MUST respond with a 460 error (Only Aggregate
Operation Allowed) if the client PLAY Request-URI is for a single
media. The media in an aggregate MUST be played in sync. If a
client wants individual control of the media, it needs to use
separate RTSP sessions for each media.
For aggregated sessions where the initial SETUP request (creating a
session) is followed by one or more additional SETUP requests, a PLAY
request MAY be pipelined (Section 12) after those additional SETUP
requests without awaiting their responses. This procedure can reduce
the delay from the start of session establishment until media playout
has started with one RTT. However, a client needs to be aware that
using this procedure will result in the playout of the server state
established at the time of processing the PLAY, i.e., after the
processing of all the requests prior to the PLAY request in the
pipeline. This state may not be the intended one due to failure of
any of the prior requests. A client can easily determine this based
on the responses from those requests. In case of failure, the client
can halt the media playout using PAUSE and try to establish the
intended state again before issuing another PLAY request.
13.4.3. Updating Current PLAY Requests
Clients can issue PLAY requests while the stream is in Play state and
thus updating their request.
The important difference compared to a PLAY request in Ready state is
the handling of the current play point and how the Range header in
the request is constructed. The session is actively playing media
and the play point will be moving, making the exact time a request
will take effect hard to predict. Depending on how the PLAY header
appears, two different cases exist: total replacement or
continuation. A total replacement is signaled by having the first
range specification have an explicit start value, e.g., "npt=45-" or
"npt=45-60", in which case the server stops playout at the current
playout point and then starts delivering media according to the Range
header. This is equivalent to having the client first send a PAUSE
and then a new PLAY request that isn't based on the pause point. In
the case of continuation, the first range specifier has an implicit
start point and an explicit stop value (Z), e.g., "npt=-60", which
indicate that it MUST convert the range specifier being played prior
to this PLAY request (X to Y) into (X to Z) and continue as if this
was the request originally played. If the current delivery point is
beyond the stop point, the server SHALL immediately pause delivery.
As the request has been completed successfully, it shall be responded
to with a 200 OK response. A PLAY_NOTIFY with end-of-stream is also
sent to indicate the actual stop point. The pause point is set to
the requested stop point.
The following is an example of this behavior: The server has received
requests to play ranges 10 to 15. If the new PLAY request arrives at
the server 4 seconds after the previous one, it will take effect
while the server still plays the first range (10-15). The server
changes the current play to continue to 25 seconds, i.e., the
equivalent single request would be PLAY with "range: npt=10-25".
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 834
Session: apzA8LnjQ5KWTdw0kUkiRh
Range: npt=10-15
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 834
Date: Thu, 23 Jan 1997 15:35:06 GMT
Session: apzA8LnjQ5KWTdw0kUkiRh
Server: PhonyServer/1.0
Range: npt=10-15
Seek-Style: Next
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934207921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=789DAF12:seq=57654;rtptime=2792482193
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 835
Session: apzA8LnjQ5KWTdw0kUkiRh
Range: npt=-25
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 835
Date: Thu, 23 Jan 1997 15:35:09 GMT
Session: apzA8LnjQ5KWTdw0kUkiRh
Server: PhonyServer/1.0
Range: npt=14-25
Seek-Style: Next
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934239921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=789DAF12:seq=57654;rtptime=2792842193
A common use of a PLAY request while in Play state is changing the
scale of the media, i.e., entering or leaving fast forward or fast
rewind. The client can issue an updating PLAY request that is either
a continuation or a complete replacement, as discussed above this
section. Below is an example of a client that is requesting a fast
forward (scale = 2) without giving a stop point and then a change
from fast forward to regular playout (scale = 1). In the second PLAY
request, the time is set explicitly to be wherever the server
currently plays out (npt=now-) and the server responds with the
actual playback point where the new scale actually takes effect
(npt=02:17:27.144-).
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 2034
Session: apzA8LnjQ5KWTdw0kUkiRh
Range: npt=now-
Scale: 2.0
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 2034
Date: Thu, 23 Jan 1997 15:35:06 GMT
Session: apzA8LnjQ5KWTdw0kUkiRh
Server: PhonyServer/1.0
Range: npt=02:17:21.394-
Seek-Style: Next
Scale: 2.0
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934207921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=789DAF12:seq=57654;rtptime=2792482193
[playing in fast forward and now returning to scale = 1]
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 2035
Session: apzA8LnjQ5KWTdw0kUkiRh
Range: npt=now-
Scale: 1.0
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 2035
Date: Thu, 23 Jan 1997 15:35:09 GMT
Session: apzA8LnjQ5KWTdw0kUkiRh
Server: PhonyServer/1.0
Range: npt=02:17:27.144-
Seek-Style: Next
Scale: 1.0
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934239921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=789DAF12:seq=57654;rtptime=2792842193
13.4.4. Playing On-Demand Media
On-demand media is indicated by the content of the Media-Properties
header in the SETUP response when (see also Section 18.29):
o the Random Access property is set to Random-Access;
o the Content Modifications property is set to Immutable;
o the Retention property is set to Unlimited or Time-Limited.
Playing on-demand media follows the general usage as described in
Section 13.4.1.
13.4.5. Playing Dynamic On-Demand Media
Dynamic on-demand media is indicated by the content of the Media-
Properties header in the SETUP response when (see also
Section 18.29):
o the Random Access property is set to Random-Access;
o the Content Modifications property is set to Dynamic;
o the Retention property is set to Unlimited or Time-Limited.
Playing on-demand media follows the general usage as described in
Section 13.4.1 as long as the media has not been changed.
There are two ways for the client to be informed about changes of
media resources in Play state. The first being that the client will
receive a PLAY_NOTIFY request with the Notify-Reason header set to
media-properties-update (see Section 13.5.2). The client can use the
value of the Media-Range header to decide further actions, if the
Media-Range header is present in the PLAY_NOTIFY request. The second
way is that the client issues a GET_PARAMETER request without a body
but including a Media-Range header. The 200 OK response MUST include
the current Media-Range header (see Section 18.30).
13.4.6. Playing Live Media
Live media is indicated by the content of the Media-Properties header
in the SETUP response when (see also Section 18.29):
o the Random Access property is set to No-Seeking;
o the Content Modifications property is set to Time-Progressing;
o the Retention property's Time-Duration is set to 0.0.
For live media, the SETUP 200 OK response MUST include the Media-
Range header (see Section 18.30).
A client MAY send PLAY requests without the Range header. If the
request includes the Range header, it MUST use a symbolic value
representing "now". For NPT, that range specification is "npt=now-".
The server MUST include the Range header in the response, and it MUST
indicate an explicit time value and not a symbolic value. In other
words, "npt=now-" cannot be used in the response. Instead, the time
since session start is recommended, expressed as an open interval,
e.g., "npt=96.23-". An absolute time value (clock) for the
corresponding time MAY be given, i.e., "clock=20030213T143205Z-".
The Absolute Time format can only be used if the client has shown
support for it using the Accept-Ranges header.
13.4.7. Playing Live with Recording
Certain media servers may offer recording services of live sessions
to their clients. This recording would normally be from the
beginning of the media session. Clients can randomly access the
media between now and the beginning of the media session. This live
media with recording is indicated by the content of the Media-
Properties header in the SETUP response when (see also
Section 18.29):
o the Random Access property is set to Random-Access;
o the Content Modifications property is set to Time-Progressing;
o the Retention property is set to Time-Limited or Unlimited
The SETUP 200 OK response MUST include the Media-Range header (see
Section 18.30) for this type of media. For live media with
recording, the Range header indicates the current delivery point in
the media and the Media-Range header indicates the currently
available media window around the current time. This window can
cover recorded content in the past (seen from current time in the
media) or recorded content in the future (seen from current time in
the media). The server adjusts the delivery point to the requested
border of the window. If the client requests a delivery point that
is located outside the recording window, e.g., if the requested point
is too far in the past, the server selects the oldest point in the
recording. The considerations in Section 13.5.3 apply if a client
requests delivery with scale (Section 18.46) values other than 1.0
(normal playback rate) while delivering live media with recording.
13.4.8. Playing Live with Time-Shift
Certain media servers may offer time-shift services to their clients.
This time shift records a fixed interval in the past, i.e., a sliding
window recording mechanism, but not past this interval. Clients can
randomly access the media between now and the interval. This live
media with recording is indicated by the content of the Media-
Properties header in the SETUP response when (see also
Section 18.29):
o the Random Access property is set to Random-Access;
o the Content Modifications property is set to Time-Progressing;
o the Retention property is set to Time-Duration and a value
indicating the recording interval (>0).
The SETUP 200 OK response MUST include the Media-Range header (see
Section 18.30) for this type of media. For live media with
recording, the Range header indicates the current time in the media
and the Media-Range header indicates a window around the current
time. This window can cover recorded content in the past (seen from
current time in the media) or recorded content in the future (seen
from current time in the media). The server adjusts the play point
to the requested border of the window, if the client requests a play
point that is located outside the recording windows, e.g., if
requested too far in the past, the server selects the oldest range in
the recording. The considerations in Section 13.5.3 apply if a
client requests delivery using a scale (Section 18.46) value other
than 1.0 (normal playback rate) while delivering live media with
time-shift.
13.5. PLAY_NOTIFY
The PLAY_NOTIFY method is issued by a server to inform a client about
an asynchronous event for a session in Play state. The Session
header MUST be presented in a PLAY_NOTIFY request and indicates the
scope of the request. Sending of PLAY_NOTIFY requests requires a
persistent connection between server and client; otherwise, there is
no way for the server to send this request method to the client.
PLAY_NOTIFY requests have an end-to-end (i.e., server-to-client)
scope, as they carry the Session header, and apply only to the given
session. The client SHOULD immediately return a response to the
server.
PLAY_NOTIFY requests MAY use both an aggregate control URI and
individual media resource URIs, depending on the scope of the
notification. This scope may have important distinctions for
aggregated sessions, and each reason for a PLAY_NOTIFY request needs
to specify the interpretation as well as if aggregated control URIs
or individual URIs may be used in requests.
PLAY_NOTIFY requests can be used with a message body, depending on
the value of the Notify-Reason header. It is described in the
particular section for each Notify-Reason if a message body is used.
However, currently there is no Notify-Reason that allows the use of a
message body. In this case, there is a need to obey some limitations
when adding new Notify-Reasons that intend to use a message body: the
server can send any type of message body, but it is not ensured that
the client can understand the received message body. This is related
to DESCRIBE (see Section 13.2 ); but, in this particular case, the
client can state its acceptable message bodies by using the Accept
header. In the case of PLAY_NOTIFY, the server does not know which
message bodies are understood by the client.
The Notify-Reason header (see Section 18.32) specifies the reason why
the server sends the PLAY_NOTIFY request. This is extensible and new
reasons can be added in the future (see Section 22.8). In case the
client does not understand the reason for the notification, it MUST
respond with a 465 (Notification Reason Unknown) (Section 17.4.29)
error code. This document defines how servers can send PLAY_NOTIFY
with Notify-Reason values of these types:
o end-of-stream (see Section 13.5.1);
o media-properties-update (see Section 13.5.2);
o scale-change (see Section 13.5.3).
13.5.1. End-of-Stream
A PLAY_NOTIFY request with the Notify-Reason header set to end-of-
stream indicates the completion or near completion of the PLAY
request and the ending delivery of the media stream(s). The request
MUST NOT be issued unless the server is in the Play state. The end
of the media stream delivery notification may be used either to
indicate a successful completion of the PLAY request currently being
served or to indicate some error resulting in failure to complete the
request. The Request-Status header (Section 18.42) MUST be included
to indicate which request the notification is for and its completion
status. The message response status codes (Section 8.1.1) are used
to indicate how the PLAY request concluded. The sender of a
PLAY_NOTIFY MAY issue an updated PLAY_NOTIFY, in the case of a
PLAY_NOTIFY sent with wrong information. For instance, a PLAY_NOTIFY
was issued before reaching the end-of-stream, but some error occurred
resulting in that the previously sent PLAY_NOTIFY contained a wrong
time when the stream will end. In this case, a new PLAY_NOTIFY MUST
be sent including the correct status for the completion and all
additional information.
PLAY_NOTIFY requests with the Notify-Reason header set to end-of-
stream MUST include a Range header and the Scale header if the scale
value is not 1. The Range header indicates the point in the stream
or streams where delivery is ending with the timescale that was used
by the server in the PLAY response for the request being fulfilled.
The server MUST NOT use the "now" constant in the Range header; it
MUST use the actual numeric end position in the proper timescale.
When end-of-stream notifications are issued prior to having sent the
last media packets, this is made evident because the end time in the
Range header is beyond the current time in the media being received
by the client, e.g., "npt=-15", if npt is currently at 14.2 seconds.
The Scale header is to be included so that it is evident if the media
timescale is moving backwards or has a non-default pace. The end-of-
stream notification does not prevent the client from sending a new
PLAY request.
If RTP is used as media transport, an RTP-Info header MUST be
included, and the RTP-Info header MUST indicate the last sequence
number in the sequence parameter.
For an RTSP Session where media resources are under aggregated
control, the media resources will normally end at approximately the
same time, although some small differences may exist, on the scale of
a few hundred milliseconds. In those cases, an RTSP session under
aggregated control SHOULD send only a single PLAY_NOTIFY request. By
using the aggregate control URI in the PLAY_NOTIFY request, the RTSP
server indicates that this applies to all media resources within the
session. In cases in which RTP is used for media delivery,
corresponding RTP-Info needs to be included for all media resources.
In cases where one or more media resources have a significantly
shorter duration than some other resources in the aggregated session,
the server MAY send end-of-stream notifications using individual
media resource URIs to indicate to agents that there will be no more
media for this particular media resource related to the current
active PLAY request. In such cases, when the remaining media
resources come to the end of the stream, they MUST send a PLAY_NOTIFY
request using the aggregate control URI to indicate that no more
resources remain.
A PLAY_NOTIFY request with Notify-Reason header set to end-of-stream
MUST NOT carry a message body.
This example request notifies the client about a future end-of-stream
event:
S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 854
Notify-Reason: end-of-stream
Request-Status: cseq=853 status=200 reason="OK"
Range: npt=-145
RTP-Info:url="rtsp://example.com/fizzle/foo/audio"
ssrc=0D12F123:seq=14783;rtptime=2345962545,
url="rtsp://example.com/fizzle/video"
ssrc=789DAF12:seq=57654;rtptime=2792482193
Session: CDtUJfDQXJWtJ7Iqua2xOi
Date: Mon, 08 Mar 2010 13:37:16 GMT
C->S: RTSP/2.0 200 OK
CSeq: 854
User-Agent: PhonyClient/1.2
Session: CDtUJfDQXJWtJ7Iqua2xOi
13.5.2. Media-Properties-Update
A PLAY_NOTIFY request with a Notify-Reason header set to media-
properties-update indicates an update of the media properties for the
given session (see Section 18.29) or the available media range that
can be played as indicated by the Media-Range header (Section 18.30).
PLAY_NOTIFY requests with Notify-Reason header set to media-
properties-update MUST include a Media-Properties and Date header and
SHOULD include a Media-Range header. The Media-Properties header has
session scope; thus, for aggregated sessions, the PLAY_NOTIFY request
MUST use the aggregated control URI.
This notification MUST be sent for media that are time-progressing
every time an event happens that changes the basis for making
estimates on how the available for play-back media range will
progress with wall clock time. In addition, it is RECOMMENDED that
the server send these notifications approximately every 5 minutes for
time-progressing content to ensure the long-term stability of the
client estimation and allow for clock skew detection by the client.
The time between notifications should be greater than 1 minute and
less than 2 hours. For the reasons just explained, requests MUST
include a Media-Range header to provide current Media duration and a
Range header to indicate the current playing point and any remaining
parts of the requested range.
The recommendation for sending updates every 5 minutes is due to
any clock skew issues. In 5 minutes, the clock skew should not
become too significant as this is not used for media playback and
synchronization, it is only for determining which content is
available to the user.
A PLAY_NOTIFY request with Notify-Reason header set to media-
properties-update MUST NOT carry a message body.
S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0
Date: Tue, 14 Apr 2008 15:48:06 GMT
CSeq: 854
Notify-Reason: media-properties-update
Session: CDtUJfDQXJWtJ7Iqua2xOi
Media-Properties: Time-Progressing,
Time-Limited=20080415T153919.36Z, Random-Access=5.0
Media-Range: npt=00:00:00-01:37:21.394
Range: npt=01:15:49.873-
C->S: RTSP/2.0 200 OK
CSeq: 854
User-Agent: PhonyClient/1.2
Session: CDtUJfDQXJWtJ7Iqua2xOi
13.5.3. Scale-Change
The server may be forced to change the rate of media time per
playback time when a client requests delivery using a scale
(Section 18.46) value other than 1.0 (normal playback rate). For
time-progressing media with some retention, i.e., the server stores
already-sent content, a client requesting to play with scale values
larger than 1 may catch up with the front end of the media. The
server will then be unable to continue to provide content at scale
larger than 1 as content is only made available by the server at
scale = 1. Another case is when scale < 1 and the media retention is
Time-Duration limited. In this case, the delivery point can reach
the oldest media unit available, and further playback at this scale
becomes impossible as there will be no media available. To avoid
having the client lose any media, the scale will need to be adjusted
to the same rate at which the media is removed from the storage
buffer, commonly scale = 1.0.
Another case is when the content itself consists of spliced pieces or
is dynamically updated. In these cases, the server may be required
to change from one supported scale value (different than scale = 1.0)
to another. In this case, the server will pick the closest value and
inform the client of what it has picked. In these cases, the media
properties will also be sent, updating the supported scale values.
This enables a client to adjust the scale value used.
To minimize impact on playback in any of the above cases, the server
MUST modify the playback properties, set scale to a supportable
value, and continue delivery of the media. When doing this
modification, it MUST send a PLAY_NOTIFY message with the Notify-
Reason header set to "scale-change". The request MUST contain a
Range header with the media time when the change took effect, a Scale
header with the new value in use, a Session header with the
identifier for the session to which it applies, and a Date header
with the server wallclock time of the change. For time-progressing
content, the Media-Range and the Media-Properties headers at this
point in time also MUST be included. The Media-Properties header
MUST be included if the scale change was due to the content changing
what scale values ("Scales") are supported.
For media streams delivered using RTP, an RTP-Info header MUST also
be included. It MUST contain the rtptime parameter with a value
corresponding to the point of change in that media and optionally the
sequence number.
PLAY_NOTIFY requests for aggregated sessions MUST use the aggregated
control URI in the request. The scale change for any aggregated
session applies to all media streams that are part of the aggregate.
A PLAY_NOTIFY request with Notify-Reason header set to "Scale-Change"
MUST NOT carry a message body.
S->C: PLAY_NOTIFY rtsp://example.com/fizzle/foo RTSP/2.0
Date: Tue, 14 Apr 2008 15:48:06 GMT
CSeq: 854
Notify-Reason: scale-change
Session: CDtUJfDQXJWtJ7Iqua2xOi
Media-Properties: Time-Progressing,
Time-Limited=20080415T153919.36Z, Random-Access=5.0
Media-Range: npt=00:00:00-01:37:21.394
Range: npt=01:37:21.394-
Scale: 1
RTP-Info: url="rtsp://example.com/fizzle/foo/audio"
ssrc=0D12F123:rtptime=2345962545,
url="rtsp://example.com/fizzle/foo/videotrack"
ssrc=789DAF12:seq=57654;rtptime=2792482193
C->S: RTSP/2.0 200 OK
CSeq: 854
User-Agent: PhonyClient/1.2
Session: CDtUJfDQXJWtJ7Iqua2xOi
13.6. PAUSE
The PAUSE request causes the stream delivery to immediately be
interrupted (halted). A PAUSE request MUST be made either with the
aggregated control URI for aggregated sessions, resulting in all
media being halted, or with the media URI for non-aggregated
sessions. Any attempt to mute a single media with a PAUSE request in
an aggregated session MUST be responded to with a 460 (Only Aggregate
Operation Allowed) error. After resuming playback, synchronization
of the tracks MUST be maintained. Any server resources are kept,
though servers MAY close the session and free resources after being
paused for the duration specified with the timeout parameter of the
Session header in the SETUP message.
Example:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 834
Session: OoOUPyUwt0VeY9fFRHuZ6L
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 834
Date: Thu, 23 Jan 1997 15:35:06 GMT
Session: OoOUPyUwt0VeY9fFRHuZ6L
Range: npt=45.76-75.00
The PAUSE request causes stream delivery to be interrupted
immediately on receipt of the message, and the pause point is set to
the current point in the presentation. That pause point in the media
stream needs to be maintained. A subsequent PLAY request without a
Range header resumes from the pause point and plays until media end.
The pause point after any PAUSE request MUST be returned to the
client by adding a Range header with what remains unplayed of the
PLAY request's range. For media with random access properties, if
one desires to resume playing a ranged request, one simply includes
the Range header from the PAUSE response and includes the Seek-Style
header with the Next policy in the PLAY request. For media that is
time-progressing and has retention duration=0, the follow-up PLAY
request to start media delivery again MUST use "npt=now-" and not the
answer given in the response to PAUSE.
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 834
Session: OccldOFFq23KwjYpAnBbUr
Range: npt=10-30
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 834
Date: Thu, 23 Jan 1997 15:35:06 GMT
Server: PhonyServer/1.0
Range: npt=10-30
Seek-Style: First-Prior
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934207921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=4FAD8726:seq=57654;rtptime=2792482193
Session: OccldOFFq23KwjYpAnBbUr
After 11 seconds, i.e., at 21 seconds into the presentation:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 835
Session: OccldOFFq23KwjYpAnBbUr
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 835
Date: 23 Jan 1997 15:35:17 GMT
Server: PhonyServer/1.0
Range: npt=21-30
Session: OccldOFFq23KwjYpAnBbUr
If a client issues a PAUSE request and the server acknowledges and
enters the Ready state, the proper server response, if the player
issues another PAUSE, is still 200 OK. The 200 OK response MUST
include the Range header with the current pause point. See examples
below:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 834
Session: OccldOFFq23KwjYpAnBbUr
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 834
Session: OccldOFFq23KwjYpAnBbUr
Date: Thu, 23 Jan 1997 15:35:06 GMT
Range: npt=45.76-98.36
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 835
Session: OccldOFFq23KwjYpAnBbUr
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 835
Session: OccldOFFq23KwjYpAnBbUr
Date: 23 Jan 1997 15:35:07 GMT
Range: npt=45.76-98.36
13.7. TEARDOWN
13.7.1. Client to Server
The TEARDOWN client-to-server request stops the stream delivery for
the given URI, freeing the resources associated with it. A TEARDOWN
request can be performed on either an aggregated or a media control
URI. However, some restrictions apply depending on the current
state. The TEARDOWN request MUST contain a Session header indicating
to what session the request applies. The TEARDOWN request MUST NOT
include a Terminate-Reason header.
A TEARDOWN using the aggregated control URI or the media URI in a
session under non-aggregated control (single media session) MAY be
done in any state (Ready and Play). A successful request MUST result
in that media delivery being immediately halted and the session state
being destroyed. This MUST be indicated through the lack of a
Session header in the response.
A TEARDOWN using a media URI in an aggregated session can only be
done in Ready state. Such a request only removes the indicated media
stream and associated resources from the session. This may result in
a session returning to non-aggregated control, because it only
contains a single media after the request's completion. A session
that will exist after the processing of the TEARDOWN request MUST, in
the response to that TEARDOWN request, contain a Session header.
Thus, the presence of the Session header indicates to the receiver of
the response if the session is still extant or has been removed.
Example:
C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 892
Session: OccldOFFq23KwjYpAnBbUr
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 892
Server: PhonyServer/1.0
13.7.2. Server to Client
The server can send TEARDOWN requests in the server-to-client
direction to indicate that the server has been forced to terminate
the ongoing session. This may happen for several reasons, such as
server maintenance without available backup, or that the session has
been inactive for extended periods of time. The reason is provided
in the Terminate-Reason header (Section 18.52).
When an RTSP client has maintained an RTSP session that otherwise is
inactive for an extended period of time, the server may reclaim the
resources. That is done by issuing a TEARDOWN request with the
Terminate-Reason set to "Session-Timeout". This MAY be done when the
client has been inactive in the RTSP session for more than one
Session Timeout period (Section 18.49). However, the server is NOT
RECOMMENDED to perform this operation until an extended period of
inactivity of 10 times the Session-Timeout period has passed. It is
up to the operator of the RTSP server to actually configure how long
this extended period of inactivity is. An operator should take into
account, when doing this configuration, what the served content is
and what this means for the extended period of inactivity.
In case the server needs to stop providing service to the established
sessions and there is no server to point at in a REDIRECT request,
then TEARDOWN SHALL be used to terminate the session. This method
can also be used when non-recoverable internal errors have happened
and the server has no other option than to terminate the sessions.
The TEARDOWN request MUST be made only on the session aggregate
control URI (i.e., it is not allowed to terminate individual media
streams, if it is a session aggregate), and it MUST include the
following headers: Session and Terminate-Reason. The request only
applies to the session identified in the Session header. The server
may include a message to the client's user with the "user-msg"
parameter.
The TEARDOWN request may alternatively be done on the wildcard URI
"*" and without any session header. The scope of such a request is
limited to the next-hop (i.e., the RTSP agent in direct communication
with the server) and applies, as well, to the RTSP connection between
the next-hop RTSP agent and the server. This request indicates that
all sessions and pending requests being managed via the connection
are terminated. Any intervening proxies SHOULD do all of the
following in the order listed:
1. respond to the TEARDOWN request
2. disconnect the control channel from the requesting server
3. pass the TEARDOWN request to each applicable client (typically
those clients with an active session or an unanswered request)
Note: The proxy is responsible for accepting TEARDOWN responses
from its clients; these responses MUST NOT be passed on to either
the original server or the target server in the redirect.
13.8. GET_PARAMETER
The GET_PARAMETER request retrieves the value of any specified
parameter or parameters for a presentation or stream specified in the
URI. If the Session header is present in a request, the value of a
parameter MUST be retrieved in the specified session context. There
are two ways of specifying the parameters to be retrieved.
The first approach includes headers that have been defined to be
usable for this purpose. Headers for this purpose should allow
empty, or stripped value parts to avoid having to specify bogus data
when indicating the desire to retrieve a value. The successful
completion of the request should also be evident from any filled out
values in the response. The headers in this specification that MAY
be used for retrieving their current value using GET_PARAMETER are
listed below; additional headers MAY be specified in the future:
o Accept-Ranges
o Media-Range
o Media-Properties
o Range
o RTP-Info
The other way is to specify a message body that lists the
parameter(s) that are desired to be retrieved. The Content-Type
header (Section 18.19) is used to specify which format the message
body has. If the receiver of the request does not support the media
type used for the message body, it SHALL respond using the error code
415 (Unsupported Media Type). The responder to a GET_PARAMETER
request MUST use the media type of the request for the response. For
additional considerations regarding message body negotiation, see
Section 9.3.
RTSP agents implementing support for responding to GET_PARAMETER
requests SHALL implement the "text/parameters" format (Appendix F).
This to ensure that at least one known format for parameters is
implemented and, thus, prevent parameter format negotiation failure.
Parameters specified within the body of the message must all be
understood by the request-receiving agent. If one or more parameters
are not understood a 451 (Parameter Not Understood) MUST be sent
including a body listing the parameters that weren't understood. If
all parameters are understood, their values are filled in and
returned in the response message body.
The method can also be used without a message body or any header that
requests parameters for keep-alive purposes. The keep-alive timer
has been updated for any request that is successful, i.e., a 200 OK
response is received. Any non-required header present in such a
request may or may not have been processed. Normally, the presence
of filled-out values in the header will be indication that the header
has been processed. However, for cases when this is difficult to
determine, it is recommended to use a feature tag and the Require
header. For this reason, it is usually easier if any parameters to
be retrieved are sent in the body, rather than using any header.
Example:
S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 431
User-Agent: PhonyClient/1.2
Session: OccldOFFq23KwjYpAnBbUr
Content-Length: 26
Content-Type: text/parameters
packets_received
jitter
C->S: RTSP/2.0 200 OK
CSeq: 431
Session: OccldOFFq23KwjYpAnBbUr
Server: PhonyServer/1.1
Date: Mon, 08 Mar 2010 13:43:23 GMT
Content-Length: 38
Content-Type: text/parameters
packets_received: 10
jitter: 0.3838
13.9. SET_PARAMETER
This method requests the setting of the value of a parameter or a set
of parameters for a presentation or stream specified by the URI. If
the Session header is present in a request, the value of a parameter
MUST be retrieved in the specified session context. The method MAY
also be used without a message body. It is the RECOMMENDED method to
be used in a request sent for the sole purpose of updating the keep-
alive timer. If this request is successful, i.e., a 200 OK response
is received, then the keep-alive timer has been updated. Any non-
required header present in such a request may or may not have been
processed. To allow a client to determine if any such header has
been processed, it is necessary to use a feature tag and the Require
header. Due to this reason it is RECOMMENDED that any parameters are
sent in the body rather than using any header.
When using a message body to list the parameter(s) desired to be set,
the Content-Type header (Section 18.19) is used to specify which
format the message body has. If the receiver of the request is not
supporting the media type used for the message body, it SHALL respond
using the error code 415 (Unsupported Media Type). For additional
considerations regarding message body negotiation, see Section 9.3.
The responder to a SET_PARAMETER request MUST use the media type of
the request for the response. For additional considerations
regarding message body negotiation, see Section 9.3.
RTSP agents implementing support for responding to SET_PARAMETER
requests SHALL implement the text/parameters format (Appendix F).
This is to ensure that at least one known format for parameters is
implemented and, thus, prevent parameter format negotiation failure.
A request is RECOMMENDED to only contain a single parameter to allow
the client to determine why a particular request failed. If the
request contains several parameters, the server MUST only act on the
request if all of the parameters can be set successfully. A server
MUST allow a parameter to be set repeatedly to the same value, but it
MAY disallow changing parameter values. If the receiver of the
request does not understand or cannot locate a parameter, error 451
(Parameter Not Understood) MUST be used. When a parameter is not
allowed to change, the error code is 458 (Parameter Is Read-Only).
The response body MUST contain only the parameters that have errors.
Otherwise, a body MUST NOT be returned. The response body MUST use
the media type of the request for the response.
Note: transport parameters for the media stream MUST only be set with
the SETUP command.
Restricting setting transport parameters to SETUP is for the
benefit of firewalls connected to border RTSP proxies.
The parameters are split in a fine-grained fashion so that there
can be more meaningful error indications. However, it may make
sense to allow the setting of several parameters if an atomic
setting is desirable. Imagine device control where the client
does not want the camera to pan unless it can also tilt to the
right angle at the same time.
Example:
C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 421
User-Agent: PhonyClient/1.2
Session: iixT43KLc
Date: Mon, 08 Mar 2010 14:45:04 GMT
Content-length: 20
Content-type: text/parameters
barparam: barstuff
S->C: RTSP/2.0 451 Parameter Not Understood
CSeq: 421
Session: iixT43KLc
Server: PhonyServer/1.0
Date: Mon, 08 Mar 2010 14:44:56 GMT
Content-length: 20
Content-type: text/parameters
barparam: barstuff
13.10. REDIRECT
The REDIRECT method is issued by a server to inform a client that the
service provided will be terminated and where a corresponding service
can be provided instead. This may happen for different reasons. One
is that the server is being administered such that it must stop
providing service. Thus, the client is required to connect to
another server location to access the resource indicated by the
Request-URI.
The REDIRECT request SHALL contain a Terminate-Reason header
(Section 18.52) to inform the client of the reason for the request.
Additional parameters related to the reason may also be included.
The intention here is to allow a server administrator to do a
controlled shutdown of the RTSP server. That requires sufficient
time to inform all entities having associated state with the server
and for them to perform a controlled migration from this server to a
fall-back server.
A REDIRECT request with a Session header has end-to-end (i.e.,
server-to-client) scope and applies only to the given session. Any
intervening proxies SHOULD NOT disconnect the control channel while
there are other remaining end-to-end sessions. The REQUIRED Location
header MUST contain a complete absolute URI pointing to the resource
to which the client SHOULD reconnect. Specifically, the Location
MUST NOT contain just the host and port. A client may receive a
REDIRECT request with a Session header, if and only if, an end-to-end
session has been established.
A client may receive a REDIRECT request without a Session header at
any time when it has communication or a connection established with a
server. The scope of such a request is limited to the next-hop
(i.e., the RTSP agent in direct communication with the server) and
applies to all sessions controlled, as well as the connection between
the next-hop RTSP agent and the server. A REDIRECT request without a
Session header indicates that all sessions and pending requests being
managed via the connection MUST be redirected. The Location header,
if included in such a request, SHOULD contain an absolute URI with
only the host address and the OPTIONAL port number of the server to
which the RTSP agent SHOULD reconnect. Any intervening proxies
SHOULD do all of the following in the order listed:
1. respond to the REDIRECT request
2. disconnect the control channel from the requesting server
3. connect to the server at the given host address
4. pass the REDIRECT request to each applicable client (typically
those clients with an active session or an unanswered request)
Note: The proxy is responsible for accepting REDIRECT responses
from its clients; these responses MUST NOT be passed on to either
the original server or the redirected server.
A server that needs to terminate a session or all its sessions and
lacks an alternative server to redirect to, SHALL instead use
TEARDOWN requests.
When no Terminate-Reason "time" parameter is included in a REDIRECT
request, the client SHALL perform the redirection immediately and
return a response to the server. The server shall consider the
session to be terminated and can free any associated state after it
receives the successful (2xx) response. The server MAY close the
signaling connection upon receiving the response, and the client
SHOULD close the signaling connection after sending the 2xx response.
The exception to this is when the client has several sessions on the
server being managed by the given signaling connection. In this
case, the client SHOULD close the connection when it has received and
responded to REDIRECT requests for all the sessions managed by the
signaling connection.
The Terminate-Reason header "time" parameter MAY be used to indicate
the wallclock time by which the redirection MUST have taken place.
To allow a client to determine that redirect time without being time
synchronized with the server, the server MUST include a Date header
in the request. The client should have terminated the session and
closed the connection before the redirection time-line terminated.
The server MAY simply cease to provide service when the deadline time
has been reached, or it can issue a TEARDOWN requests to the
remaining sessions.
If the REDIRECT request times out following the rules in
Section 10.4, the server MAY terminate the session or transport
connection that would be redirected by the request. This is a
safeguard against misbehaving clients that refuse to respond to a
REDIRECT request. This action removes any incentive of not
acknowledging the reception of a REDIRECT request.
After a REDIRECT request has been processed, a client that wants to
continue to receive media for the resource identified by the Request-
URI will have to establish a new session with the designated host.
If the URI given in the Location header is a valid resource URI, a
client SHOULD issue a DESCRIBE request for the URI.
Note: The media resource indicated by the Location header can be
identical, slightly different, or totally different. This is the
reason why a new DESCRIBE request SHOULD be issued.
If the Location header contains only a host address, the client may
assume that the media on the new server is identical to the media on
the old server, i.e., all media configuration information from the
old session is still valid except for the host address. However, the
usage of conditional SETUP using MTag identifiers is RECOMMENDED as a
means to verify the assumption.
This example request redirects traffic for this session to the new
server at the given absolute time:
S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 732
Location: rtsp://s2.example.com:8001/fizzle/foo
Terminate-Reason: Server-Admin ;time=19960213T143205Z
Session: uZ3ci0K+Ld-M
Date: Thu, 13 Feb 1996 14:30:43 GMT
C->S: RTSP/2.0 200 OK
CSeq: 732
User-Agent: PhonyClient/1.2
Session: uZ3ci0K+Ld-M
14. Embedded (Interleaved) Binary Data
In order to fulfill certain requirements on the network side, e.g.,
in conjunction with network address translators that block RTP
traffic over UDP, it may be necessary to interleave RTSP messages and
media-stream data. This interleaving should generally be avoided
unless necessary since it complicates client and server operation and
imposes additional overhead. Also, head-of-line blocking may cause
problems. Interleaved binary data SHOULD only be used if RTSP is
carried over TCP. Interleaved data is not allowed inside RTSP
messages.
Stream data, such as RTP packets, is encapsulated by an ASCII dollar
sign (36 decimal) followed by a one-octet channel identifier and the
length of the encapsulated binary data as a binary, two-octet
unsigned integer in network octet order (Appendix B of [RFC791]).
The stream data follows immediately afterwards, without a CRLF, but
including the upper-layer protocol headers. Each dollar sign block
MUST contain exactly one upper-layer protocol data unit, e.g., one
RTP packet.
Note that this mechanism does not support PDUs larger than 65535
octets, which matches the maximum payload size of regular, non-
jumbo IPv4 and IPv6 packets. If the media delivery protocol
intended to be used has larger PDUs than that, a definition of a
PDU fragmentation mechanism will be required to support embedded
binary data.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| "$" = 36 | Channel ID | Length in octets |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: Binary data (Length according to Length field) :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: Embedded Interleaved Binary Data Format
The channel identifier is defined in the Transport header with the
interleaved parameter (Section 18.54).
When the transport choice is RTP, RTCP messages are also interleaved
by the server over the TCP connection. The usage of RTCP messages is
indicated by including an interval containing a second channel in the
interleaved parameter of the Transport header (see Section 18.54).
If RTCP is used, packets MUST be sent on the first available channel
that is higher than the RTP channel. The channels are bidirectional,
using the same Channel ID in both directions; therefore, RTCP traffic
is sent on the second channel in both directions.
RTCP is sometimes needed for synchronization when two or more
streams are interleaved in such a fashion. Also, this provides a
convenient way to tunnel RTP/RTCP packets through the RTSP
connection (TCP or TCP/TLS) when required by the network
configuration and to transfer them onto UDP when possible.
C->S: SETUP rtsp://example.com/bar.file RTSP/2.0
CSeq: 2
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
Accept-Ranges: npt, smpte, clock
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 2
Date: Thu, 05 Jun 1997 18:57:18 GMT
Transport: RTP/AVP/TCP;unicast;interleaved=5-6
Session: OccldOFFq23KwjYpAnBbUr
Accept-Ranges: npt
Media-Properties: Random-Access=0.2, Immutable, Unlimited
C->S: PLAY rtsp://example.com/bar.file RTSP/2.0
CSeq: 3
Session: OccldOFFq23KwjYpAnBbUr
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 3
Session: OccldOFFq23KwjYpAnBbUr
Date: Thu, 05 Jun 1997 18:57:19 GMT
RTP-Info: url="rtsp://example.com/bar.file"
ssrc=0D12F123:seq=232433;rtptime=972948234
Range: npt=0-56.8
Seek-Style: RAP
S->C: $005{2 octet length}{"length" octets data, w/RTP header}
S->C: $005{2 octet length}{"length" octets data, w/RTP header}
S->C: $006{2 octet length}{"length" octets RTCP packet}
15. Proxies
RTSP Proxies are RTSP agents that are located in between a client and
a server. A proxy can take on the roles of both client and server
depending on what it tries to accomplish. RTSP proxies use two
transport-layer connections: one from the RTSP client to the RTSP
proxy and a second from the RTSP proxy to the RTSP server. Proxies
are introduced for several different reasons; those listed below are
often combined.
Caching Proxy: This type of proxy is used to reduce the workload on
servers and connections. By caching the description and media
streams, i.e., the presentation, the proxy can serve a client
with content, but without requesting it from the server once it
has been cached and has not become stale. See Section 16.
This type of proxy is also expected to understand RTSP endpoint
functionality, i.e., functionality identified in the Require
header in addition to what Proxy-Require demands.
Translator Proxy: This type of proxy is used to ensure that an RTSP
client gets access to servers and content on an external
network or gets access by using content encodings not supported
by the client. The proxy performs the necessary translation of
addresses, protocols, or encodings. This type of proxy is
expected also to understand RTSP endpoint functionality, i.e.,
functionality identified in the Require header in addition to
what Proxy-Require demands.
Access Proxy: This type of proxy is used to ensure that an RTSP
client gets access to servers on an external network. Thus,
this proxy is placed on the border between two domains, e.g., a
private address space and the public Internet. The proxy
performs the necessary translation, usually addresses. This
type of proxy is required to redirect the media to itself or a
controlled gateway that performs the translation before the
media can reach the client.
Security Proxy: This type of proxy is used to help facilitate
security functions around RTSP. For example, in the case of a
firewalled network, the security proxy requests that the
necessary pinholes in the firewall are opened when a client in
the protected network wants to access media streams on the
external side. This proxy can perform its function without
redirecting the media between the server and client. However,
in deployments with private address spaces, this proxy is
likely to be combined with the access proxy. The functionality
of this proxy is usually closely tied into understanding all
aspects of the media transport.
Auditing Proxy: RTSP proxies can also provide network owners with a
logging and auditing point for RTSP sessions, e.g., for
corporations that track their employees usage of the network.
This type of proxy can perform its function without inserting
itself or any other node in the media transport. This proxy
type can also accept unknown methods as it doesn't interfere
with the clients' requests.
All types of proxies can also be used when using secured
communication with TLS, as RTSP 2.0 allows the client to approve
certificate chains used for connection establishment from a proxy;
see Section 19.3.2. However, that trust model may not be suitable
for all types of deployment. In those cases, the secured sessions do
bypass the proxies.
Access proxies SHOULD NOT be used in equipment like NATs and
firewalls that aren't expected to be regularly maintained, like home
or small office equipment. In these cases, it is better to use the
NAT traversal procedures defined for RTSP 2.0 [RFC7825]. The reason
for these recommendations is that any extensions of RTSP resulting in
new media-transport protocols or profiles, new parameters, etc., may
fail in a proxy that isn't maintained. This would impede RTSP's
future development and usage.
15.1. Proxies and Protocol Extensions
The existence of proxies must always be considered when developing
new RTSP extensions. Most types of proxies will need to implement
any new method to operate correctly in the presence of that
extension. New headers can be introduced and will not be blocked by
older proxies. However, it is important to consider if this header
and its function are required to be understood by the proxy or if it
can be simply forwarded. If the header needs to be understood, a
feature tag representing the functionality MUST be included in the
Proxy-Require header. Below are guidelines for analysis whether the
header needs to be understood. The Transport header and its
parameters are extensible, which requires handling rules for a proxy
in order to ensure a correct interpretation.
Whether or not a proxy needs to understand a header is not easy to
determine as they serve a broad variety of functions. When
evaluating if a header needs to be understood, one can divide the
functionality into three main categories:
Media modifying: The caching and translator proxies modify the
actual media and therefore need also to understand the request
directed to the server that affects how the media is rendered.
Thus, this type of proxy also needs to understand the server-side
functionality.
Transport modifying: The access and the security proxy both need to
understand how the media transport is performed, either for
opening pinholes or translating the outer headers, e.g., IP and
UDP or TCP.
Non-modifying: The audit proxy is special in that it does not modify
the messages in other ways than to insert the Via header. That
makes it possible for this type to forward RTSP messages that
contain different types of unknown methods, headers, or header
parameters.
An extension has to be classified as mandatory to be implemented for
a proxy, if an extension has to be understood by a "Transport
modifying" type of proxy.
15.2. Multiplexing and Demultiplexing of Messages
RTSP proxies may have to multiplex several RTSP sessions from their
clients towards RTSP servers. This requires that RTSP requests from
multiple clients be multiplexed onto a common connection for requests
outgoing to an RTSP server, and, on the way back, the responses be
demultiplexed from the server to per-client responses. On the
protocol level, this requires that request and response messages be
handled in both directions, requiring that there be a mechanism to
correlate which request/response pair exchanged between proxy and
server is mapped to which client (or client request).
This multiplexing of requests and demultiplexing of responses is done
by using the CSeq header field. The proxy has to rewrite the CSeq in
requests to the server and responses from the server and remember
which CSeq is mapped to which client. The proxy also needs to ensure
that the order of the message related to each client is maintained.
Section 18.20 defines the handling of how requests and responses are
rewritten.
16. Caching
In HTTP, request/response pairs are cached. RTSP differs
significantly in that respect. Responses are not cacheable, with the
exception of the presentation description returned by DESCRIBE.
(Since the responses for anything but DESCRIBE and GET_PARAMETER do
not return any data, caching is not really an issue for these
requests.) However, it is desirable for the continuous media data,
typically delivered out-of-band with respect to RTSP, to be cached,
as well as the session description.
On receiving a SETUP or PLAY request, a proxy ascertains whether it
has an up-to-date copy of the continuous media content and its
description. It can determine whether the copy is up to date by
issuing a SETUP or DESCRIBE request, respectively, and comparing the
Last-Modified header with that of the cached copy. If the copy is
not up to date, it modifies the SETUP transport parameters as
appropriate and forwards the request to the origin server.
Subsequent control commands such as PLAY or PAUSE then pass the proxy
unmodified. The proxy delivers the continuous media data to the
client, while possibly making a local copy for later reuse. The
exact allowed behavior of the cache is given by the cache-response
directives described in Section 18.11. A cache MUST answer any
DESCRIBE requests if it is currently serving the stream to the
requester, as it is possible that low-level details of the stream
description may have changed on the origin server.
Note that an RTSP cache is of the "cut-through" variety. Rather than
retrieving the whole resource from the origin server, the cache
simply copies the streaming data as it passes by on its way to the
client. Thus, it does not introduce additional latency.
To the client, an RTSP proxy cache appears like a regular media
server. To the media origin server, an RTSP proxy cache appears like
a client. Just as an HTTP cache has to store the content type,
content language, and so on for the objects it caches, a media cache
has to store the presentation description. Typically, a cache
eliminates all transport references (e.g., multicast information)
from the presentation description, since these are independent of the
data delivery from the cache to the client. Information on the
encodings remains the same. If the cache is able to translate the
cached media data, it would create a new presentation description
with all the encoding possibilities it can offer.
16.1. Validation Model
When a cache has a stale entry that it would like to use as a
response to a client's request, it first has to check with the origin
server (or possibly an intermediate cache with a fresh response) to
see if its cached entry is still usable. This is called "validating"
the cache entry. To avoid having to pay the overhead of
retransmitting the full response if the cached entry is good, and at
the same time avoiding having to pay the overhead of an extra round
trip if the cached entry is invalid, RTSP supports the use of
conditional methods.
The key protocol features for supporting conditional methods are
those concerned with "cache validators." When an origin server
generates a full response, it attaches some sort of validator to it,
which is kept with the cache entry. When a client (user agent or
proxy cache) makes a conditional request for a resource for which it
has a cache entry, it includes the associated validator in the
request.
The server then checks that validator against the current validator
for the requested resource, and, if they match (see Section 16.1.3),
it responds with a special status code (usually, 304 (Not Modified))
and no message body. Otherwise, it returns a full response
(including message body). Thus, avoiding transmitting the full
response if the validator matches and avoiding an extra round trip if
it does not match.
In RTSP, a conditional request looks exactly the same as a normal
request for the same resource, except that it carries a special
header (which includes the validator) that implicitly turns the
method (usually DESCRIBE or SETUP) into a conditional.
The protocol includes both positive and negative senses of cache-
validating conditions. That is, it is possible to request that a
method be performed either if and only if a validator matches or if
and only if no validators match.
Note: a response that lacks a validator may still be cached, and
served from cache until it expires, unless this is explicitly
prohibited by a cache directive (see Section 18.11). However, a
cache cannot perform a conditional retrieval if it does not have a
validator for the resource, which means it will not be refreshable
after it expires.
Media streams that are being adapted based on the transport capacity
between the server and the cache make caching more difficult. A
server needs to consider how it views the caching of media streams
that it adapts and potentially instruct any caches not to cache such
streams.
16.1.1. Last-Modified Dates
The Last-Modified header (Section 18.27) value is often used as a
cache validator. In simple terms, a cache entry is considered to be
valid if the cache entry was created after the Last-Modified time.
16.1.2. Message Body Tag Cache Validators
The MTag response-header field-value, a message body tag, provides
for an "opaque" cache validator. This might allow more reliable
validation in situations where it is inconvenient to store
modification dates, where the one-second resolution of RTSP-date
values is not sufficient, or where the origin server wishes to avoid
certain paradoxes that might arise from the use of modification
dates.
Message body tags are described in Section 4.6
16.1.3. Weak and Strong Validators
Since both origin servers and caches will compare two validators to
decide if they represent the same or different entities, one normally
would expect that if the message body (i.e., the presentation
description) or any associated message body headers changes in any
way, then the associated validator would change as well. If this is
true, then this validator is a "strong validator". The Message body
(i.e., the presentation description) or any associated message body
headers is named an entity for a better understanding.
However, there might be cases when a server prefers to change the
validator only on semantically significant changes and not when
insignificant aspects of the entity change. A validator that does
not always change when the resource changes is a "weak validator".
Message body tags are normally strong validators, but the protocol
provides a mechanism to tag a message body tag as "weak". One can
think of a strong validator as one that changes whenever the bits of
an entity changes, while a weak value changes whenever the meaning of
an entity changes. Alternatively, one can think of a strong
validator as part of an identifier for a specific entity, while a
weak validator is part of an identifier for a set of semantically
equivalent entities.
Note: One example of a strong validator is an integer that is
incremented in stable storage every time an entity is changed.
An entity's modification time, if represented with one-second
resolution, could be a weak validator, since it is possible that
the resource might be modified twice during a single second.
Support for weak validators is optional. However, weak validators
allow for more efficient caching of equivalent objects.
A "use" of a validator is either when a client generates a request
and includes the validator in a validating header field or when a
server compares two validators.
Strong validators are usable in any context. Weak validators are
only usable in contexts that do not depend on exact equality of an
entity. For example, either kind is usable for a conditional
DESCRIBE of a full entity. However, only a strong validator is
usable for a subrange retrieval, since otherwise the client might end
up with an internally inconsistent entity.
Clients MAY issue DESCRIBE requests with either weak or strong
validators. Clients MUST NOT use weak validators in other forms of
requests.
The only function that RTSP defines on validators is comparison.
There are two validator comparison functions, depending on whether or
not the comparison context allows the use of weak validators:
o The strong comparison function: in order to be considered equal,
both validators MUST be identical in every way, and both MUST NOT
be weak.
o The weak comparison function: in order to be considered equal,
both validators MUST be identical in every way, but either or both
of them MAY be tagged as "weak" without affecting the result.
A message body tag is strong unless it is explicitly tagged as weak.
A Last-Modified time, when used as a validator in a request, is
implicitly weak unless it is possible to deduce that it is strong,
using the following rules:
o The validator is being compared by an origin server to the actual
current validator for the entity and,
o That origin server reliably knows that the associated entity did
not change more than once during the second covered by the
presented validator.
OR
o The validator is about to be used by a client in an If-Modified-
Since, because the client has a cache entry for the associated
entity, and
o That cache entry includes a Date value, which gives the time when
the origin server sent the original response, and
o The presented Last-Modified time is at least 60 seconds before the
Date value.
OR
o The validator is being compared by an intermediate cache to the
validator stored in its cache entry for the entity, and
o That cache entry includes a Date value, which gives the time when
the origin server sent the original response, and
o The presented Last-Modified time is at least 60 seconds before the
Date value.
This method relies on the fact that if two different responses were
sent by the origin server during the same second, but both had the
same Last-Modified time, then at least one of those responses would
have a Date value equal to its Last-Modified time. The arbitrary
60-second limit guards against the possibility that the Date and
Last-Modified values are generated from different clocks or at
somewhat different times during the preparation of the response. An
implementation MAY use a value larger than 60 seconds, if it is
believed that 60 seconds is too short.
If a client wishes to perform a subrange retrieval on a value for
which it has only a Last-Modified time and no opaque validator, it
MAY do this only if the Last-Modified time is strong in the sense
described here.
16.1.4. Rules for When to Use Message Body Tags and Last-Modified Dates
This document adopts a set of rules and recommendations for origin
servers, clients, and caches regarding when various validator types
ought to be used, and for what purposes.
RTSP origin servers:
o SHOULD send a message body tag validator unless it is not feasible
to generate one.
o MAY send a weak message body tag instead of a strong message body
tag, if performance considerations support the use of weak message
body tags, or if it is unfeasible to send a strong message body
tag.
o SHOULD send a Last-Modified value if it is feasible to send one,
unless the risk of a breakdown in semantic transparency that could
result from using this date in an If-Modified-Since header would
lead to serious problems.
In other words, the preferred behavior for an RTSP origin server is
to send both a strong message body tag and a Last-Modified value.
In order to be legal, a strong message body tag MUST change whenever
the associated entity value changes in any way. A weak message body
tag SHOULD change whenever the associated entity changes in a
semantically significant way.
Note: in order to provide semantically transparent caching, an
origin server MUST avoid reusing a specific strong message body
tag value for two different entities or reusing a specific weak
message body tag value for two semantically different entities.
Cache entries might persist for arbitrarily long periods,
regardless of expiration times, so it might be inappropriate to
expect that a cache will never again attempt to validate an entry
using a validator that it obtained at some point in the past.
RTSP clients:
o If a message body tag has been provided by the origin server, MUST
use that message body tag in any cache-conditional request (using
If-Match or If-None-Match).
o If only a Last-Modified value has been provided by the origin
server, SHOULD use that value in non-subrange cache-conditional
requests (using If-Modified-Since).
o If both a message body tag and a Last-Modified value have been
provided by the origin server, SHOULD use both validators in
cache-conditional requests.
An RTSP origin server, upon receiving a conditional request that
includes both a Last-Modified date (e.g., in an If-Modified-Since
header) and one or more message body tags (e.g., in an If-Match,
If-None-Match, or If-Range header field) as cache validators, MUST
NOT return a response status of 304 (Not Modified) unless doing so is
consistent with all of the conditional header fields in the request.
Note: The general principle behind these rules is that RTSP
servers and clients should transmit as much non-redundant
information as is available in their responses and requests. RTSP
systems receiving this information will make the most conservative
assumptions about the validators they receive.
16.1.5. Non-validating Conditionals
The principle behind message body tags is that only the service
author knows the semantics of a resource well enough to select an
appropriate cache validation mechanism, and the specification of any
validator comparison function more complex than octet equality would
open up a can of worms. Thus, comparisons of any other headers are
never used for purposes of validating a cache entry.
16.2. Invalidation after Updates or Deletions
The effect of certain methods performed on a resource at the origin
server might cause one or more existing cache entries to become non-
transparently invalid. That is, although they might continue to be
"fresh," they do not accurately reflect what the origin server would
return for a new request on that resource.
There is no way for RTSP to guarantee that all such cache entries are
marked invalid. For example, the request that caused the change at
the origin server might not have gone through the proxy where a cache
entry is stored. However, several rules help reduce the likelihood
of erroneous behavior.
In this section, the phrase "invalidate an entity" means that the
cache will either remove all instances of that entity from its
storage or mark these as "invalid" and in need of a mandatory
revalidation before they can be returned in response to a subsequent
request.
Some RTSP methods MUST cause a cache to invalidate an entity. This
is either the entity referred to by the Request-URI or by the
Location or Content-Location headers (if present). These methods
are:
o DESCRIBE
o SETUP
In order to prevent DoS attacks, an invalidation based on the URI in
a Location or Content-Location header MUST only be performed if the
host part is the same as in the Request-URI.
A cache that passes through requests for methods it does not
understand SHOULD invalidate any entities referred to by the Request-
URI.
17. Status Code Definitions
Where applicable, HTTP status codes (see Section 6 of [RFC7231]) are
reused. See Table 4 in Section 8.1 for a listing of which status
codes may be returned by which requests. All error messages, 4xx and
5xx, MAY return a body containing further information about the
error.
17.1. Informational 1xx
17.1.1. 100 Continue
The requesting agent SHOULD continue with its request. This interim
response is used to inform the requesting agent that the initial part
of the request has been received and has not yet been rejected by the
responding agent. The requesting agent SHOULD continue by sending
the remainder of the request or, if the request has already been
completed, continue to wait for a final response (see Section 10.4).
The responding agent MUST send a final response after the request has
been completed.
17.2. Success 2xx
This class of status code indicates that the agent's request was
successfully received, understood, and accepted.
17.2.1. 200 OK
The request has succeeded. The information returned with the
response is dependent on the method used in the request.
17.3. Redirection 3xx
The notation "3xx" indicates response codes from 300 to 399 inclusive
that are meant for redirection. We use the notation "3rr" to
indicate all 3xx codes used for redirection, i.e., excluding 304.
The 304 response code appears here, rather than a 2xx response code,
which would have been appropriate; 304 has also been used in RTSP 1.0
[RFC2326].
Within RTSP, redirection may be used for load-balancing or
redirecting stream requests to a server topologically closer to the
agent. Mechanisms to determine topological proximity are beyond the
scope of this specification.
A 3rr code MAY be used to respond to any request. The Location
header MUST be included in any 3rr response. It is RECOMMENDED that
they are used if necessary before a session is established, i.e., in
response to DESCRIBE or SETUP. However, in cases where a server is
not able to send a REDIRECT request to the agent, the server MAY need
to resort to using 3rr responses to inform an agent with an
established session about the need for redirecting the session. If a
3rr response is received for a request in relation to an established
session, the agent SHOULD send a TEARDOWN request for the session and
MAY reestablish the session using the resource indicated by the
Location.
If the Location header is used in a response, it MUST contain an
absolute URI pointing out the media resource the agent is redirected
to; the URI MUST NOT only contain the hostname.
In the event that an unknown 3rr status code is received, the agent
SHOULD behave as if a 302 response code had been received
(Section 17.3.3).
17.3.1. 300
The 300 response code is not used in RTSP 2.0.
17.3.2. 301 Moved Permanently
The requested resource is moved permanently and resides now at the
URI given by the Location header. The user agent SHOULD redirect
automatically to the given URI. This response MUST NOT contain a
message body. The Location header MUST be included in the response.
17.3.3. 302 Found
The requested resource resides temporarily at the URI given by the
Location header. This response is intended to be used for many types
of temporary redirects, e.g., load balancing. It is RECOMMENDED that
the server set the reason phrase to something more meaningful than
"Found" in these cases. The Location header MUST be included in the
response. The user agent SHOULD redirect automatically to the given
URI. This response MUST NOT contain a message body.
This example shows a client being redirected to a different server:
C->S: SETUP rtsp://example.com/fizzle/foo RTSP/2.0
CSeq: 2
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
Accept-Ranges: npt, smpte, clock
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 302 Try Other Server
CSeq: 2
Location: rtsp://s2.example.com:8001/fizzle/foo
17.3.4. 303 See Other
This status code MUST NOT be used in RTSP 2.0. However, it was
allowed in RTSP 1.0 [RFC2326].
17.3.5. 304 Not Modified
If the agent has performed a conditional DESCRIBE or SETUP (see
Sections 18.25 and 18.26) and the requested resource has not been
modified, the server SHOULD send a 304 response. This response MUST
NOT contain a message body.
The response MUST include the following header fields:
o Date
o MTag or Content-Location, if the headers would have been sent in a
200 response to the same request.
o Expires and Cache-Control if the field-value might differ from
that sent in any previous response for the same variant.
This response is independent for the DESCRIBE and SETUP requests.
That is, a 304 response to DESCRIBE does NOT imply that the resource
content is unchanged (only the session description) and a 304
response to SETUP does NOT imply that the resource description is
unchanged. The MTag and If-Match header (Section 18.24) may be used
to link the DESCRIBE and SETUP in this manner.
17.3.6. 305 Use Proxy
The requested resource MUST be accessed through the proxy given by
the Location header that MUST be included. The Location header
field-value gives the URI of the proxy. The recipient is expected to
repeat this single request via the proxy. 305 responses MUST only be
generated by origin servers.
17.4. Client Error 4xx
17.4.1. 400 Bad Request
The request could not be understood by the agent due to malformed
syntax. The agent SHOULD NOT repeat the request without
modifications. If the request does not have a CSeq header, the agent
MUST NOT include a CSeq in the response.
17.4.2. 401 Unauthorized
The request requires user authentication using the HTTP
authentication mechanism [RFC7235]. The usage of the error code is
defined in [RFC7235] and any applicable HTTP authentication scheme,
such as Digest [RFC7616]. The response is to include a WWW-
Authenticate header (Section 18.58) field containing a challenge
applicable to the requested resource. The agent can repeat the
request with a suitable Authorization header field. If the request
already included authorization credentials, then the 401 response
indicates that authorization has been refused for those credentials.
If the 401 response contains the same challenge as the prior
response, and the user agent has already attempted authentication at
least once, then the user SHOULD be presented the message body that
was given in the response, since that message body might include
relevant diagnostic information.
17.4.3. 402 Payment Required
This code is reserved for future use.
17.4.4. 403 Forbidden
The agent understood the request, but is refusing to fulfill it.
Authorization will not help, and the request SHOULD NOT be repeated.
If the agent wishes to make public why the request has not been
fulfilled, it SHOULD describe the reason for the refusal in the
message body. If the agent does not wish to make this information
available to the agent, the status code 404 (Not Found) can be used
instead.
17.4.5. 404 Not Found
The agent has not found anything matching the Request-URI. No
indication is given of whether the condition is temporary or
permanent. The 410 (Gone) status code SHOULD be used if the agent
knows, through some internally configurable mechanism, that an old
resource is permanently unavailable and has no forwarding address.
This status code is commonly used when the agent does not wish to
reveal exactly why the request has been refused, or when no other
response is applicable.
17.4.6. 405 Method Not Allowed
The method specified in the request is not allowed for the resource
identified by the Request-URI. The response MUST include an Allow
header containing a list of valid methods for the requested resource.
This status code is also to be used if a request attempts to use a
method not indicated during SETUP.
17.4.7. 406 Not Acceptable
The resource identified by the request is only capable of generating
response message bodies that have content characteristics not
acceptable according to the Accept headers sent in the request.
The response SHOULD include a message body containing a list of
available message body characteristics and location(s) from which the
user or user agent can choose the one most appropriate. The message
body format is specified by the media type given in the Content-Type
header field. Depending upon the format and the capabilities of the
user agent, selection of the most appropriate choice MAY be performed
automatically. However, this specification does not define any
standard for such automatic selection.
If the response could be unacceptable, a user agent SHOULD
temporarily stop receipt of more data and query the user for a
decision on further actions.
17.4.8. 407 Proxy Authentication Required
This code is similar to 401 (Unauthorized) (Section 17.4.2), but it
indicates that the client must first authenticate itself with the
proxy. The usage of this error code is defined in [RFC7235] and any
applicable HTTP authentication scheme, such as Digest [RFC7616]. The
proxy MUST return a Proxy-Authenticate header field (Section 18.34)
containing a challenge applicable to the proxy for the requested
resource.
17.4.9. 408 Request Timeout
The agent did not produce a request within the time that the agent
was prepared to wait. The agent MAY repeat the request without
modifications at any later time.
17.4.10. 410 Gone
The requested resource is no longer available at the server and the
forwarding address is not known. This condition is expected to be
considered permanent. If the server does not know, or has no
facility to determine, whether or not the condition is permanent, the
status code 404 (Not Found) SHOULD be used instead. This response is
cacheable unless indicated otherwise.
The 410 response is primarily intended to assist the task of
repository maintenance by notifying the recipient that the resource
is intentionally unavailable and that the server owners desire that
remote links to that resource be removed. Such an event is common
for limited-time, promotional services and for resources belonging to
individuals no longer working at the server's site. It is not
necessary to mark all permanently unavailable resources as "gone" or
to keep the mark for any length of time -- that is left to the
discretion of the owner of the server.
17.4.11. 412 Precondition Failed
The precondition given in one or more of the 'if-' request-header
fields evaluated to false when it was tested on the agent. See these
sections for the 'if-' headers: If-Match Section 18.24, If-Modified-
Since Section 18.25, and If-None-Match Section 18.26. This response
code allows the agent to place preconditions on the current resource
meta-information (header field data) and, thus, prevent the requested
method from being applied to a resource other than the one intended.
17.4.12. 413 Request Message Body Too Large
The agent is refusing to process a request because the request
message body is larger than the agent is willing or able to process.
The agent MAY close the connection to prevent the requesting agent
from continuing the request.
If the condition is temporary, the agent SHOULD include a Retry-After
header field to indicate that it is temporary and after what time the
requesting agent MAY try again.
17.4.13. 414 Request-URI Too Long
The responding agent is refusing to service the request because the
Request-URI is longer than the agent is willing to interpret. This
rare condition is only likely to occur when an agent has used a
request with long query information, when the agent has descended
into a URI "black hole" of redirection (e.g., a redirected URI prefix
that points to a suffix of itself), or when the agent is under attack
by an agent attempting to exploit security holes present in some
agents using fixed-length buffers for reading or manipulating the
Request-URI.
17.4.14. 415 Unsupported Media Type
The server is refusing to service the request because the message
body of the request is in a format not supported by the requested
resource for the requested method.
17.4.15. 451 Parameter Not Understood
The recipient of the request does not support one or more parameters
contained in the request. When returning this error message the
agent SHOULD return a message body containing the offending
parameter(s).
17.4.16. 452 Illegal Conference Identifier
This status code MUST NOT be used in RTSP 2.0. However, it was
allowed in RTSP 1.0 [RFC2326].
17.4.17. 453 Not Enough Bandwidth
The request was refused because there was insufficient bandwidth.
This may, for example, be the result of a resource reservation
failure.
17.4.18. 454 Session Not Found
The RTSP session identifier in the Session header is missing, is
invalid, or has timed out.
17.4.19. 455 Method Not Valid in This State
The agent cannot process this request in its current state. The
response MUST contain an Allow header to make error recovery
possible.
17.4.20. 456 Header Field Not Valid for Resource
The targeted agent could not act on a required request-header. For
example, if PLAY request contains the Range header field but the
stream does not allow seeking. This error message may also be used
for specifying when the time format in Range is impossible for the
resource. In that case, the Accept-Ranges header MUST be returned to
inform the agent of which formats are allowed.
17.4.21. 457 Invalid Range
The Range value given is out of bounds, e.g., beyond the end of the
presentation.
17.4.22. 458 Parameter Is Read-Only
The parameter to be set by SET_PARAMETER can be read but not
modified. When returning this error message, the sender SHOULD
return a message body containing the offending parameter(s).
17.4.23. 459 Aggregate Operation Not Allowed
The requested method may not be applied on the URI in question since
it is an aggregate (presentation) URI. The method may be applied on
a media URI.
17.4.24. 460 Only Aggregate Operation Allowed
The requested method may not be applied on the URI in question since
it is not an aggregate control (presentation) URI. The method may be
applied on the aggregate control URI.
17.4.25. 461 Unsupported Transport
The Transport field did not contain a supported transport
specification.
17.4.26. 462 Destination Unreachable
The data transmission channel could not be established because the
agent address could not be reached. This error will most likely be
the result of an agent attempt to place an invalid dest_addr
parameter in the Transport field.
17.4.27. 463 Destination Prohibited
The data transmission channel was not established because the server
prohibited access to the agent address. This error is most likely
the result of an agent attempt to redirect media traffic to another
destination with a dest_addr parameter in the Transport header.
17.4.28. 464 Data Transport Not Ready Yet
The data transmission channel to the media destination is not yet
ready for carrying data. However, the responding agent still expects
that the data transmission channel will be established at some point
in time. Note, however, that this may result in a permanent failure
like 462 (Destination Unreachable).
An example of when this error may occur is in the case in which a
client sends a PLAY request to a server prior to ensuring that the
TCP connections negotiated for carrying media data were successfully
established (in violation of this specification). The server would
use this error code to indicate that the requested action could not
be performed due to the failure of completing the connection
establishment.
17.4.29. 465 Notification Reason Unknown
This indicates that the client has received a PLAY_NOTIFY
(Section 13.5) with a Notify-Reason header (Section 18.32) unknown to
the client.
17.4.30. 466 Key Management Error
This indicates that there has been an error in a Key Management
function used in conjunction with a request. For example, usage of
Multimedia Internet KEYing (MIKEY) [RFC3830] according to
Appendix C.1.4.1 may result in this error.
17.4.31. 470 Connection Authorization Required
The secured connection attempt needs user or client authorization
before proceeding. The next hop's certificate is included in this
response in the Accept-Credentials header.
17.4.32. 471 Connection Credentials Not Accepted
When performing a secure connection over multiple connections, an
intermediary has refused to connect to the next hop and carry out the
request due to unacceptable credentials for the used policy.
17.4.33. 472 Failure to Establish Secure Connection
A proxy fails to establish a secure connection to the next-hop RTSP
agent. This is primarily caused by a fatal failure at the TLS
handshake, for example, due to the agent not accepting any cipher
suites.
17.5. Server Error 5xx
Response status codes beginning with the digit "5" indicate cases in
which the server is aware that it has erred or is incapable of
performing the request. The server SHOULD include a message body
containing an explanation of the error situation and whether it is a
temporary or permanent condition. User agents SHOULD display any
included message body to the user. These response codes are
applicable to any request method.
17.5.1. 500 Internal Server Error
The agent encountered an unexpected condition that prevented it from
fulfilling the request.
17.5.2. 501 Not Implemented
The agent does not support the functionality required to fulfill the
request. This is the appropriate response when the agent does not
recognize the request method and is not capable of supporting it for
any resource.
17.5.3. 502 Bad Gateway
The agent, while acting as a gateway or proxy, received an invalid
response from the upstream agent it accessed in attempting to fulfill
the request.
17.5.4. 503 Service Unavailable
The server is currently unable to handle the request due to a
temporary overloading or maintenance of the server. The implication
is that this is a temporary condition that will be alleviated after
some delay. If known, the length of the delay MAY be indicated in a
Retry-After header. If no Retry-After is given, the agent SHOULD
handle the response as it would for a 500 response. The agent MUST
honor the length, if given, in the Retry-After header.
Note: The existence of the 503 status code does not imply that
a server must use it when becoming overloaded. Some servers
may wish to simply refuse the transport connection.
The response scope is dependent on the request. If the request was
in relation to an existing RTSP session, the scope of the overload
response is to this individual RTSP session. If the request was not
session specific or intended to form an RTSP session, it applies to
the RTSP server identified by the hostname in the Request-URI.
17.5.5. 504 Gateway Timeout
The agent, while acting as a proxy, did not receive a timely response
from the upstream agent specified by the URI or some other auxiliary
server (e.g., DNS) that it needed to access in attempting to complete
the request.
17.5.6. 505 RTSP Version Not Supported
The agent does not support, or refuses to support, the RTSP version
that was used in the request message. The agent is indicating that
it is unable or unwilling to complete the request using the same
major version as the agent other than with this error message. The
response SHOULD contain a message body describing why that version is
not supported and what other protocols are supported by that agent.
17.5.7. 551 Option Not Supported
A feature tag given in the Require or the Proxy-Require fields was
not supported. The Unsupported header MUST be returned stating the
feature for which there is no support.
17.5.8. 553 Proxy Unavailable
The proxy is currently unable to handle the request due to a
temporary overloading or maintenance of the proxy. The implication
is that this is a temporary condition that will be alleviated after
some delay. If known, the length of the delay MAY be indicated in a
Retry-After header. If no Retry-After is given, the agent SHOULD
handle the response as it would for a 500 response. The agent MUST
honor the length, if given in the Retry-After header.
Note: The existence of the 553 status code does not imply that
a proxy must use it when becoming overloaded. Some proxies may
wish to simply refuse the connection.
The response scope is dependent on the Request. If the request was
in relation to an existing RTSP session, the scope of the overload
response is to this individual RTSP session. If the request was non-
session specific or intended to form an RTSP session, it applies to
all such requests to this proxy.
18. Header Field Definitions
+---------------+----------------+--------+---------+------+
| method | direction | object | acronym | Body |
+---------------+----------------+--------+---------+------+
| DESCRIBE | C -> S | P,S | DES | r |
| | | | | |
| GET_PARAMETER | C -> S, S -> C | P,S | GPR | R,r |
| | | | | |
| OPTIONS | C -> S, S -> C | P,S | OPT | |
| | | | | |
| PAUSE | C -> S | P,S | PSE | |
| | | | | |
| PLAY | C -> S | P,S | PLY | |
| | | | | |
| PLAY_NOTIFY | S -> C | P,S | PNY | R |
| | | | | |
| REDIRECT | S -> C | P,S | RDR | |
| | | | | |
| SETUP | C -> S | S | STP | |
| | | | | |
| SET_PARAMETER | C -> S, S -> C | P,S | SPR | R,r |
| | | | | |
| TEARDOWN | C -> S | P,S | TRD | |
| | | | | |
| | S -> C | P | TRD | |
+---------------+----------------+--------+---------+------+
This table is an overview of RTSP methods, their direction, and what
objects (P: presentation, S: stream) they operate on. "Body" denotes
if a method is allowed to carry body and in which direction; R =
request, r=response. Note: All error messages for statuses 4xx and
5xx are allowed to carry a body.
Table 8: Overview of RTSP Methods
The general syntax for header fields is covered in Section 5.2. This
section lists the full set of header fields along with notes on
meaning and usage. The syntax definitions for header fields are
present in Section 20.2.3. Examples of each header field are given.
Information about header fields in relation to methods and proxy
processing is summarized in Figures 2, 3, 4, and 5.
The "where" column describes the request and response types in which
the header field can be used. Values in this column are:
R: header field may only appear in requests;
r: header field may only appear in responses;
2xx, 4xx, etc.: numerical value or range indicates response codes
with which the header field can be used;
c: header field is copied from the request to the
response.
G: header field is a general-header and may be present
in both requests and responses.
Note: General headers do not always use the "G" value in the "where"
column. This is due to differences when the header may be applied in
requests compared to responses. When such differences exist, they
are expressed using two different rows: one with "where" being "R"
and one with it being "r".
The "proxy" column describes the operations a proxy may perform on a
header field. An empty proxy column indicates that the proxy MUST
NOT make any changes to that header, all allowed operations are
explicitly stated:
a: A proxy can add or concatenate the header field if not present.
m: A proxy can modify an existing header field value.
d: A proxy can delete a header field-value.
r: A proxy needs to be able to read the header field; thus, this
header field cannot be encrypted.
The rest of the columns relate to the presence of a header field in a
method. The method names when abbreviated, are according to Table 8:
c: Conditional; requirements on the header field depend on the
context of the message.
m: The header field is mandatory.
m*: The header field SHOULD be sent, but agents need to be prepared
to receive messages without that header field.
o: The header field is optional.
*: The header field MUST be present if the message body is not
empty. See Sections 18.17, 18.19 and 5.3 for details.
-: The header field is not applicable.
"Optional" means that an agent MAY include the header field in a
request or response. The agent behavior when receiving such headers
varies; for some, it may ignore the header field. In other cases, it
is a request to process the header. This is regulated by the method
and header descriptions. Examples of headers that require processing
are the Require and Proxy-Require header fields discussed in Sections
18.43 and 18.37. A "mandatory" header field MUST be present in a
request, and it MUST be understood by the agent receiving the
request. A mandatory response-header field MUST be present in the
response, and the header field MUST be understood by the processing
the response. "Not applicable" means that the header field MUST NOT
be present in a request. If one is placed in a request by mistake,
it MUST be ignored by the agent receiving the request. Similarly, a
header field labeled "not applicable" for a response means that the
agent MUST NOT place the header field in the response, and the agent
MUST ignore the header field in the response.
An RTSP agent MUST ignore extension headers that are not understood.
The From and Location header fields contain a URI. If the URI
contains a comma (') or semicolon (;), the URI MUST be enclosed in
double quotes ("). Any URI parameters are contained within these
quotes. If the URI is not enclosed in double quotes, any semicolon-
delimited parameters are header-parameters, not URI parameters.
+-------------------+------+------+----+----+-----+-----+-----+-----+
| Header |Where |Proxy |DES | OPT| STP | PLY | PSE | TRD |
+-------------------+------+------+----+----+-----+-----+-----+-----+
| Accept | R | | o | - | - | - | - | - |
| Accept- | R | rm | o | o | o | o | o | o |
| Credentials | | | | | | | | |
| Accept-Encoding | R | r | o | - | - | - | - | - |
| Accept-Language | R | r | o | - | - | - | - | - |
| Accept-Ranges | G | r | - | - | m | - | - | - |
| Accept-Ranges | 456 | r | - | - | - | m | - | - |
| Allow | r | am | c | c | c | - | - | - |
| Allow | 405 | am | m | m | m | m | m | m |
| Authentication- | r | | o | o | o | o | o | o/- |
| Info | | | | | | | | |
| Authorization | R | | o | o | o | o | o | o/- |
| Bandwidth | R | | o | o | o | o | - | - |
| Blocksize | R | | o | - | o | o | - | - |
| Cache-Control | G | r | o | - | o | - | - | - |
| Connection | G | ad | o | o | o | o | o | o |
| Connection- | 470, | ar | o | o | o | o | o | o |
| Credentials | 407 | | | | | | | |
| Content-Base | r | | o | - | - | - | - | - |
| Content-Base | 4xx, | | o | o | o | o | o | o |
| | 5xx | | | | | | | |
| Content-Encoding | R | r | - | - | - | - | - | - |
| Content-Encoding | r | r | o | - | - | - | - | - |
| Content-Encoding | 4xx, | r | o | o | o | o | o | o |
| | 5xx | | | | | | | |
| Content-Language | R | r | - | - | - | - | - | - |
| Content-Language | r | r | o | - | - | - | - | - |
| Content-Language | 4xx, | r | o | o | o | o | o | o |
| | 5xx | | | | | | | |
| Content-Length | r | r | * | - | - | - | - | - |
| Content-Length | 4xx, | r | * | * | * | * | * | * |
| | 5xx | | | | | | | |
| Content-Location | r | r | o | - | - | - | - | - |
| Content-Location | 4xx, | r | o | o | o | o | o | o |
| | 5xx | | | | | | | |
| Content-Type | r | r | * | - | - | - | - | - |
| Content-Type | 4xx, | ar | * | * | * | * | * | * |
| | 5xx | | | | | | | |
| CSeq | Gc | rm | m | m | m | m | m | m |
| Date | G | am | o/*| o/*| o/* | o/* | o/* | o/* |
| Expires | r | r | o | - | o | - | - | - |
| From | R | r | o | o | o | o | o | o |
| If-Match | R | r | - | - | o | - | - | - |
| If-Modified-Since | R | r | o | - | o | - | - | - |
| If-None-Match | R | r | o | - | o | - | - | - |
| Last-Modified | r | r | o | - | o | - | - | - |
| Location | 3rr | | m | m | m | m | m | m |
+-------------------+------+------+----+----+-----+-----+-----+-----+
| Header |Where |Proxy |DES | OPT| STP | PLY | PSE | TRD |
+-------------------+------+------+----+----+-----+-----+-----+-----+
Figure 2: Overview of RTSP Header Fields (A-L) Related to Methods
DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN
+------------------+---------+-----+----+----+----+-----+-----+-----+
| Header | Where |Proxy|DES |OPT |STP | PLY | PSE | TRD |
+------------------+---------+-----+----+----+----+-----+-----+-----+
| Media-Properties | r | | - | - | m | o | o | - |
| Media-Range | r | | - | - | c | c | c | - |
| MTag | r | r | o | - | o | - | - | - |
| Pipelined- | G | amd | - | o | o | o | o | o |
| Requests | | r | | | | | | |
| Proxy- | 407 | amr | m | m | m | m | m | m |
| Authenticate | | | | | | | | |
| Proxy- | r | amd | o | o | o | o | o | o/- |
| Authentication- | | r | | | | | | |
| Info | | | | | | | | |
| Proxy- | R | rd | o | o | o | o | o | o |
| Authorization | | | | | | | | |
| Proxy-Require | R | ar | o | o | o | o | o | o |
| Proxy-Require | r | r | c | c | c | c | c | c |
| Proxy-Supported | R | amr | c | c | c | c | c | c |
| Proxy-Supported | r | | c | c | c | c | c | c |
| Public | r | amr | - | m | - | - | - | - |
| Public | 501 | amr | m | m | m | m | m | m |
| Range | R | | - | - | - | o | - | - |
| Range | r | | - | - | c | m | m | - |
| Referrer | R | | o | o | o | o | o | o |
| Request-Status | R | | - | - | - | - | - | - |
| Require | R | | o | o | o | o | o | o |
| Retry-After | 3rr,503 | | o | o | o | o | o | - |
| | ,553 | | | | | | | |
| Retry-After | 413 | | o | - | - | - | - | - |
| RTP-Info | r | | - | - | c | c | - | - |
| Scale | R | r | - | - | - | o | - | - |
| Scale | r | amr | - | - | c | c | c | - |
| Seek-Style | R | | - | - | - | o | - | - |
| Seek-Style | r | | - | - | - | m | - | - |
| Server | R | r | - | o | - | - | - | o |
| Server | r | r | o | o | o | o | o | o |
| Session | R | r | - | o | o | m | m | m |
| Session | r | r | - | c | m | m | m | o |
| Speed | R | admr| - | - | - | o | - | - |
| Speed | r | admr| - | - | - | c | - | - |
| Supported | R | r | o | o | o | o | o | o |
| Supported | r | r | c | c | c | c | c | c |
| Terminate-Reason | R | r | - | - | - | - | - | -/o |
| Timestamp | R | admr| o | o | o | o | o | o |
| Timestamp | c | admr| m | m | m | m | m | m |
| Transport | G | mr | - | - | m | - | - | - |
| Unsupported | r | | c | c | c | c | c | c |
| User-Agent | R | | m* | m* | m* | m* | m* | m* |
| Via | R | amr | c | c | c | c | c | c |
| Via | r | amr | c | c | c | c | c | c |
| WWW-Authenticate | 401 | | m | m | m | m | m | m |
+------------------+---------+-----+----+----+----+-----+-----+-----+
| Header | Where |Proxy|DES |OPT |STP | PLY | PSE | TRD |
+------------------+---------+-----+----+----+----+-----+-----+-----+
Figure 3: Overview of RTSP Header Fields (M-W) Related to Methods
DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN
+---------------------------+-------+-------+-----+-----+-----+-----+
| Header | Where | Proxy | GPR | SPR | RDR | PNY |
+---------------------------+-------+-------+-----+-----+-----+-----+
| Accept-Credentials | R | rm | o | o | o | - |
| Accept-Encoding | R | r | o | o | o | - |
| Accept-Language | R | r | o | o | o | - |
| Accept-Ranges | G | rm | o | - | - | - |
| Allow | 405 | amr | m | m | m | m |
| Authentication-Info | r | | o/- | o/- | - | - |
| Authorization | R | | o | o | o | - |
| Bandwidth | R | | - | o | - | - |
| Blocksize | R | | - | o | - | - |
| Cache-Control | G | r | o | o | - | - |
| Connection | G | | o | o | o | o |
| Connection-Credentials | 470, | ar | o | o | o | - |
| | 407 | | | | | |
| Content-Base | R | | o | o | - | o |
| Content-Base | r | | o | o | - | - |
| Content-Base | 4xx, | | o | o | o | o |
| | 5xx | | | | | |
| Content-Encoding | R | r | o | o | - | o |
| Content-Encoding | r | r | o | o | - | - |
| Content-Encoding | 4xx, | r | o | o | o | o |
| | 5xx | | | | | |
| Content-Language | R | r | o | o | - | o |
| Content-Language | r | r | o | o | - | - |
| Content-Language | 4xx, | r | o | o | o | o |
| | 5xx | | | | | |
| Content-Length | R | r | * | * | - | * |
| Content-Length | r | r | * | * | - | - |
| Content-Length | 4xx, | r | * | * | * | * |
| | 5xx | | | | | |
| Content-Location | R | | o | o | - | o |
| Content-Location | r | | o | o | - | - |
| Content-Location | 4xx, | | o | o | o | o |
| | 5xx | | | | | |
| Content-Type | R | | * | * | - | * |
| Content-Type | r | | * | * | - | - |
| Content-Type | 4xx, | | * | * | * | * |
| | 5xx | | | | | |
| CSeq | R,c | mr | m | m | m | m |
| Date | R | a | o/* | o/* | m | o/* |
| Date | r | am | o/* | o/* | o/* | o/* |
| Expires | r | r | - | - | - | - |
| From | R | r | o | o | o | - |
| If-Match | R | r | - | - | - | - |
| If-Modified-Since | R | am | o | - | - | - |
| If-None-Match | R | am | o | - | - | - |
| Last-Modified | R | r | - | - | - | - |
| Last-Modified | r | r | o | - | - | - |
| Location | 3rr | | m | m | m | - |
| Location | R | | - | - | m | - |
+---------------------------+-------+-------+-----+-----+-----+-----+
| Header | Where | Proxy | GPR | SPR | RDR | PNY |
+---------------------------+-------+-------+-----+-----+-----+-----+
Figure 4: Overview of RTSP Header Fields (A-L) Related to Methods
GET_PARAMETER, SET_PARAMETER, REDIRECT, and PLAY_NOTIFY
+---------------------------+---------+-------+-----+-----+-----+-----+
| Header | Where | Proxy | GPR | SPR | RDR | PNY |
+---------------------------+---------+-------+-----+-----+-----+-----+
| Media-Properties | R | amr | o | - | - | c |
| Media-Properties | r | mr | c | - | - | - |
| Media-Range | R | | o | - | - | c |
| Media-Range | r | | c | - | - | - |
| MTag | r | r | o | - | - | - |
| Notify-Reason | R | | - | - | - | m |
| Pipelined-Requests | R | amdr | o | o | - | - |
| Proxy-Authenticate | 407 | amdr | m | m | m | - |
| Proxy-Authentication-Info | r | amdr | o/- | o/- | - | - |
| Proxy-Authorization | R | amdr | o | o | o | - |
| Proxy-Require | R | ar | o | o | o | - |
| Proxy-Supported | R | amr | c | c | c | - |
| Proxy-Supported | r | | c | c | c | - |
| Public | 501 | admr | m | m | m | - |
| Range | R | | o | - | - | m |
| Range | r | | c | - | - | - |
| Referrer | R | | o | o | o | - |
| Request-Status | R | mr | - | - | - | c |
| Require | R | r | o | o | o | o |
| Retry-After | 3rr,503,| | o | o | - | - |
| | 553 | | | | | |
| Retry-After | 413 | | o | o | - | - |
| RTP-Info | R | r | o | - | - | C |
| RTP-Info | r | r | c | - | - | - |
| Scale | G | | c | - | c | c |
| Seek-Style | G | | - | - | - | - |
| Server | R | r | o | o | o | o |
| Server | r | r | o | o | - | - |
| Session | R | r | o | o | o | m |
| Session | r | r | c | c | o | m |
| Speed | G | | - | - | - | - |
| Supported | R | r | o | o | o | - |
| Supported | r | r | c | c | c | - |
| Terminate-Reason | R | r | - | - | m | - |
| Timestamp | R | adrm | o | o | o | o |
| Timestamp | c | adrm | m | m | m | m |
| Transport | G | mr | - | - | - | - |
| Unsupported | r | arm | c | c | c | c |
| User-Agent | R | r | m* | m* | - | - |
| User-Agent | r | r | m* | m* | m* | m* |
| Via | R | amr | c | c | c | c |
| Via | r | amr | c | c | c | c |
| WWW-Authenticate | 401 | | m | m | m | - |
+---------------------------+---------+-------+-----+-----+-----+-----+
| Header | Where | Proxy | GPR | SPR | RDR | PNY |
+---------------------------+---------+-------+-----+-----+-----+-----+
Figure 5: Overview of RTSP Header Fields (M-W) Related to Methods
GET_PARAMETER, SET_PARAMETER, REDIRECT, and PLAY_NOTIFY
18.1. Accept
The Accept request-header field can be used to specify certain
presentation description and parameter media types [RFC6838] that are
acceptable for the response to the DESCRIBE request.
See Section 20.2.3 for the syntax.
The asterisk "*" character is used to group media types into ranges,
with "*/*" indicating all media types and "type/*" indicating all
subtypes of that type. The range MAY include media type parameters
that are generally applicable to that range.
Each media type or range MAY be followed by one or more accept-
params, beginning with the "q" parameter to indicate a relative
quality factor. The first "q" parameter (if any) separates the media
type or range's parameters from the accept-params. Quality factors
allow the user or user agent to indicate the relative degree of
preference for that media type, using the qvalue scale from 0 to 1
(Section 5.3.1 of [RFC7231]). The default value is q=1.
Example of use:
Accept: application/example ;q=0.7, application/sdp
Indicates that the requesting agent prefers the media type
application/sdp through the default 1.0 rating but also accepts the
application/example media type with a 0.7 quality rating.
If no Accept header field is present, then it is assumed that the
client accepts all media types. If an Accept header field is
present, and if the server cannot send a response that is acceptable
according to the combined Accept field-value, then the server SHOULD
send a 406 (Not Acceptable) response.
18.2. Accept-Credentials
The Accept-Credentials header is a request-header used to indicate to
any trusted intermediary how to handle further secured connections to
proxies or servers. It MUST NOT be included in server-to-client
requests. See Section 19 for the usage of this header
In a request, the header MUST contain the method (User, Proxy, or
Any) for approving credentials selected by the requester. The method
MUST NOT be changed by any proxy, unless it is "Proxy" when a proxy
MAY change it to "user" to take the role of user approving each
further hop. If the method is "User", the header contains zero or
more of the credentials that the client accepts. The header may
contain zero credentials in the first RTSP request to an RTSP server
via a proxy when using the "User" method. This is because the client
has not yet received any credentials to accept. Each credential MUST
consist of one URI identifying the proxy or server, the hash
algorithm identifier, and the hash over that agent's ASN.1 DER-
encoded certificate [RFC5280] in Base64, according to Section 4 of
[RFC4648] and where the padding bits are set to zero. All RTSP
clients and proxies MUST implement the SHA-256 [FIPS180-4] algorithm
for computation of the hash of the DER-encoded certificate. The
SHA-256 algorithm is identified by the token "sha-256".
The intention of allowing for other hash algorithms is to enable the
future retirement of algorithms that are not implemented somewhere
other than here. Thus, the definition of future algorithms for this
purpose is intended to be extremely limited. A feature tag can be
used to ensure that support for the replacement algorithm exists.
Example:
Accept-Credentials:User
"rtsps://proxy2.example.com/";sha-256;exaIl9VMbQMOFGClx5rXnPJKVNI=,
"rtsps://server.example.com/";sha-256;lurbjj5khhB0NhIuOXtt4bBRH1M=
18.3. Accept-Encoding
The Accept-Encoding request-header field is similar to Accept, but it
restricts the content-codings (see Section 18.15), i.e.,
transformation codings of the message body, such as gzip compression,
that are acceptable in the response.
A server tests whether a content-coding is acceptable, according to
an Accept-Encoding field, using these rules:
1. If the content-coding is one of the content-codings listed in the
Accept-Encoding field, then it is acceptable, unless it is
accompanied by a qvalue of 0. (As defined in Section 5.3.1 of
[RFC7231], a qvalue of 0 means "not acceptable.")
2. The special "*" symbol in an Accept-Encoding field matches any
available content-coding not explicitly listed in the header
field.
3. If multiple content-codings are acceptable, then the acceptable
content-coding with the highest non-zero qvalue is preferred.
4. The "identity" content-coding is always acceptable, i.e., no
transformation at all, unless specifically refused because the
Accept-Encoding field includes "identity;q=0" or because the
field includes "*;q=0" and does not explicitly include the
"identity" content-coding. If the Accept-Encoding field-value is
empty, then only the "identity" encoding is acceptable.
If an Accept-Encoding field is present in a request, and if the
server cannot send a response that is acceptable according to the
Accept-Encoding header, then the server SHOULD send an error response
with the 406 (Not Acceptable) status code.
If no Accept-Encoding field is present in a request, the server MAY
assume that the client will accept any content-coding. In this case,
if "identity" is one of the available content-codings, then the
server SHOULD use the "identity" content-coding, unless it has
additional information that a different content-coding is meaningful
to the client.
18.4. Accept-Language
The Accept-Language request-header field is similar to Accept, but
restricts the set of natural languages that are preferred as a
response to the request. Note that the language specified applies to
the presentation description (response message body) and any reason
phrases, but not the media content.
A language tag identifies a natural language spoken, written, or
otherwise conveyed by human beings for communication of information
to other human beings. Computer languages are explicitly excluded.
The syntax and registry of RTSP 2.0 language tags are the same as
those defined by [RFC5646].
Each language-range MAY be given an associated quality value that
represents an estimate of the user's preference for the languages
specified by that range. The quality value defaults to "q=1". For
example:
Accept-Language: da, en-gb;q=0.8, en;q=0.7
would mean: "I prefer Danish, but will accept British English and
other types of English." A language-range matches a language tag if
it exactly equals the full tag or if it exactly equals a prefix of
the tag, i.e., the primary-tag in the ABNF, such that the character
following primary-tag is "-". The special range "*", if present in
the Accept-Language field, matches every tag not matched by any other
range present in the Accept-Language field.
Note: This use of a prefix matching rule does not imply that
language tags are assigned to languages in such a way that it is
always true that if a user understands a language with a certain
tag, then this user will also understand all languages with tags
for which this tag is a prefix. The prefix rule simply allows the
use of prefix tags if this is the case.
In the process of selecting a language, each language tag is assigned
a qualification factor, i.e., if a language being supported by the
client is actually supported by the server and what "preference"
level the language achieves. The quality value (q-value) of the
longest language-range in the field that matches the language tag is
assigned as the qualification factor for a particular language tag.
If no language-range in the field matches the tag, the language
qualification factor assigned is 0. If no Accept-Language header is
present in the request, the server SHOULD assume that all languages
are equally acceptable. If an Accept-Language header is present,
then all languages that are assigned a qualification factor greater
than 0 are acceptable.
18.5. Accept-Ranges
The Accept-Ranges general-header field allows indication of the
format supported in the Range header. The client MUST include the
header in SETUP requests to indicate which formats are acceptable
when received in PLAY and PAUSE responses and REDIRECT requests. The
server MUST include the header in SETUP responses and 456 (Header
Field Not Valid for Resource) error responses to indicate the formats
supported for the resource indicated by the Request-URI. The header
MAY be included in GET_PARAMETER request and response pairs. The
GET_PARAMETER request MUST contain a Session header to identify the
session context the request is related to. The requester and
responder will indicate their capabilities regarding Range formats
respectively.
Accept-Ranges: npt, smpte, clock
The syntax is defined in Section 20.2.3.
18.6. Allow
The Allow message body header field lists the methods supported by
the resource identified by the Request-URI. The purpose of this
field is to inform the recipient of the complete set of valid methods
associated with the resource. An Allow header field MUST be present
in a 405 (Method Not Allowed) response. The Allow header MUST also
be present in all OPTIONS responses where the content of the header
will not include exactly the same methods as listed in the Public
header.
The Allow message body header MUST also be included in SETUP and
DESCRIBE responses, if the methods allowed for the resource are
different from the complete set of methods defined in this memo.
Example of use:
Allow: SETUP, PLAY, SET_PARAMETER, DESCRIBE
18.7. Authentication-Info
The Authentication-Info response-header is used by the server to
communicate some information regarding the successful HTTP
authentication [RFC7235] in the response message. The definition of
the header is in [RFC7615], and any applicable HTTP authentication
schemes appear in other RFCs, such as Digest [RFC7616]. This header
MUST only be used in response messages related to client to server
requests.
18.8. Authorization
An RTSP client that wishes to authenticate itself with a server using
the authentication mechanism from HTTP [RFC7235], usually (but not
necessarily) after receiving a 401 response, does so by including an
Authorization request-header field with the request. The
Authorization field-value consists of credentials containing the
authentication information of the user agent for the realm of the
resource being requested. The definition of the header is in
[RFC7235], and any applicable HTTP authentication schemes appear in
other RFCs, such as Digest [RFC7616] and Basic [RFC7617]. This
header MUST only be used in client-to-server requests.
If a request is authenticated and a realm specified, the same
credentials SHOULD be valid for all other requests within this realm
(assuming that the authentication scheme itself does not require
otherwise, such as credentials that vary according to a challenge
value or using synchronized clocks). Each client-to-server request
MUST be individually authorized by including the Authorization header
with the information.
When a shared cache (see Section 16) receives a request containing an
Authorization field, it MUST NOT return the corresponding response as
a reply to any other request, unless one of the following specific
exceptions holds:
1. If the response includes the "max-age" cache directive, the cache
MAY use that response in replying to a subsequent request. But
(if the specified maximum age has passed) a proxy cache MUST
first revalidate it with the origin server, using the request-
headers from the new request to allow the origin server to
authenticate the new request. (This is the defined behavior for
max-age.) If the response includes "max-age=0", the proxy MUST
always revalidate it before reusing it.
2. If the response includes the "must-revalidate" cache-control
directive, the cache MAY use that response in replying to a
subsequent request. But if the response is stale, all caches
MUST first revalidate it with the origin server, using the
request-headers from the new request to allow the origin server
to authenticate the new request.
3. If the response includes the "public" cache directive, it MAY be
returned in reply to any subsequent request.
18.9. Bandwidth
The Bandwidth request-header field describes the estimated bandwidth
available to the client, expressed as a positive integer and measured
in kilobits per second. The bandwidth available to the client may
change during an RTSP session, e.g., due to mobility, congestion,
etc.
Clients may not be able to accurately determine the available
bandwidth, for example, because the first hop is not a bottleneck.
Such a case is when the local area network (LAN) is not the
bottleneck, instead the LAN's Internet access link is, if the server
is not in the same LAN. Thus, link speeds of WLAN or Ethernet
networks are normally not a basis for estimating the available
bandwidth. Cellular devices or other devices directly connected to a
modem or connection-enabling device may more accurately estimate the
bottleneck bandwidth and what is a reasonable share of it for RTSP-
controlled media. The client will also need to take into account
other traffic sharing the bottleneck. For example, by only assigning
a certain fraction to RTSP and its media streams. It is RECOMMENDED
that only clients that have accurate and explicit information about
bandwidth bottlenecks use this header.
This header is not a substitute for proper congestion control. It is
only a method providing an initial estimate and coarsely determines
if the selected content can be delivered at all.
Example:
Bandwidth: 62360
18.10. Blocksize
The Blocksize request-header field is sent from the client to the
media server asking the server for a particular media packet size.
This packet size does not include lower-layer headers such as IP,
UDP, or RTP. The server is free to use a blocksize that is lower
than the one requested. The server MAY truncate this packet size to
the closest multiple of the minimum, media-specific block size or
override it with the media-specific size, if necessary. The block
size MUST be a positive decimal number measured in octets. The
server only returns an error (4xx) if the value is syntactically
invalid.
18.11. Cache-Control
The Cache-Control general-header field is used to specify directives
that MUST be obeyed by all caching mechanisms along the request/
response chain.
Cache directives MUST be passed through by a proxy or gateway
application, regardless of their significance to that application,
since the directives may be applicable to all recipients along the
request/response chain. It is not possible to specify a cache-
directive for a specific cache.
Cache-Control should only be specified in a DESCRIBE, GET_PARAMETER,
SET_PARAMETER, and SETUP request and its response. Note: Cache-
Control does not govern only the caching of responses for the RTSP
messages, instead it also applies to the media stream identified by
the SETUP request. The RTSP requests are generally not cacheable;
for further information, see Section 16. Below are the descriptions
of the cache directives that can be included in the Cache-Control
header.
no-cache: Indicates that the media stream or RTSP response MUST NOT
be cached anywhere. This allows an origin server to prevent
caching even by caches that have been configured to return
stale responses to client requests. Note: there is no security
function preventing the caching of content.
public: Indicates that the media stream or RTSP response is
cacheable by any cache.
private: Indicates that the media stream or RTSP response is
intended for a single user and MUST NOT be cached by a shared
cache. A private (non-shared) cache may cache the media
streams.
no-transform: An intermediate cache (proxy) may find it useful to
convert the media type of a certain stream. A proxy might, for
example, convert between video formats to save cache space or
to reduce the amount of traffic on a slow link. Serious
operational problems may occur, however, when these
transformations have been applied to streams intended for
certain kinds of applications. For example, applications for
medical imaging, scientific data analysis and those using end-
to-end authentication all depend on receiving a stream that is
bit-for-bit identical to the original media stream or RTSP
response. Therefore, if a response includes the no-transform
directive, an intermediate cache or proxy MUST NOT change the
encoding of the stream or response. Unlike HTTP, RTSP does not
provide for partial transformation at this point, e.g.,
allowing translation into a different language.
only-if-cached: In some cases, such as times of extremely poor
network connectivity, a client may want a cache to return only
those media streams or RTSP responses that it currently has
stored and not to receive these from the origin server. To do
this, the client may include the only-if-cached directive in a
request. If the cache receives this directive, it SHOULD
either respond using a cached media stream or response that is
consistent with the other constraints of the request or respond
with a 504 (Gateway Timeout) status. However, if a group of
caches is being operated as a unified system with good internal
connectivity, such a request MAY be forwarded within that group
of caches.
max-stale: Indicates that the client is willing to accept a media
stream or RTSP response that has exceeded its expiration time.
If max-stale is assigned a value, then the client is willing to
accept a response that has exceeded its expiration time by no
more than the specified number of seconds. If no value is
assigned to max-stale, then the client is willing to accept a
stale response of any age.
min-fresh: Indicates that the client is willing to accept a media
stream or RTSP response whose freshness lifetime is no less
than its current age plus the specified time in seconds. That
is, the client wants a response that will still be fresh for at
least the specified number of seconds.
must-revalidate: When the must-revalidate directive is present in a
SETUP response received by a cache, that cache MUST NOT use the
cache entry after it becomes stale to respond to a subsequent
request without first revalidating it with the origin server.
That is, the cache is required to do an end-to-end revalidation
every time, if, based solely on the origin server's Expires,
the cached response is stale.
proxy-revalidate: The proxy-revalidate directive has the same
meaning as the must-revalidate directive, except that it does
not apply to non-shared user agent caches. It can be used on a
response to an authenticated request to permit the user's cache
to store and later return the response without needing to
revalidate it (since it has already been authenticated once by
that user), while still requiring proxies that service many
users to revalidate each time (in order to make sure that each
user has been authenticated). Note that such authenticated
responses also need the "public" cache directive in order to
allow them to be cached at all.
max-age: When an intermediate cache is forced, by means of a max-
age=0 directive, to revalidate its own cache entry, and the
client has supplied its own validator in the request, the
supplied validator might differ from the validator currently
stored with the cache entry. In this case, the cache MAY use
either validator in making its own request without affecting
semantic transparency.
However, the choice of validator might affect performance. The
best approach is for the intermediate cache to use its own
validator when making its request. If the server replies with
304 (Not Modified), then the cache can return its now validated
copy to the client with a 200 (OK) response. If the server
replies with a new message body and cache validator, however,
the intermediate cache can compare the returned validator with
the one provided in the client's request, using the strong
comparison function. If the client's validator is equal to the
origin server's, then the intermediate cache simply returns 304
(Not Modified). Otherwise, it returns the new message body
with a 200 (OK) response.
18.12. Connection
The Connection general-header field allows the sender to specify
options that are desired for that particular connection. It MUST NOT
be communicated by proxies over further connections.
RTSP 2.0 proxies MUST parse the Connection header field before a
message is forwarded and, for each connection-token in this field,
remove any header field(s) from the message with the same name as the
connection-token. Connection options are signaled by the presence of
a connection-token in the Connection header field, not by any
corresponding additional header field(s), since the additional header
field may not be sent if there are no parameters associated with that
connection option.
Message headers listed in the Connection header MUST NOT include end-
to-end headers, such as Cache-Control.
RTSP 2.0 defines the "close" connection option for the sender to
signal that the connection will be closed after completion of the
response. For example, "Connection: close in either the request or
the response-header fields" indicates that the connection SHOULD NOT
be considered "persistent" (Section 10.2) after the current request/
response is complete.
The use of the connection option "close" in RTSP messages SHOULD be
limited to error messages when the server is unable to recover and
therefore sees it necessary to close the connection. The reason
being that the client has the choice of continuing using a connection
indefinitely, as long as it sends valid messages.
18.13. Connection-Credentials
The Connection-Credentials response-header is used to carry the chain
of credentials for any next hop that needs to be approved by the
requester. It MUST only be used in server-to-client responses.
The Connection-Credentials header in an RTSP response MUST, if
included, contain the credential information (in the form of a list
of certificates providing the chain of certification) of the next hop
to which an intermediary needs to securely connect. The header MUST
include the URI of the next hop (proxy or server) and a
Base64-encoded (according to Section 4 of [RFC4648] and where the
padding bits are set to zero) binary structure containing a sequence
of DER-encoded X.509v3 certificates [RFC5280].
The binary structure starts with the number of certificates
(NR_CERTS) included as a 16-bit unsigned integer. This is followed
by an NR_CERTS number of 16-bit unsigned integers providing the size,
in octets, of each DER-encoded certificate. This is followed by an
NR_CERTS number of DER-encoded X.509v3 certificates in a sequence
(chain). This format is exemplified in Figure 6. The certificate of
the proxy or server must come first in the structure. Each following
certificate must directly certify the one preceding it. Because
certificate validation requires that root keys be distributed
independently, the self-signed certificate that specifies the root
certificate authority may optionally be omitted from the chain, under
the assumption that the remote end must already possess it in order
to validate it in any case.
Example:
Connection-Credentials:"rtsps://proxy2.example.com/";MIIDNTCC...
Where MIIDNTCC... is a Base64 encoding of the following structure:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Number of certificates | Size of certificate #1 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Size of certificate #2 | Size of certificate #3 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: DER Encoding of Certificate #1 :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: DER Encoding of Certificate #2 :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: DER Encoding of Certificate #3 :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 6: Format Example of Connection-Credentials Header Certificate
18.14. Content-Base
The Content-Base message body header field may be used to specify the
base URI for resolving relative URIs within the message body.
Content-Base: rtsp://media.example.com/movie/twister/
If no Content-Base field is present, the base URI of a message body
is defined by either its Content-Location (if that Content-Location
URI is an absolute URI) or the URI used to initiate the request, in
that order of precedence. Note, however, that the base URI of the
contents within the message body may be redefined within that message
body.
18.15. Content-Encoding
The Content-Encoding message body header field is used as a modifier
of the media-type. When present, its value indicates what additional
content-codings have been applied to the message body, and thus what
decoding mechanisms must be applied in order to obtain the media-type
referenced by the Content-Type header field. Content-Encoding is
primarily used to allow a document to be compressed without losing
the identity of its underlying media type.
The content-coding is a characteristic of the message body identified
by the Request-URI. Typically, the message body is stored with this
encoding and is only decoded before rendering or analogous usage.
However, an RTSP proxy MAY modify the content-coding if the new
coding is known to be acceptable to the recipient, unless the "no-
transform" cache directive is present in the message.
If the content-coding of a message body is not "identity", then the
message MUST include a Content-Encoding message body header that
lists the non-identity content-coding(s) used.
If the content-coding of a message body in a request message is not
acceptable to the origin server, the server SHOULD respond with a
status code of 415 (Unsupported Media Type).
If multiple encodings have been applied to a message body, the
content-codings MUST be listed in the order in which they were
applied, first to last from left to right. Additional information
about the encoding parameters MAY be provided by other header fields
not defined by this specification.
18.16. Content-Language
The Content-Language message body header field describes the natural
language(s) of the intended audience for the enclosed message body.
Note that this might not be equivalent to all the languages used
within the message body.
Language tags are mentioned in Section 18.4. The primary purpose of
Content-Language is to allow a user to identify and differentiate
entities according to the user's own preferred language. Thus, if
the body content is intended only for a Danish-literate audience, the
appropriate field is
Content-Language: da
If no Content-Language is specified, the default is that the content
is intended for all language audiences. This might mean that the
sender does not consider it to be specific to any natural language or
that the sender does not know for which language it is intended.
Multiple languages MAY be listed for content that is intended for
multiple audiences. For example, a rendition of the "Treaty of
Waitangi", presented simultaneously in the original Maori and English
versions, would call for
Content-Language: mi, en
However, just because multiple languages are present within a message
body does not mean that it is intended for multiple linguistic
audiences. An example would be a beginner's language primer, such as
"A First Lesson in Latin", which is clearly intended to be used by an
English-literate audience. In this case, the Content-Language would
properly only include "en".
Content-Language MAY be applied to any media type -- it is not
limited to textual documents.
18.17. Content-Length
The Content-Length message body header field contains the length of
the message body of the RTSP message (i.e., after the double CRLF
following the last header) in octets of bits. Unlike HTTP, it MUST
be included in all messages that carry a message body beyond the
header portion of the RTSP message. If it is missing, a default
value of zero is assumed. Any Content-Length greater than or equal
to zero is a valid value.
18.18. Content-Location
The Content-Location message body header field MAY be used to supply
the resource location for the message body enclosed in the message
when that body is accessible from a location separate from the
requested resource's URI. A server SHOULD provide a Content-Location
for the variant corresponding to the response message body;
especially in the case where a resource has multiple variants
associated with it, and those entities actually have separate
locations by which they might be individually accessed, the server
SHOULD provide a Content-Location for the particular variant that is
returned.
As an example, if an RTSP client performs a DESCRIBE request on a
given resource, e.g., "rtsp://a.example.com/movie/
Plan9FromOuterSpace", then the server may use additional information,
such as the User-Agent header, to determine the capabilities of the
agent. The server will then return a media description tailored to
that class of RTSP agents. To indicate which specific description
the agent receives, the resource identifier
("rtsp://a.example.com/movie/Plan9FromOuterSpace/FullHD.sdp") is
provided in Content-Location, while the description is still a valid
response for the generic resource identifier, thus enabling both
debugging and cache operation as discussed below.
The Content-Location value is not a replacement for the original
requested URI; it is only a statement of the location of the resource
corresponding to this particular variant at the time of the request.
Future requests MAY specify the Content-Location URI as the Request-
URI if the desire is to identify the source of that particular
variant. This is useful if the RTSP agent desires to verify if the
resource variant is current through a conditional request.
A cache cannot assume that a message body with a Content-Location
different from the URI used to retrieve it can be used to respond to
later requests on that Content-Location URI. However, the Content-
Location can be used to differentiate between multiple variants
retrieved from a single requested resource.
If the Content-Location is a relative URI, the relative URI is
interpreted relative to the Request-URI.
Note that Content-Location can be used in some cases to derive the
base-URI for relative URI(s) present in session description formats.
This needs to be taken into account when Content-Location is used.
The easiest way to avoid needing to consider that issue is to include
the Content-Base whenever the Content-Location is included.
Note also, when using Media Tags in conjunction with Content-
Location, it is important that the different versions have different
MTags, even if provided under different Content-Location URIs. This
is because the different content variants still have been provided in
response to the same request URI.
Note also, as in most cases, the URIs used in the DESCRIBE and the
SETUP requests are different: the URI provided in a DESCRIBE Content-
Location response can't directly be used in a SETUP request.
Instead, the steps of deriving the media resource URIs are necessary.
This commonly involves combing the media description's relative URIs,
e.g., from the SDP's a=control attribute, with the base-URI to create
the absolute URIs needed in the SETUP request.
18.19. Content-Type
The Content-Type message body header indicates the media type of the
message body sent to the recipient. Note that the content types
suitable for RTSP are likely to be restricted in practice to
presentation descriptions and parameter-value types.
18.20. CSeq
The CSeq general-header field specifies the sequence number (integer)
for an RTSP request/response pair. This field MUST be present in all
requests and responses. RTSP agents maintain a sequence number
series for each responder to which they have an open message
transport channel. For each new RTSP request an agent originates on
a particular RTSP message transport, the CSeq value MUST be
incremented by one. The initial sequence number can be any number;
however, it is RECOMMENDED to start at 0. Each sequence number
series is unique between each requester and responder, i.e., the
client has one series for its requests to a server and the server has
another when sending requests to the client. Each requester and
responder is identified by its socket address (IP address and port
number), i.e., per direction of a TCP connection. Any retransmitted
request MUST contain the same sequence number as the original, i.e.,
the sequence number is not incremented for retransmissions of the
same request. The RTSP agent receiving requests MUST process the
requests arriving on a particular transport in the order of the
sequence numbers. Responses are sent in the order that they are
generated. The RTSP response MUST have the same sequence number as
was present in the corresponding request. An RTSP agent receiving a
response MAY receive the responses out of order compared to the order
of the requests it sent. Thus, the agent MUST use the sequence
number in the response to pair it with the corresponding request.
The main purpose of the sequence number is to map responses to
requests.
The requirement to use a sequence-number increment of one for each
new request is to support any future specification of RTSP message
transport over a protocol that does not provide in-order delivery
or is unreliable.
The above rules relating to the initial sequence number may appear
unnecessarily loose. The reason for this is to cater to some
common behavior of existing implementations: when using multiple
reliable connections in sequence, it may still be easiest to use a
single sequence-number series for a client connecting with a
particular server. Thus, the initial sequence number may be
arbitrary depending on the number of previous requests. For any
unreliable transport, a stricter definition or other solution will
be required to enable detection of any loss of the first request.
When using multiple sequential transport connections, there is no
protocol mechanism to ensure in-order processing as the sequence
number is scoped on the individual transport connection and its
five tuple. Thus, there are potential issues with opening a new
transport connection to the same host for which there already
exists a transport connection with outstanding requests and
previously dispatched requests related to the same RTSP session.
RTSP Proxies also need to follow the above rules. This implies that
proxies that aggregate requests from multiple clients onto a single
transport towards a server or a next-hop proxy need to renumber these
requests to form a unified sequence on that transport, fulfilling the
above rules. A proxy capable of fulfilling some agent's request
without emitting its own request (e.g., a caching proxy that fulfills
a request from its cache) also causes a need to renumber as the
number of received requests with a particular target may not be the
same as the number of emitted requests towards that target agent. A
proxy that needs to renumber needs to perform the corresponding
renumbering back to the original sequence number for any received
response before forwarding it back to the originator of the request.
A client connected to a proxy, and using that transport to send
requests to multiple servers, creates a situation where it is
quite likely to receive the responses out of order. This is
because the proxy will establish separate transports from the
proxy to the servers on which to forward the client's requests.
When the responses arrive from the different servers, they will be
forwarded to the client in the order they arrive at the proxy and
can be processed, not the order of the client's original sequence
numbers. This is intentional to avoid some session's requests
being blocked by another server's slow processing of requests.
18.21. Date
The Date general-header field represents the date and time at which
the message was originated. The inclusion of the Date header in an
RTSP message follows these rules:
o An RTSP message, sent by either the client or the server,
containing a body MUST include a Date header, if the sending host
has a clock;
o Clients and servers are RECOMMENDED to include a Date header in
all other RTSP messages, if the sending host has a clock;
o If the server does not have a clock that can provide a reasonable
approximation of the current time, its responses MUST NOT include
a Date header field. In this case, this rule MUST be followed:
some origin-server implementations might not have a clock
available. An origin server without a clock MUST NOT assign
Expires or Last-Modified values to a response, unless these values
were associated with the resource by a system or user with a
reliable clock. It MAY assign an Expires value that is known, at
or before server-configuration time, to be in the past (this
allows "pre-expiration" of responses without storing separate
Expires values for each resource).
A received message that does not have a Date header field MUST be
assigned one by the recipient if the message will be cached by that
recipient. An RTSP implementation without a clock MUST NOT cache
responses without revalidating them on every use. An RTSP cache,
especially a shared cache, SHOULD use a mechanism, such as the
Network Time Protocol (NTP) [RFC5905], to synchronize its clock with
a reliable external standard.
The RTSP-date, a full date as specified by Section 3.3 of [RFC5322],
sent in a Date header SHOULD NOT represent a date and time subsequent
to the generation of the message. It SHOULD represent the best
available approximation of the date and time of message generation,
unless the implementation has no means of generating a reasonably
accurate date and time. In theory, the date ought to represent the
moment just before the message body is generated. In practice, the
date can be generated at any time during the message origination
without affecting its semantic value.
Note: The RTSP 2.0 date format is defined to be the full-date
format in RFC 5322. This format is more flexible than the date
format in RFC 1123 used by RTSP 1.0. Thus, implementations should
use single spaces as separators, as recommended by RFC 5322, and
support receiving the obsolete format.
Further, note that the syntax allows for a comment to be added at
the end of the date.
18.22. Expires
The Expires message body header field gives a date and time after
which the description or media-stream should be considered stale.
The interpretation depends on the method:
DESCRIBE response: The Expires header indicates a date and time
after which the presentation description (body) SHOULD be
considered stale.
SETUP response: The Expires header indicates a date and time after
which the media stream SHOULD be considered stale.
A stale cache entry should not be returned by a cache (either a proxy
cache or a user agent cache) unless it is first validated with the
origin server (or with an intermediate cache that has a fresh copy of
the message body). See Section 16 for further discussion of the
expiration model.
The presence of an Expires field does not imply that the original
resource will change or cease to exist at, before, or after that
time.
The format is an absolute date and time as defined by RTSP-date. An
example of its use is
Expires: Wed, 23 Jan 2013 15:36:52 +0000
RTSP 2.0 clients and caches MUST treat other invalid date formats,
especially those including the value "0", as having occurred in the
past (i.e., already expired).
To mark a response as "already expired," an origin server should use
an Expires date that is equal to the Date header value. To mark a
response as "never expires", an origin server SHOULD use an Expires
date approximately one year from the time the response is sent. RTSP
2.0 servers SHOULD NOT send Expires dates that are more than one year
in the future.
18.23. From
The From request-header field, if given, SHOULD contain an Internet
email address for the human user who controls the requesting user
agent. The address SHOULD be machine usable, as defined by "mailbox"
in [RFC1123].
This header field MAY be used for logging purposes and as a means for
identifying the source of invalid or unwanted requests. It SHOULD
NOT be used as an insecure form of access protection. The
interpretation of this field is that the request is being performed
on behalf of the person given, who accepts responsibility for the
method performed. In particular, robot agents SHOULD include this
header so that the person responsible for running the robot can be
contacted if problems occur on the receiving end.
The Internet email address in this field MAY be separate from the
Internet host that issued the request. For example, when a request
is passed through a proxy, the original issuer's address SHOULD be
used.
The client SHOULD NOT send the From header field without the user's
approval, as it might conflict with the user's privacy interests or
their site's security policy. It is strongly recommended that the
user be able to disable, enable, and modify the value of this field
at any time prior to a request.
18.24. If-Match
The If-Match request-header field is especially useful for ensuring
the integrity of the presentation description, independent of how the
presentation description was received. The presentation description
can be fetched via means external to RTSP (such as HTTP) or via the
DESCRIBE message. In the case of retrieving the presentation
description via RTSP, the server implementation is guaranteeing the
integrity of the description between the time of the DESCRIBE message
and the SETUP message. By including the MTag given in or with the
session description in an If-Match header part of the SETUP request,
the client ensures that resources set up are matching the
description. A SETUP request with the If-Match header for which the
MTag validation check fails MUST generate a response using 412
(Precondition Failed).
This validation check is also very useful if a session has been
redirected from one server to another.
18.25. If-Modified-Since
The If-Modified-Since request-header field is used with the DESCRIBE
and SETUP methods to make them conditional. If the requested variant
has not been modified since the time specified in this field, a
description will not be returned from the server (DESCRIBE) or a
stream will not be set up (SETUP). Instead, a 304 (Not Modified)
response MUST be returned without any message body.
An example of the field is:
If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT
18.26. If-None-Match
This request-header can be used with one or several message body tags
to make DESCRIBE requests conditional. A client that has one or more
message bodies previously obtained from the resource can verify that
none of those entities is current by including a list of their
associated message body tags in the If-None-Match header field. The
purpose of this feature is to allow efficient updates of cached
information with a minimum amount of transaction overhead. As a
special case, the value "*" matches any current entity of the
resource.
If any of the message body tags match the message body tag of the
message body that would have been returned in the response to a
similar DESCRIBE request (without the If-None-Match header) on that
resource, or if "*" is given and any current entity exists for that
resource, then the server MUST NOT perform the requested method,
unless required to do so because the resource's modification date
fails to match that supplied in an If-Modified-Since header field in
the request. Instead, if the request method was DESCRIBE, the server
SHOULD respond with a 304 (Not Modified) response, including the
cache-related header fields (particularly MTag) of one of the message
bodies that matched. For all other request methods, the server MUST
respond with a status of 412 (Precondition Failed).
See Section 16.1.3 for rules on how to determine if two message body
tags match.
If none of the message body tags match, then the server MAY perform
the requested method as if the If-None-Match header field did not
exist, but MUST also ignore any If-Modified-Since header field(s) in
the request. That is, if no message body tags match, then the server
MUST NOT return a 304 (Not Modified) response.
If the request would, without the If-None-Match header field, result
in anything other than a 2xx or 304 status, then the If-None-Match
header MUST be ignored. (See Section 16.1.4 for a discussion of
server behavior when both If-Modified-Since and If-None-Match appear
in the same request.)
The result of a request having both an If-None-Match header field and
an If-Match header field is unspecified and MUST be considered an
illegal request.
18.27. Last-Modified
The Last-Modified message body header field indicates the date and
time at which the origin server believes the presentation description
or media stream was last modified. For the DESCRIBE method, the
header field indicates the last modification date and time of the
description, for the SETUP of the media stream.
An origin server MUST NOT send a Last-Modified date that is later
than the server's time of message origination. In such cases, where
the resource's last modification would indicate some time in the
future, the server MUST replace that date with the message
origination date.
An origin server SHOULD obtain the Last-Modified value of the message
body as close as possible to the time that it generates the Date
value of its response. This allows a recipient to make an accurate
assessment of the message body's modification time, especially if the
message body changes near the time that the response is generated.
RTSP servers SHOULD send Last-Modified whenever feasible.
18.28. Location
The Location response-header field is used to redirect the recipient
to a location other than the Request-URI for completion of the
request or identification of a new resource. For 3rr responses, the
location SHOULD indicate the server's preferred URI for automatic
redirection to the resource. The field-value consists of a single
absolute URI.
Note: The Content-Location header field (Section 18.18) differs from
Location in that the Content-Location identifies the original
location of the message body enclosed in the request. Therefore, it
is possible for a response to contain header fields for both Location
and Content-Location. Also, see Section 16.2 for cache requirements
of some methods.
18.29. Media-Properties
This general-header is used in SETUP responses or PLAY_NOTIFY
requests to indicate the media's properties that currently are
applicable to the RTSP session. PLAY_NOTIFY MAY be used to modify
these properties at any point. However, the client SHOULD have
received the update prior to any action related to the new media
properties taking effect. For aggregated sessions, the Media-
Properties header will be returned in each SETUP response. The
header received in the latest response is the one that applies on the
whole session from this point until any future update. The header
MAY be included without value in GET_PARAMETER requests to the server
with a Session header included to query the current Media-Properties
for the session. The responder MUST include the current session's
media properties.
The media properties expressed by this header are the ones applicable
to all media in the RTSP session. For aggregated sessions, the
header expressed the combined media-properties. As a result,
aggregation of media MAY result in a change of the media properties
and, thus, the content of the Media-Properties header contained in
subsequent SETUP responses.
The header contains a list of property values that are applicable to
the currently setup media or aggregate of media as indicated by the
RTSP URI in the request. No ordering is enforced within the header.
Property values should be placed into a single group that handles a
particular orthogonal property. Values or groups that express
multiple properties SHOULD NOT be used. The list of properties that
can be expressed MAY be extended at any time. Unknown property
values MUST be ignored.
This specification defines the following four groups and their
property values:
Random Access:
Random-Access: Indicates that random access is possible. May
optionally include a floating-point value in seconds indicating
the longest duration between any two random access points in
the media.
Beginning-Only: Seeking is limited to the beginning only.
No-Seeking: No seeking is possible.
Content Modifications:
Immutable: The content will not be changed during the lifetime of
the RTSP session.
Dynamic: The content may be changed based on external methods or
triggers.
Time-Progressing: The media accessible progresses as wallclock
time progresses.
Retention:
Unlimited: Content will be retained for the duration of the
lifetime of the RTSP session.
Time-Limited: Content will be retained at least until the
specified wallclock time. The time must be provided in the
absolute time format specified in Section 4.4.3.
Time-Duration: Each individual media unit is retained for at
least the specified Time-Duration. This definition allows for
retaining data with a time-based sliding window. The time
duration is expressed as floating-point number in seconds. The
value 0.0 is a valid as this indicates that no data is retained
in a time-progressing session.
Supported Scale:
Scales: A quoted comma-separated list of one or more decimal
values or ranges of scale values supported by the content in
arbitrary order. A range has a start and stop value separated
by a colon. A range indicates that the content supports a
fine-grained selection of scale values. Fine-graining allows
for steps at least as small as one tenth of a scale value.
Content is considered to support fine-grained selection when
the server in response to a given scale value can produce
content with an actual scale that is less than one tenth of
scale unit, i.e., 0.1, from the requested value. Negative
values are supported. The value 0 has no meaning and MUST NOT
be used.
Examples of this header for on-demand content and a live stream
without recording are:
On-demand:
Media-Properties: Random-Access=2.5, Unlimited, Immutable,
Scales="-20, -10, -4, 0.5:1.5, 4, 8, 10, 15, 20"
Live stream without recording/timeshifting:
Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0.0
18.30. Media-Range
The Media-Range general-header is used to give the range of the media
at the time of sending the RTSP message. This header MUST be
included in the SETUP response, PLAY and PAUSE responses for media
that are time-progressing, PLAY and PAUSE responses after any change
for media that are Dynamic, and in PLAY_NOTIFY requests that are sent
due to Media-Property-Update. A Media-Range header without any range
specifications MAY be included in GET_PARAMETER requests to the
server to request the current range. In this case, the server MUST
include the current range at the time of sending the response.
The header MUST include range specifications for all time formats
supported for the media, as indicated in Accept-Ranges header
(Section 18.5) when setting up the media. The server MAY include
more than one range specification of any given time format to
indicate media that has non-continuous range. The range
specifications SHALL be ordered with the range with the lowest value
or earliest start time first, followed by ranges with increasingly
higher values or later start time.
For media that has the time-progressing property, the Media-Range
header values will only be valid for the particular point in time
when it was issued. As the wallclock progresses, so will the media
range. However, it shall be assumed that media time progresses in
direct relationship to wallclock time (with the exception of clock
skew) so that a reasonably accurate estimation of the media range can
be calculated.
18.31. MTag
The MTag response-header MAY be included in DESCRIBE, GET_PARAMETER,
or SETUP responses. The message body tags (Section 4.6) returned in
a DESCRIBE response and the one in SETUP refer to the presentation,
i.e., both the returned session description and the media stream.
This allows for verification that one has the right session
description to a media resource at the time of the SETUP request.
However, it has the disadvantage that a change in any of the parts
results in invalidation of all the parts.
If the MTag is provided both inside the message body, e.g., within
the "a=mtag" attribute in SDP, and in the response message, then both
tags MUST be identical. It is RECOMMENDED that the MTag be primarily
given in the RTSP response message, to ensure that caches can use the
MTag without requiring content inspection. However, for session
descriptions that are distributed outside of RTSP, for example, using
HTTP, etc., it will be necessary to include the message body tag in
the session description as specified in Appendix D.1.9.
SETUP and DESCRIBE requests can be made conditional upon the MTag
using the headers If-Match (Section 18.24) and If-None-Match
(Section 18.26).
18.32. Notify-Reason
The Notify-Reason response-header is solely used in the PLAY_NOTIFY
method. It indicates the reason why the server has sent the
asynchronous PLAY_NOTIFY request (see Section 13.5).
18.33. Pipelined-Requests
The Pipelined-Requests general-header is used to indicate that a
request is to be executed in the context created by a previous
request(s). The primary usage of this header is to allow pipelining
of SETUP requests so that any additional SETUP request after the
first one does not need to wait for the session ID to be sent back to
the requesting agent. The header contains a unique identifier that
is scoped by the persistent connection used to send the requests.
Upon receiving a request with the Pipelined-Requests, the responding
agent MUST look up if there exists a binding between this Pipelined-
Requests identifier for the current persistent connection and an RTSP
session ID. If the binding exists, then the received request is
processed the same way as if it contained the Session header with the
found session ID. If there does not exist a mapping and no Session
header is included in the request, the responding agent MUST create a
binding upon the successful completion of a session creating request,
i.e., SETUP. A binding MUST NOT be created, if the request failed to
create an RTSP session. In case the request contains both a Session
header and the Pipelined-Requests header, the Pipelined-Requests
header MUST be ignored.
Note: Based on the above definition, at least the first request
containing a new unique Pipelined-Requests header will be required to
be a SETUP request (unless the protocol is extended with new methods
of creating a session). After that first one, additional SETUP
requests or requests of any type using the RTSP session context may
include the Pipelined-Requests header.
When responding to any request that contained the Pipelined-Requests
header, the server MUST also include the Session header when a
binding to a session context exists. An RTSP agent that knows the
session identifier SHOULD NOT use the Pipelined-Requests header in
any request and only use the Session header. This as the Session
identifier is persistent across transport contexts, like TCP
connections, which the Pipelined-Requests identifier is not.
The RTSP agent sending the request with a Pipelined-Requests header
has the responsibility for using a unique and previously unused
identifier within the transport context. Currently, only a TCP
connection is defined as such a transport context. A server MUST
delete the Pipelined-Requests identifier and its binding to a session
upon the termination of that session. Despite the previous mandate,
RTSP agents are RECOMMENDED not to reuse identifiers to allow for
better error handling and logging.
RTSP Proxies may need to translate Pipelined-Requests identifier
values from incoming requests to outgoing to allow for aggregation of
requests onto a persistent connection.
18.34. Proxy-Authenticate
The Proxy-Authenticate response-header field MUST be included as part
of a 407 (Proxy Authentication Required) response. The field-value
consists of a challenge that indicates the authentication scheme and
parameters applicable to the proxy for this Request-URI. The
definition of the header is in [RFC7235], and any applicable HTTP
authentication schemes appear in other RFCs, such as Digest [RFC7616]
and Basic [RFC7617].
The HTTP access authentication process is described in [RFC7235].
This header MUST only be used in response messages related to client-
to-server requests.
18.35. Proxy-Authentication-Info
The Proxy-Authentication-Info response-header is used by the proxy to
communicate some information regarding the successful authentication
to the proxy in the message response in some authentication schemes,
such as the Digest scheme [RFC7616]. The definition of the header is
in [RFC7615], and any applicable HTTP authentication schemes appear
in other RFCs. This header MUST only be used in response messages
related to client-to-server requests. This header has hop-by-hop
scope.
18.36. Proxy-Authorization
The Proxy-Authorization request-header field allows the client to
identify itself (or its user) to a proxy that requires
authentication. The Proxy-Authorization field-value consists of
credentials containing the authentication information of the user
agent for the proxy or realm of the resource being requested. The
definition of the header is in [RFC7235], and any applicable HTTP
authentication schemes appear in other RFCs, such as Digest [RFC7616]
and Basic [RFC7617].
The HTTP access authentication process is described in [RFC7235].
Unlike Authorization, the Proxy-Authorization header field applies
only to the next-hop proxy. This header MUST only be used in client-
to-server requests.
18.37. Proxy-Require
The Proxy-Require request-header field is used to indicate proxy-
sensitive features that MUST be supported by the proxy. Any Proxy-
Require header features that are not supported by the proxy MUST be
negatively acknowledged by the proxy to the client using the
Unsupported header. The proxy MUST use the 551 (Option Not
Supported) status code in the response. Any feature tag included in
the Proxy-Require does not apply to the endpoint (server or client).
To ensure that a feature is supported by both proxies and servers,
the tag needs to be included in also a Require header.
See Section 18.43 for more details on the mechanics of this message
and a usage example. See discussion in the proxies section
(Section 15.1) about when to consider that a feature requires proxy
support.
Example of use:
Proxy-Require: play.basic
18.38. Proxy-Supported
The Proxy-Supported general-header field enumerates all the
extensions supported by the proxy using feature tags. The header
carries the intersection of extensions supported by the forwarding
proxies. The Proxy-Supported header MAY be included in any request
by a proxy. It MUST be added by any proxy if the Supported header is
present in a request. When present in a request, the receiver MUST
copy the received Proxy-Supported header in the response.
The Proxy-Supported header field contains a list of feature tags
applicable to proxies, as described in Section 4.5. The list is the
intersection of all feature tags understood by the proxies. To
achieve an intersection, the proxy adding the Proxy-Supported header
includes all proxy feature tags it understands. Any proxy receiving
a request with the header MUST check the list and remove any feature
tag(s) it does not support. A Proxy-Supported header present in the
response MUST NOT be modified by the proxies. These feature tags are
the ones the proxy chains support in general and are not specific to
the request resource.
Example:
C->P1: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
User-Agent: PhonyClient/1.2
P1->P2: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
Proxy-Supported: proxy-foo, proxy-bar, proxy-blech
Via: 2.0 pro.example.com
P2->S: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
Proxy-Supported: proxy-foo, proxy-blech
Via: 2.0 pro.example.com, 2.0 prox2.example.com
S->C: RTSP/2.0 200 OK
Supported: foo, bar, baz
Proxy-Supported: proxy-foo, proxy-blech
Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
Via: 2.0 pro.example.com, 2.0 prox2.example.com
18.39. Public
The Public response-header field lists the set of methods supported
by the response sender. This header applies to the general
capabilities of the sender, and its only purpose is to indicate the
sender's capabilities to the recipient. The methods listed may or
may not be applicable to the Request-URI; the Allow header field
(Section 18.6) MAY be used to indicate methods allowed for a
particular URI.
Example of use:
Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
In the event that there are proxies between the sender and the
recipient of a response, each intervening proxy MUST modify the
Public header field to remove any methods that are not supported via
that proxy. The resulting Public header field will contain an
intersection of the sender's methods and the methods allowed through
by the intervening proxies.
In general, proxies should allow all methods to transparently pass
through from the sending RTSP agent to the receiving RTSP agent,
but there may be cases where this is not desirable for a given
proxy. Modification of the Public response-header field by the
intervening proxies ensures that the request sender gets an
accurate response indicating the methods that can be used on the
target agent via the proxy chain.
18.40. Range
The Range general-header specifies a time range in PLAY
(Section 13.4), PAUSE (Section 13.6), SETUP (Section 13.3), and
PLAY_NOTIFY (Section 13.5) requests and responses. It MAY be
included in GET_PARAMETER requests from the client to the server with
only a Range format and no value to request the current media
position, whether the session is in Play or Ready state in the
included format. The server SHALL, if supporting the range format,
respond with the current playing point or pause point as the start of
the range. If an explicit stop point was used in the previous PLAY
request, then that value shall be included as stop point. Note that
if the server is currently under any type of media playback
manipulation affecting the interpretation of the Range header, like
scale value other than 1, that fact is also required to be included
in any GET_PARAMETER response by including the Scale header to
provide complete information.
The range can be specified in a number of units. This specification
defines smpte (Section 4.4.1), npt (Section 4.4.2), and clock
(Section 4.4.3) range units. While octet ranges (Byte Ranges) (see
Section 2.1 of [RFC7233]) and other extended units MAY be used, their
behavior is unspecified since they are not normally meaningful in
RTSP. Servers supporting the Range header MUST understand the NPT
range format and SHOULD understand the SMPTE range format. If the
Range header is sent in a time format that is not understood, the
recipient SHOULD return 456 (Header Field Not Valid for Resource) and
include an Accept-Ranges header indicating the supported time formats
for the given resource.
Example:
Range: clock=19960213T143205Z-
The Range header contains a range of one single range format. A
range is a half-open interval with a start and an end point,
including the start point but excluding the end point. A range may
either be fully specified with explicit values for start point and
end point or have either the start or end point be implicit. An
implicit start point indicates the session's pause point, and if no
pause point is set, the start of the content. An implicit end point
indicates the end of the content. The usage of both implicit start
and end points is not allowed in the same Range header; however, the
omission of the Range header has that meaning, i.e., from pause point
(or start) until end of content.
As noted, Range headers define half-open intervals. A range of
A-B starts exactly at time A, but ends just before B. Only the
start time of a media unit such as a video or audio frame is
relevant. For example, assume that video frames are generated
every 40 ms. A range of 10.0-10.1 would include a video frame
starting at 10.0 or later time and would include a video frame
starting at 10.08, even though it lasted beyond the interval. A
range of 10.0-10.08, on the other hand, would exclude the frame at
10.08.
Please note the difference between NPT timescales' "now" and an
implicit start value. Implicit values reference the current
pause-point, while "now" is the current time. In a time-
progressing session with recording (retention for some or full
time), the pause point may be 2 min into the session while now
could be 1 hour into the session.
By default, range intervals increase, where the second point is
larger than the first point.
Example:
Range: npt=10-15
However, range intervals can also decrease if the Scale header (see
Section 18.46) indicates a negative scale value. For example, this
would be the case when a playback in reverse is desired.
Example:
Scale: -1
Range: npt=15-10
Decreasing ranges are still half-open intervals as described above.
Thus, for range A-B, A is closed and B is open. In the above
example, 15 is closed and 10 is open. An exception to this rule is
the case when B=0 is in a decreasing range. In this case, the range
is closed on both ends, as otherwise there would be no way to reach 0
on a reverse playback for formats that have such a notion, like NPT
and SMPTE.
Example:
Scale: -1
Range: npt=15-0
In this range, both 15 and 0 are closed.
A decreasing range interval without a corresponding negative value in
the Scale header is not valid.
18.41. Referrer
The Referrer request-header field allows the client to specify, for
the server's benefit, the address (URI) of the resource from which
the Request-URI was obtained. The URI refers to that of the
presentation description, typically retrieved via HTTP. The Referrer
request-header allows a server to generate lists of back-links to
resources for interest, logging, optimized caching, etc. It also
allows obsolete or mistyped links to be traced for maintenance. The
Referrer field MUST NOT be sent if the Request-URI was obtained from
a source that does not have its own URI, such as input from the user
keyboard.
If the field-value is a relative URI, it SHOULD be interpreted
relative to the Request-URI. The URI MUST NOT include a fragment
identifier.
Because the source of a link might be private information or might
reveal an otherwise private information source, it is strongly
recommended that the user be able to select whether or not the
Referrer field is sent. For example, a streaming client could have a
toggle switch for openly/anonymously, which would respectively
enable/disable the sending of Referrer and From information.
Clients SHOULD NOT include a Referrer header field in an (non-secure)
RTSP request if the referring page was transferred with a secure
protocol.
18.42. Request-Status
This request-header is used to indicate the end result for requests
that take time to complete, such as PLAY (Section 13.4). It is sent
in PLAY_NOTIFY (Section 13.5) with the end-of-stream reason to report
how the PLAY request concluded, either in success or in failure. The
header carries a reference to the request it reports on using the
CSeq number and the Session ID used in the request reported on. This
is not ensured to be unambiguous due to the fact that the CSeq number
is scoped by the transport connection. Agents originating requests
can reduce the issue by using a monotonically increasing counter
across all sequential transports used. The header provides both a
numerical status code (according to Section 8.1.1) and a human-
readable reason phrase.
Example:
Request-Status: cseq=63 status=500 reason="Media data unavailable"
Proxies that renumber the CSeq header need to perform corresponding
remapping of the cseq parameter in this header when forwarding the
request to the next-hop agent.
18.43. Require
The Require request-header field is used by agents to ensure that the
other endpoint supports features that are required in respect to this
request. It can also be used to query if the other endpoint supports
certain features; however, the use of the Supported general-header
(Section 18.51) is much more effective in this purpose. In case any
of the feature tags listed by the Require header are not supported by
the server or client receiving the request, it MUST respond to the
request using the error code 551 (Option Not Supported) and include
the Unsupported header listing those feature tags that are NOT
supported. This header does not apply to proxies; for the same
functionality with respect to proxies, see the Proxy-Require header
(Section 18.37) with the exception of media-modifying proxies.
Media-modifying proxies, due to their nature of handling media in a
way that is very similar to a server, do need to understand also the
server's features to correctly serve the client.
This is to make sure that the client-server interaction will
proceed without delay when all features are understood by both
sides and only slow down if features are not understood (as in the
example below). For a well-matched client-server pair, the
interaction proceeds quickly, saving a round trip often required
by negotiation mechanisms. In addition, it also removes state
ambiguity when the client requires features that the server does
not understand.
Example (Not complete):
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/2.0
CSeq: 302
Require: funky-feature
Funky-Parameter: funkystuff
S->C: RTSP/2.0 551 Option not supported
CSeq: 302
Unsupported: funky-feature
In this example, "funky-feature" is the feature tag that indicates to
the client that the fictional Funky-Parameter field is required. The
relationship between "funky-feature" and Funky-Parameter is not
communicated via the RTSP exchange, since that relationship is an
immutable property of "funky-feature" and thus should not be
transmitted with every exchange.
Proxies and other intermediary devices MUST ignore this header. If a
particular extension requires that intermediate devices support it,
the extension should be tagged in the Proxy-Require field instead
(see Section 18.37). See discussion in the proxies section
(Section 15.1) about when to consider that a feature requires proxy
support.
18.44. Retry-After
The Retry-After response-header field can be used with a 503 (Service
Unavailable) or 553 (Proxy Unavailable) response to indicate how long
the service is expected to be unavailable to the requesting client.
This field MAY also be used with any 3rr (Redirection) response to
indicate the minimum time the user agent is asked to wait before
issuing the redirected request. A response using 413 (Request
Message Body Too Large) when the restriction is temporary MAY also
include the Retry-After header. The value of this field can be
either an RTSP-date or an integer number of seconds (in decimal)
after the time of the response.
Example:
Retry-After: Fri, 31 Dec 1999 23:59:59 GMT
Retry-After: 120
In the latter example, the delay is 2 minutes.
18.45. RTP-Info
The RTP-Info general-header field is used to set RTP-specific
parameters in the PLAY and GET_PARAMETER responses or PLAY_NOTIFY and
GET_PARAMETER requests. For streams using RTP as transport protocol,
the RTP-Info header SHOULD be part of a 200 response to PLAY.
The exclusion of the RTP-Info in a PLAY response for RTP-
transported media will result in a client needing to synchronize
the media streams using RTCP. This may have negative impact as
the RTCP can be lost and does not need to be particularly timely
in its arrival. Also, functionality that informs the client from
which packet a seek has occurred is affected.
The RTP-Info MAY be included in SETUP responses to provide
synchronization information when changing transport parameters, see
Section 13.3. The RTP-Info header and the Range header MAY be
included in a GET_PARAMETER request from client to server without any
values to request the current playback point and corresponding RTP
synchronization information. When the RTP-Info header is included in
a Request, the Range header MUST also be included. The server
response SHALL include both the Range header and the RTP-Info header.
If the session is in Play state, then the value of the Range header
SHALL be filled in with the current playback point and with the
corresponding RTP-Info values. If the server is in another state, no
values are included in the RTP-Info header. The header is included
in PLAY_NOTIFY requests with the Notify-Reason of the end of stream
to provide RTP information about the end of the stream.
The header can carry the following parameters:
url: Indicates the stream URI for which the following RTP parameters
correspond; this URI MUST be the same as used in the SETUP
request for this media stream. Any relative URI MUST use the
Request-URI as base URI. This parameter MUST be present.
ssrc: The SSRC to which the RTP timestamp and sequence number
provided applies. This parameter MUST be present.
seq: Indicates the sequence number of the first packet of the stream
that is direct result of the request. This allows clients to
gracefully deal with packets when seeking. The client uses
this value to differentiate packets that originated before the
seek from packets that originated after the seek. Note that a
client may not receive the packet with the expressed sequence
number and instead may receive packets with a higher sequence
number due to packet loss or reordering. This parameter is
RECOMMENDED to be present.
rtptime: MUST indicate the RTP timestamp value corresponding to the
start time value in the Range response-header or, if not
explicitly given, the implied start point. The client uses
this value to calculate the mapping of RTP time to NPT or other
media timescale. This parameter SHOULD be present to ensure
inter-media synchronization is achieved. There exists no
requirement that any received RTP packet will have the same RTP
timestamp value as the one in the parameter used to establish
synchronization.
A mapping from RTP timestamps to NTP format timestamps (wallclock)
is available via RTCP. However, this information is not
sufficient to generate a mapping from RTP timestamps to media
clock time (NPT, etc.). Furthermore, in order to ensure that this
information is available at the necessary time (immediately at
startup or after a seek), and that it is delivered reliably, this
mapping is placed in the RTSP control channel.
In order to compensate for drift for long, uninterrupted
presentations, RTSP clients should additionally map NPT to NTP,
using initial RTCP sender reports to do the mapping, and later
reports to check drift against the mapping.
Example:
Range:npt=3.25-15
RTP-Info:url="rtsp://example.com/foo/audio" ssrc=0A13C760:seq=45102;
rtptime=12345678,url="rtsp://example.com/foo/video"
ssrc=9A9DE123:seq=30211;rtptime=29567112
Lets assume that Audio uses a 16 kHz RTP timestamp clock and Video
a 90 kHz RTP timestamp clock. Then, the media synchronization is
depicted in the following way.
NPT 3.0---3.1---3.2-X-3.3---3.4---3.5---3.6
Audio PA A
Video V PV
X: NPT time value = 3.25, from Range header.
A: RTP timestamp value for Audio from RTP-Info header (12345678).
V: RTP timestamp value for Video from RTP-Info header (29567112).
PA: RTP audio packet carrying an RTP timestamp of 12344878, which
corresponds to NPT = (12344878 - A) / 16000 + 3.25 = 3.2
PV: RTP video packet carrying an RTP timestamp of 29573412, which
corresponds to NPT = (29573412 - V) / 90000 + 3.25 = 3.32
18.46. Scale
The Scale general-header indicates the requested or used view rate
for the media resource being played back. A scale value of 1
indicates normal play at the normal forward viewing rate. If not 1,
the value corresponds to the rate with respect to normal viewing
rate. For example, a value of 2 indicates twice the normal viewing
rate ("fast forward") and a value of 0.5 indicates half the normal
viewing rate. In other words, a value of 2 has content time increase
at twice the playback time. For every second of elapsed (wallclock)
time, 2 seconds of content time will be delivered. A negative value
indicates reverse direction. For certain media transports, this may
require certain considerations to work consistently; see Appendix C.1
for description on how RTP handles this.
The transmitted-data rate SHOULD NOT be changed by selection of a
different scale value. The resulting bitrate should be reasonably
close to the nominal bitrate of the content for scale = 1. The
server has to actively manipulate the data when needed to meet the
bitrate constraints. Implementation of scale changes depends on the
server and media type. For video, a server may, for example, deliver
only key frames or selected frames. For audio, it may time-scale the
audio while preserving pitch or, less desirably, deliver fragments of
audio, or completely mute the audio.
The server and content may restrict the range of scale values that it
supports. The supported values are indicated by the Media-Properties
header (Section 18.29). The client SHOULD only indicate request
values to be supported. However, as the values may change as the
content progresses, a requested value may no longer be valid when the
request arrives. Thus, a non-supported value in a request does not
generate an error, it only forces the server to choose the closest
value. The response MUST always contain the actual scale value
chosen by the server.
If the server does not implement the possibility to scale, it will
not return a Scale header. A server supporting scale operations for
PLAY MUST indicate this with the use of the "play.scale" feature tag.
When indicating a negative scale for a reverse playback, the Range
header MUST indicate a decreasing range as described in
Section 18.40.
Example of playing in reverse at 3.5 times normal rate:
Scale: -3.5
Range: npt=15-10
18.47. Seek-Style
When a client sends a PLAY request with a Range header to perform a
random access to the media, the client does not know if the server
will pick the first media samples or the first random access point
prior to the request range. Depending on the use case, the client
may have a strong preference. To express this preference and provide
the client with information on how the server actually acted on that
preference, the Seek-Style general-header is defined.
Seek-Style is a general-header that MAY be included in any PLAY
request to indicate the client's preference for any media stream that
has the random access properties. The server MUST always include the
header in any PLAY response for media with random access properties
to indicate what policy was applied. A server that receives an
unknown Seek-Style policy MUST ignore it and select the server
default policy. A client receiving an unknown policy MUST ignore it
and use the Range header and any media synchronization information as
basis to determine what the server did.
This specification defines the following seek policies that may be
requested (see also Section 4.7.1):
RAP: Random Access Point (RAP) is the behavior of requesting the
server to locate the closest previous random access point that
exists in the media aggregate and deliver from that. By
requesting a RAP, media quality will be the best possible as all
media will be delivered from a point where full media state can be
established in the media decoder.
CoRAP: Conditional Random Access Point (CoRAP) is a variant of the
above RAP behavior. This policy is primarily intended for cases
where there is larger distance between the random access points in
the media. CoRAP uses the RAP policy if the condition that there
is a Random Access Point closer to the requested start point than
to the current pause point is fulfilled. Otherwise, no seeking is
performed and playback will continue from the current pause point.
This policy assumes that the media state existing prior to the
pause is usable if delivery is continued. If the client or server
knows that this is not the fact, the RAP policy should be used.
In other words, in most cases when the client requests a start
point prior to the current pause point, a valid decoding
dependency chain from the media delivered prior to the pause and
to the requested media unit will not exist. If the server
searched to a random access point, the server MUST return the
CoRAP policy in the Seek-Style header and adjust the Range header
to reflect the position of the selected RAP. In case the random
access point is farther away and the server chooses to continue
from the current pause point, it MUST include the "Next" policy in
the Seek-Style header and adjust the Range header start point to
the current pause point.
First-Prior: The first-prior policy will start delivery with the
media unit that has a playout time first prior to the requested
time. For discrete media, that would only include media units
that would still be rendered at the request time. For continuous
media, that is media that will be rendered during the requested
start time of the range.
Next: The next media units after the provided start time of the
range: for continuous framed media, that would mean the first next
frame after the provided time and for discrete media, the first
unit that is to be rendered after the provided time. The main
usage for this case is when the client knows it has all media up
to a certain point and would like to continue delivery so that a
complete uninterrupted media playback can be achieved. An example
of such a scenario would be switching from a broadcast/multicast
delivery to a unicast-based delivery. This policy MUST only be
used on the client's explicit request.
Please note that these expressed preferences exist for optimizing the
startup time or the media quality. The "Next" policy breaks the
normal definition of the Range header to enable a client to request
media with minimal overlap, although some may still occur for
aggregated sessions. RAP and First-Prior both fulfill the
requirement of providing media from the requested range and forward.
However, unless RAP is used, the media quality for many media codecs
using predictive methods can be severely degraded unless additional
data is available as, for example, already buffered, or through other
side channels.
18.48. Server
The Server general-header field contains information about the
software used by the origin server to create or handle the request.
This field can contain multiple product tokens and comments
identifying the server and any significant subproducts. The product
tokens are listed in order of their significance for identifying the
application.
Example:
Server: PhonyServer/1.0
If the response is being forwarded through a proxy, the proxy
application MUST NOT modify the Server response-header. Instead, it
SHOULD include a Via field (Section 18.57). If the response is
generated by the proxy, the proxy application MUST return the Server
response-header as previously returned by the server.
18.49. Session
The Session general-header field identifies an RTSP session. An RTSP
session is created by the server as a result of a successful SETUP
request, and in the response, the session identifier is given to the
client. The RTSP session exists until destroyed by a TEARDOWN or a
REDIRECT or is timed out by the server.
The session identifier is chosen by the server (see Section 4.3) and
MUST be returned in the SETUP response. Once a client receives a
session identifier, it MUST be included in any request related to
that session. This means that the Session header MUST be included in
a request, using the following methods: PLAY, PAUSE, PLAY_NOTIFY and
TEARDOWN. It MAY be included in SETUP, OPTIONS, SET_PARAMETER,
GET_PARAMETER, and REDIRECT. It MUST NOT be included in DESCRIBE.
The Session header MUST NOT be included in the following methods, if
these requests are pipelined and if the session identifier is not yet
known: PLAY, PAUSE, TEARDOWN, SETUP, OPTIONS SET_PARAMETER, and
GET_PARAMETER.
In an RTSP response, the session header MUST be included in methods,
SETUP, PLAY, PAUSE, and PLAY_NOTIFY, and it MAY be included in
methods TEARDOWN and REDIRECT. If included in the request of the
following methods it MUST also be included in the response: OPTIONS,
GET_PARAMETER, and SET_PARAMETER. It MUST NOT be included in
DESCRIBE responses.
Note that a session identifier identifies an RTSP session across
transport sessions or connections. RTSP requests for a given session
can use different URIs (Presentation and media URIs). Note, that
there are restrictions depending on the session as to which URIs are
acceptable for a given method. However, multiple "user" sessions for
the same URI from the same client will require use of different
session identifiers.
The session identifier is needed to distinguish several delivery
requests for the same URI coming from the same client.
The response 454 (Session Not Found) MUST be returned if the session
identifier is invalid.
The header MAY include a parameter for session timeout period. If
not explicitly provided, this value is set to 60 seconds. As this
affects how often session keep-alives are needed, values smaller than
30 seconds are not recommended. However, larger-than-default values
can be useful in applications of RTSP that have inactive but
established sessions for longer time periods.
The 60-second value was chosen as the session timeout value as it
results in keep-alive messages that are not too frequent and low
sensitivity to variations in request/response timing. If one
reduces the timeout value to below 30 seconds, the corresponding
request/response timeout becomes a significant part of the session
timeout. The 60-second value also allows for reasonably rapid
recovery of committed server resources in case of client failure.
18.50. Speed
The Speed general-header field requests the server to deliver
specific amounts of nominal media time per unit of delivery time,
contingent on the server's ability and desire to serve the media
stream at the given speed. The client requests the delivery speed to
be within a given range with a lower and upper bound. The server
SHALL deliver at the highest possible speed within the range, but not
faster than the upper bound, for which the underlying network path
can support the resulting transport data rates. As long as any speed
value within the given range can be provided, the server SHALL NOT
modify the media quality. Only if the server is unable to deliver
media at the speed value provided by the lower bound shall it reduce
the media quality.
Implementation of the Speed functionality by the server is OPTIONAL.
The server can indicate its support through a feature tag,
play.speed. The lack of a Speed header in the response is an
indication of lack of support of this functionality.
The speed parameter values are expressed as a positive decimal value,
e.g., a value of 2.0 indicates that data is to be delivered twice as
fast as normal. A speed value of zero is invalid. The range is
specified in the form "lower bound - upper bound". The lower-bound
value may be smaller or equal to the upper bound. All speeds may not
be possible to support. Therefore, the server MAY modify the
requested values to the closest supported. The actual supported
speed MUST be included in the response. However, note that the use
cases may vary and that Speed value ranges such as 0.7-0.8, 0.3-2.0,
1.0-2.5, and 2.5-2.5 all have their usages.
Example:
Speed: 1.0-2.5
Use of this header changes the bandwidth used for data delivery. It
is meant for use in specific circumstances where delivery of the
presentation at a higher or lower rate is desired. The main use
cases are buffer operations or local scale operations. Implementers
should keep in mind that bandwidth for the session may be negotiated
beforehand (by means other than RTSP) and, therefore, renegotiation
may be necessary. To perform Speed operations, the server needs to
ensure that the network path can support the resulting bitrate.
Thus, the media transport needs to support feedback so that the
server can react and adapt to the available bitrate.
18.51. Supported
The Supported general-header enumerates all the extensions supported
by the client or server using feature tags. The header carries the
extensions supported by the message-sending client or server. The
Supported header MAY be included in any request. When present in a
request, the receiver MUST respond with its corresponding Supported
header. Note that the Supported header is also included in 4xx and
5xx responses.
The Supported header contains a list of feature tags, described in
Section 4.5, that are understood by the client or server. These
feature tags are the ones the server or client supports in general
and are not specific to the request resource.
Example:
C->S: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
Supported: bar, blech, baz
18.52. Terminate-Reason
The Terminate-Reason request-header allows the server, when sending a
REDIRECT or TEARDOWN request, to provide a reason for the session
termination and any additional information. This specification
identifies three reasons for Redirections and may be extended in the
future:
Server-Admin: The server needs to be shut down for some
administrative reason.
Session-Timeout: A client's session has been kept alive for extended
periods of time and the server has determined that it needs to
reclaim the resources associated with this session.
Internal-Error An internal error that is impossible to recover from
has occurred, forcing the server to terminate the session.
The Server may provide additional parameters containing information
around the redirect. This specification defines the following ones.
time: Provides a wallclock time when the server will stop providing
any service.
user-msg: A UTF-8 text string with a message from the server to the
user. This message SHOULD be displayed to the user.
18.53. Timestamp
The Timestamp general-header describes when the agent sent the
request. The value of the timestamp is of significance only to the
agent and may use any timescale. The responding agent MUST echo the
exact same value and MAY, if it has accurate information about this,
add a floating-point number indicating the number of seconds that has
elapsed since it has received the request. The timestamp can be used
by the agent to compute the round-trip time to the responding agent
so that it can adjust the timeout value for retransmissions when
running over an unreliable protocol. It also resolves retransmission
ambiguities for unreliable transport of RTSP.
Note that the present specification provides only for reliable
transport of RTSP messages. The Timestamp general-header is
specified in case the protocol is extended in the future to use
unreliable transport.
18.54. Transport
The Transport general-header indicates which transport protocol is to
be used and configures its parameters such as destination address,
compression, multicast time-to-live and destination port for a single
stream. It sets those values not already determined by a
presentation description.
A Transport request-header MAY contain a list of transport options
acceptable to the client, in the form of multiple transport
specification entries. Transport specifications are comma separated
and listed in decreasing order of preference. Each transport
specification consists of a transport protocol identifier, followed
by any number of parameters separated by semicolons. A Transport
request-header MAY contain multiple transport specifications using
the same transport protocol identifier. The server MUST return a
Transport response-header in the response to indicate the values
actually chosen, if any. If no transport specification is supported,
no transport header is returned and the response MUST use the status
code 461 (Unsupported Transport) (Section 17.4.25). In case more
than one transport specification was present in the request, the
server MUST return the single transport specification (transport-
spec) that was actually chosen, if any. The number of transport-spec
entries is expected to be limited as the client will receive guidance
on what configurations are possible from the presentation
description.
The Transport header MAY also be used in subsequent SETUP requests to
change transport parameters. A server MAY refuse to change
parameters of an existing stream.
The transport protocol identifier defines, for each transport
specification, which transport protocol to use and any related rules.
Each transport protocol identifier defines the parameters that are
required to occur; additional optional parameters MAY occur. This
flexibility is provided as parameters may be different and provide
different options to the RTSP agent. A transport specification may
only contain one of any given parameter within it. A parameter
consists of a name and optionally a value string. Parameters MAY be
given in any order. Additionally, a transport specification may only
contain either the unicast or the multicast transport type parameter.
The transport protocol identifier, and all parameters, need to be
understood in a transport specification; if not, the transport
specification MUST be ignored. An RTSP proxy of any type that uses
or modifies the transport specification, e.g., access proxy or
security proxy, MUST remove specifications with unknown parameters
before forwarding the RTSP message. If that results in no remaining
transport specification, the proxy SHALL send a 461 (Unsupported
Transport) (Section 17.4.25) response without any Transport header.
The Transport header is restricted to describing a single media
stream. (RTSP can also control multiple streams as a single
entity.) Making it part of RTSP rather than relying on a
multitude of session description formats greatly simplifies
designs of firewalls.
The general syntax for the transport protocol identifier is a list of
slash-separated tokens:
Value1/Value2/Value3...
Which, for RTP transports, takes the form:
RTP/profile/lower-transport.
The default value for the "lower-transport" parameters is specific to
the profile. For RTP/AVP, the default is UDP.
There are two different methods for how to specify where the media
should be delivered for unicast transport:
dest_addr: The presence of this parameter and its values indicates
the destination address or addresses (host address and port
pairs for IP flows) necessary for the media transport.
No dest_addr: The lack of the dest_addr parameter indicates that the
server MUST send media to the same address from which the RTSP
messages originates.
The choice of method for indicating where the media is to be
delivered depends on the use case. In some cases, the only allowed
method will be to use no explicit address indication and have the
server deliver media to the source of the RTSP messages.
For multicast, there are several methods for specifying addresses,
but they are different in how they work compared with unicast:
dest_addr with client picked address: The address and relevant
parameters, like TTL (scope), for the actual multicast group to
deliver the media to. There are security implications
(Section 21) with this method that need to be addressed because
an RTSP server can be used as a DoS attacker on an existing
multicast group.
dest_addr using Session Description Information: The information
included in the transport header can all be coming from the
session description, e.g., the SDP "c=" and "m=" lines. This
mitigates some of the security issues of the previous methods
as it is the session provider that picks the multicast group
and scope. The client MUST include the information if it is
available in the session description.
No dest_addr: The behavior when no explicit multicast group is
present in a request is not defined.
An RTSP proxy will need to take care. If the media is not desired to
be routed through the proxy, the proxy will need to introduce the
destination indication.
Below are the configuration parameters associated with transport:
General parameters:
unicast / multicast: This parameter is a mutually exclusive
indication of whether unicast or multicast delivery will be
attempted. One of the two values MUST be specified. Clients
that are capable of handling both unicast and multicast
transmission need to indicate such capability by including two
full transport-specs with separate parameters for each.
layers: The number of multicast layers to be used for this media
stream. The layers are sent to consecutive addresses starting
at the dest_addr address. If the parameter is not included, it
defaults to a single layer.
dest_addr: A general destination address parameter that can contain
one or more address specifications. Each combination of
protocol/profile/lower transport needs to have the format and
interpretation of its address specification defined. For
RTP/AVP/UDP and RTP/AVP/TCP, the address specification is a
tuple containing a host address and port. Note, only a single
destination parameter per transport spec is intended. The
usage of multiple destinations to distribute a single media to
multiple entities is unspecified.
The client originating the RTSP request MAY specify the
destination address of the stream recipient with the host
address as part of the tuple. When the destination address is
specified, the recipient may be a different party than the
originator of the request. To avoid becoming the unwitting
perpetrator of a remote-controlled DoS attack, a server MUST
perform security checks (see Section 21.2.1) and SHOULD log
such attempts before allowing the client to direct a media
stream to a recipient address not chosen by the server.
Implementations cannot rely on TCP as a reliable means of
client identification. If the server does not allow the host
address part of the tuple to be set, it MUST return 463
(Destination Prohibited).
The host address part of the tuple MAY be empty, for example
":58044", in cases when it is desired to specify only the
destination port. Responses to requests including the
Transport header with a dest_addr parameter SHOULD include the
full destination address that is actually used by the server.
The server MUST NOT remove address information that is already
present in the request when responding, unless the protocol
requires it.
src_addr: A general source address parameter that can contain one or
more address specifications. Each combination of
protocol/profile/lower transport needs to have the format and
interpretation of its address specification defined. For
RTP/AVP/UDP and RTP/AVP/TCP, the address specification is a
tuple containing a host address and port.
This parameter MUST be specified by the server if it transmits
media packets from an address other than the one RTSP messages
are sent to. This will allow the client to verify the source
address and give it a destination address for its RTCP feedback
packets, if RTP is used. The address or addresses indicated in
the src_addr parameter SHOULD be used both for the sending and
receiving of the media stream's data packets. The main reasons
are threefold: First, indicating the port and source address(s)
lets the receiver know where from the packets is expected to
originate. Second, traversal of NATs is greatly simplified
when traffic is flowing symmetrically over a NAT binding.
Third, certain NAT traversal mechanisms need to know to which
address and port to send so-called "binding packets" from the
receiver to the sender, thus creating an address binding in the
NAT that the sender-to-receiver packet flow can use.
This information may also be available through SDP.
However, since this is more a feature of transport than
media initialization, the authoritative source for this
information should be in the SETUP response.
mode: The mode parameter indicates the methods to be supported for
this session. The currently defined valid value is "PLAY". If
not provided, the default is "PLAY". The "RECORD" value was
defined in RFC 2326; in this specification, it is unspecified
but reserved. RECORD and other values may be specified in the
future.
interleaved: The interleaved parameter implies mixing the media
stream with the control stream in whatever protocol is being
used by the control stream, using the mechanism defined in
Section 14. The argument provides the channel number to be
used in the $ block (see Section 14) and MUST be present. This
parameter MAY be specified as an interval, e.g.,
interleaved=4-5 in cases where the transport choice for the
media stream requires it, e.g., for RTP with RTCP. The channel
number given in the request is only a guidance from the client
to the server on what channel number(s) to use. The server MAY
set any valid channel number in the response. The declared
channels are bidirectional, so both end parties MAY send data
on the given channel. One example of such usage is the second
channel used for RTCP, where both server and client send RTCP
packets on the same channel.
This allows RTP/RTCP to be handled similarly to the way that
it is done with UDP, i.e., one channel for RTP and the other
for RTCP.
MIKEY: This parameter is used in conjunction with transport
specifications that can utilize MIKEY [RFC3830] for security
context establishment. So far, only the SRTP-based RTP
profiles SAVP and SAVPF can utilize MIKEY, and this is defined
in Appendix C.1.4.1. This parameter can be included both in
request and response messages. The binary MIKEY message SHALL
be Base64-encoded [RFC4648] before being included in the value
part of the parameter, where the encoding adheres to the
definition in Section 4 of RFC 4648 and where the padding bits
are set to zero.
Multicast-specific:
ttl: multicast time-to-live for IPv4. When included in requests,
the value indicates the TTL value that the client requests the
server to use. In a response, the value actually being used by
the server is returned. A server will need to consider what
values that are reasonable and also the authority of the user
to set this value. Corresponding functions are not needed for
IPv6 as the scoping is part of the IPv6 multicast address
[RFC4291].
RTP-specific:
These parameters MAY only be used if the media-transport protocol is
RTP.
ssrc: The ssrc parameter, if included in a SETUP response, indicates
the RTP SSRC [RFC3550] value(s) that will be used by the media
server for RTP packets within the stream. The values are
expressed as a slash-separated sequence of SSRC values, each
SSRC expressed as an eight-digit hexadecimal value.
The ssrc parameter MUST NOT be specified in requests. The
functionality of specifying the ssrc parameter in a SETUP
request is deprecated as it is incompatible with the
specification of RTP [RFC3550]. If the parameter is included
in the Transport header of a SETUP request, the server SHOULD
ignore it, and choose appropriate SSRCs for the stream. The
server SHOULD set the ssrc parameter in the Transport header of
the response.
RTCP-mux: Used to negotiate the usage of RTP and RTCP multiplexing
[RFC5761] on a single underlying transport stream/flow. The
presence of this parameter in a SETUP request indicates the
client's support and requires the server to use RTP and RTCP
multiplexing. The client SHALL only include one transport
stream in the Transport header specification. To provide the
server with a choice between using RTP/RTCP multiplexing or
not, two different transport header specifications must be
included.
The parameter setup and connection defined below MAY only be used if
the media-transport protocol of the lower-level transport is
connection oriented (such as TCP). However, these parameters MUST
NOT be used when interleaving data over the RTSP connection.
setup: Clients use the setup parameter on the Transport line in a
SETUP request to indicate the roles it wishes to play in a TCP
connection. This parameter is adapted from [RFC4145]. The use
of this parameter in RTP/AVP/TCP non-interleaved transport is
discussed in Appendix C.2.2; the discussion below is limited to
syntactic issues. Clients may specify the following values for
the setup parameter:
active: The client will initiate an outgoing connection.
passive: The client will accept an incoming connection.
actpass: The client is willing to accept an incoming
connection or to initiate an outgoing connection.
If a client does not specify a setup value, the "active" value
is assumed.
In response to a client SETUP request where the setup parameter
is set to "active", a server's 2xx reply MUST assign the setup
parameter to "passive" on the Transport header line.
In response to a client SETUP request where the setup parameter
is set to "passive", a server's 2xx reply MUST assign the setup
parameter to "active" on the Transport header line.
In response to a client SETUP request where the setup parameter
is set to "actpass", a server's 2xx reply MUST assign the setup
parameter to "active" or "passive" on the Transport header
line.
Note that the "holdconn" value for setup is not defined for
RTSP use, and MUST NOT appear on a Transport line.
connection: Clients use the connection parameter in a transport
specification part of the Transport header in a SETUP request
to indicate the client's preference for either reusing an
existing connection between client and server (in which case
the client sets the "connection" parameter to "existing") or
requesting the creation of a new connection between client and
server (in which cast the client sets the "connection"
parameter to "new"). Typically, clients use the "new" value
for the first SETUP request for a URL, and "existing" for
subsequent SETUP requests for a URL.
If a client SETUP request assigns the "new" value to
"connection", the server response MUST also assign the "new"
value to "connection" on the Transport line.
If a client SETUP request assigns the "existing" value to
"connection", the server response MUST assign a value of
"existing" or "new" to "connection" on the Transport line, at
its discretion.
The default value of "connection" is "existing", for all SETUP
requests (initial and subsequent).
The combination of transport protocol, profile and lower transport
needs to be defined. A number of combinations are defined in the
Appendix C.
Below is a usage example, showing a client advertising the capability
to handle multicast or unicast, preferring multicast. Since this is
a unicast-only stream, the server responds with the proper transport
parameters for unicast.
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0
CSeq: 302
Transport: RTP/AVP;multicast;mode="PLAY",
RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
"192.0.2.5:3457";mode="PLAY"
Accept-Ranges: npt, smpte, clock
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 302
Date: Fri, 20 Dec 2013 10:20:32 +0000
Session: rQi1hBrGlFdiYld241FxUO
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
"192.0.2.5:3457";src_addr="192.0.2.224:6256"/
"192.0.2.224:6257";mode="PLAY"
Accept-Ranges: npt
Media-Properties: Random-Access=0.6, Dynamic,
Time-Limited=20081128T165900
18.55. Unsupported
The Unsupported response-header lists the features not supported by
the responding RTSP agent. In the case where the feature was
specified via the Proxy-Require field (Section 18.37), if there is a
proxy on the path between the client and the server, the proxy MUST
send a response message with a status code of 551 (Option Not
Supported). The request MUST NOT be forwarded.
See Section 18.43 for a usage example.
18.56. User-Agent
The User-Agent general-header field contains information about the
user agent originating the request or producing a response. This is
for statistical purposes, the tracing of protocol violations, and
automated recognition of user agents for the sake of tailoring
responses to avoid particular user agent limitations. User agents
SHOULD include this field with requests. The field can contain
multiple product tokens and comments identifying the agent and any
subproducts which form a significant part of the user agent. By
convention, the product tokens are listed in order of their
significance for identifying the application.
Example:
User-Agent: PhonyClient/1.2
18.57. Via
The Via general-header field MUST be used by proxies to indicate the
intermediate protocols and recipients between the user agent and the
server on requests and between the origin server and the client on
responses. The field is intended to be used for tracking message
forwards, avoiding request loops, and identifying the protocol
capabilities of all senders along the request/response chain.
Each of multiple values in the Via field represents each proxy that
has forwarded the message. Each recipient MUST append its
information such that the end result is ordered according to the
sequence of forwarding applications. So messages originating with
the client or server do not include the Via header. The first proxy
or other intermediate adds the header and its information into the
field. Any additional intermediate adds additional field-values.
Resulting in the server seeing the chains of intermediates in a
client-to-server request and the client seeing the full chain in the
response message.
Proxies (e.g., Access Proxy or Translator Proxy) SHOULD NOT, by
default, forward the names and ports of hosts within the private/
protected region. This information SHOULD only be propagated if
explicitly enabled. If not enabled, the via-received of any host
behind the firewall/NAT SHOULD be replaced by an appropriate
pseudonym for that host.
For organizations that have strong privacy requirements for hiding
internal structures, a proxy MAY combine an ordered subsequence of
Via header field entries with identical sent-protocol values into a
single such entry. Applications MUST NOT combine entries that have
different received-protocol values.
18.58. WWW-Authenticate
The WWW-Authenticate header is specified in [RFC7235]; its usage
depends on the used authentication schemes, such as Digest [RFC7616]
and Basic [RFC7617]. The WWW-Authenticate response-header field MUST
be included in 401 (Unauthorized) response messages. The field-value
consists of at least one challenge that indicates the authentication
scheme(s) and parameters applicable to the Request-URI. This header
MUST only be used in response messages related to client to server
requests.
The HTTP access authentication process is described in [RFC7235] with
some clarification in Section 19.1. User agents are advised to take
special care in parsing the WWW-Authenticate field-value as it might
contain more than one challenge, or if more than one WWW-Authenticate
header field is provided, the contents of a challenge itself can
contain a comma-separated list of authentication parameters.
19. Security Framework
The RTSP security framework consists of two high-level components:
the pure authentication mechanisms based on HTTP authentication and
the message transport protection based on TLS, which is independent
of RTSP. Because of the similarity in syntax and usage between RTSP
servers and HTTP servers, the security for HTTP is reused to a large
extent.
19.1. RTSP and HTTP Authentication
RTSP and HTTP share common authentication schemes; thus, they follow
the same framework as specified in [RFC7235]. RTSP uses the
corresponding RTSP error codes (401 and 407) and headers (WWW-
Authenticate, Authorization, Proxy-Authenticate, Proxy-Authorization)
by importing the definitions from [RFC7235]. Servers SHOULD
implement both the Basic [RFC7617] and the Digest [RFC7616]
authentication schemes. Clients MUST implement both the Basic and
the Digest authentication schemes so that a server that requires the
client to authenticate can trust that the capability is present. If
implementing the Digest authentication scheme, then the additional
considerations specified below in Section 19.1.1 MUST be followed.
It should be stressed that using the HTTP authentication alone does
not provide full RTSP message security. Therefore, TLS SHOULD be
used; see Section 19.2. Any RTSP message containing an Authorization
header using the Basic authentication scheme MUST be using a TLS
connection with confidentiality protection enabled, i.e., no NULL
encryption.
In cases where there is a chain of proxies between the client and the
server, each proxy may individually request the client or previous
proxy to authenticate itself. This is done using the Proxy-
Authenticate (Section 18.34), the Proxy-Authorization
(Section 18.36), and the Proxy-Authentication-Info (Section 18.35)
headers. These headers are hop-by-hop headers and are only scoped to
the current connection and hop. Thus, if a proxy chain exists, a
proxy connecting to another proxy will have to act as a client to
authorize itself towards the next proxy. The WWW-Authenticate
(Section 18.58), Authorization (Section 18.8), and Authentication-
Info (Section 18.7) headers are end-to-end and MUST NOT be modified
by proxies.
This authentication mechanism works only for client-to-server
requests as currently defined. This leaves server-to-client request
outside of the context of TLS-based communication more vulnerable to
message-injection attacks on the client. Based on the server-to-
client methods that exist, the potential risks are various: hijacking
(REDIRECT), denial of service (TEARDOWN and PLAY_NOTIFY), or attacks
with uncertain results (SET_PARAMETER).
19.1.1. Digest Authentication
This section describes the modifications and clarifications required
to apply the HTTP Digest authentication scheme to RTSP. The RTSP
scheme usage is almost completely identical to that for HTTP
[RFC7616]. These modifications are based on the procedures defined
for SIP 2.0 [RFC3261] (in Section 22.4) but updated to use RFC 7235,
RFC 7616 and RFC 7615 instead of RFC 2617.
Digest authentication uses two additional headers, Authentication-
Info and Proxy-Authentication-Info, that are defined as in [RFC7615].
The rules for Digest authentication follow those defined in
[RFC7616], with "HTTP/1.1" replaced by "RTSP/2.0" in addition to the
following differences:
1. Use the ABNF specified in the referenced documents, with the
difference that the URI parameter uses the request URI format for
RTSP, i.e. the ABNF element: Request-URI (see Section 20.2.1).
The domain parameter uses the RTSP-URI-Ref element for absolute
and relative URIs.
2. If MTags are used, then the example procedure for choosing a
nonce based on ETag can work, based on replacing the ETag with
the MTag.
3. As a clarification to the calculation of the A2 value for message
integrity assurance in the Digest authentication scheme,
implementers should assume, when the entity-body is empty (that
is, when the RTSP messages have no message body) that the hash of
the message body resolves to the hash of an empty string, or:
H(entity-body), example MD5("") =
"d41d8cd98f00b204e9800998ecf8427e".
19.2. RTSP over TLS
RTSP agents MUST implement RTSP over TLS as defined in this section
and the next Section 19.3. RTSP MUST follow the same guidelines with
regard to TLS [RFC5246] usage as specified for HTTP; see [RFC2818].
RTSP over TLS is separated from unsecured RTSP both on the URI level
and the port level. Instead of using the "rtsp" scheme identifier in
the URI, the "rtsps" scheme identifier MUST be used to signal RTSP
over TLS. If no port is given in a URI with the "rtsps" scheme, port
322 MUST be used for TLS over TCP/IP.
When a client tries to set up an insecure channel to the server
(using the "rtsp" URI), and the policy for the resource requires a
secure channel, the server MUST redirect the client to the secure
service by sending a 301 redirect response code together with the
correct Location URI (using the "rtsps" scheme). A user or client
MAY upgrade a non secured URI to a secured by changing the scheme
from "rtsp" to "rtsps". A server implementing support for "rtsps"
MUST allow this.
It should be noted that TLS allows for mutual authentication (when
using both server and client certificates). Still, one of the more
common ways TLS is used is to provide only server-side authentication
(often to avoid client certificates). TLS is then used in addition
to HTTP authentication, providing transport security and server
authentication, while HTTP Authentication is used to authenticate the
client.
RTSP includes the possibility to keep a TCP session up between the
client and server, throughout the RTSP session lifetime. It may be
convenient to keep the TCP session, not only to save the extra setup
time for TCP, but also the extra setup time for TLS (even if TLS uses
the resume function, there will be almost two extra round trips).
Still, when TLS is used, such behavior introduces extra active state
in the server, not only for TCP and RTSP, but also for TLS. This may
increase the vulnerability to DoS attacks.
There exists a potential security vulnerability when reusing TCP and
TLS state for different resources (URIs). If two different hostnames
point at the same IP address, it can be desirable to reuse the TCP/
TLS connection to that server. In that case, the RTSP agent having
the TCP/TLS connection MUST verify that the server certificate
associated with the connection has a SubjectAltName matching the
hostname present in the URI for the resource an RTSP request is to be
issued.
In addition to these recommendations, Section 19.3 gives further
recommendations of TLS usage with proxies.
19.3. Security and Proxies
The nature of a proxy is often to act as a "man in the middle", while
security is often about preventing the existence of one. This
section provides clients with the possibility to use proxies even
when applying secure transports (TLS) between the RTSP agents. The
TLS proxy mechanism allows for server and proxy identification using
certificates. However, the client cannot be identified based on
certificates. The client needs to select between using the procedure
specified below or using a TLS connection directly (bypassing any
proxies) to the server. The choice may be dependent on policies.
In general, there are two categories of proxies: the transparent
proxies (of which the client is not aware) and the non-transparent
proxies (of which the client is aware). This memo specifies only
non-transparent RTSP proxies, i.e., proxies visible to the RTSP
client and RTSP server. An infrastructure based on proxies requires
that the trust model be such that both client and server can trust
the proxies to handle the RTSP messages correctly. To be able to
trust a proxy, the client and server also need to be aware of the
proxy. Hence, transparent proxies cannot generally be seen as
trusted and will not work well with security (unless they work only
at the transport layer). In the rest of this section, any reference
to "proxy" will be to a non-transparent proxy, which inspects or
manipulates the RTSP messages.
HTTP Authentication is built on the assumption of proxies and can
provide user-proxy authentication and proxy-proxy/server
authentication in addition to the client-server authentication.
When TLS is applied and a proxy is used, the client will connect to
the proxy's address when connecting to any RTSP server. This implies
that for TLS, the client will authenticate the proxy server and not
the end server. Note that when the client checks the server
certificate in TLS, it MUST check the proxy's identity (URI or
possibly other known identity) against the proxy's identity as
presented in the proxy's Certificate message.
The problem is that for a proxy accepted by the client, the proxy
needs to be provided information on which grounds it should accept
the next-hop certificate. Both the proxy and the user may have rules
for this, and the user should have the possibility to select the
desired behavior. To handle this case, the Accept-Credentials header
(see Section 18.2) is used, where the client can request the proxy or
proxies to relay back the chain of certificates used to authenticate
any intermediate proxies as well as the server. The assumption that
the proxies are viewed as trusted gives the user a possibility to
enforce policies on each trusted proxy of whether it should accept
the next agent in the chain. However, it should be noted that not
all deployments will return the chain of certificates used to
authenticate any intermediate proxies as well as the server. An
operator of such a deployment may want to hide its topology from the
client. It should be noted well that the client does not have any
insight into the proxy's operation. Even if the proxy is trusted, it
can still return an incomplete chain of certificates.
A proxy MUST use TLS for the next hop if the RTSP request includes an
"rtsps" URI. TLS MAY be applied on intermediate links (e.g., between
client and proxy or between proxy and proxy) even if the resource and
the end server are not required to use it. The chain of proxies used
by a client to reach a server and its TLS sessions MUST have
commensurate security. Therefore, a proxy MUST, when initiating the
next-hop TLS connection, use the incoming TLS connections cipher-
suite list, only modified by removing any cipher suites that the
proxy does not support. In case a proxy fails to establish a TLS
connection due to cipher-suite mismatch between proxy and next-hop
proxy or server, this is indicated using error code 472 (Failure to
Establish Secure Connection).
19.3.1. Accept-Credentials
The Accept-Credentials header can be used by the client to distribute
simple authorization policies to intermediate proxies. The client
includes the Accept-Credentials header to dictate how the proxy
treats the server / next proxy certificate. There are currently
three methods defined:
Any: With "any", the proxy (or proxies) MUST accept whatever
certificate is presented. Of course, this is not a recommended
option to use, but it may be useful in certain circumstances
(such as testing).
Proxy: For the "proxy" method, the proxy (or proxies) MUST use its
own policies to validate the certificate and decide whether or
not to accept it. This is convenient in cases where the user
has a strong trust relation with the proxy. Reasons why a
strong trust relation may exist are personal/company proxy or
the proxy has an out-of-band policy configuration mechanism.
User: For the "user" method, the proxy (or proxies) MUST send
credential information about the next hop to the client for
authorization. The client can then decide whether or not the
proxy should accept the certificate. See Section 19.3.2 for
further details.
If the Accept-Credentials header is not included in the RTSP request
from the client, then the "Proxy" method MUST be used as default. If
a method other than the "Proxy" is to be used, then the Accept-
Credentials header MUST be included in all of the RTSP requests from
the client. This is because it cannot be assumed that the proxy
always keeps the TLS state or the user's previous preference between
different RTSP messages (in particular, if the time interval between
the messages is long).
With the "Any" and "Proxy" methods, the proxy will apply the policy
as defined for each method. If the policy does not accept the
credentials of the next hop, the proxy MUST respond with a message
using status code 471 (Connection Credentials Not Accepted).
An RTSP request in the direction server to client MUST NOT include
the Accept-Credentials header. As for the non-secured communication,
the possibility for these requests depends on the presence of a
client established connection. However, if the server-to-client
request is in relation to a session established over a TLS secured
channel, it MUST be sent in a TLS secured connection. That secured
connection MUST also be the one used by the last client-to-server
request. If no such transport connection exists at the time when the
server desires to send the request, the server MUST discard the
message.
Further policies MAY be defined and registered, but this should be
done with caution.
19.3.2. User-Approved TLS Procedure
For the "User" method, each proxy MUST perform the following
procedure for each RTSP request:
o Set up the TLS session to the next hop if not already present
(i.e., run the TLS handshake, but do not send the RTSP request).
o Extract the peer certificate chain for the TLS session.
o Check if a matching identity and hash of the peer certificate are
present in the Accept-Credentials header. If present, send the
message to the next hop and conclude these procedures. If not, go
to the next step.
o The proxy responds to the RTSP request with a 470 or 407 response
code. The 407 response code MAY be used when the proxy requires
both user and connection authorization from user or client. In
this message the proxy MUST include a Connection-Credentials
header, see Section 18.13, with the next hop's identity and
certificate.
The client MUST upon receiving a 470 (Connection Authorization
Required) or 407 (Proxy Authentication Required) response with
Connection-Credentials header take the decision on whether or not to
accept the certificate (if it cannot do so, the user SHOULD be
consulted). Using IP addresses in the next-hop URI and certificates
rather than domain names makes it very difficult for a user to
determine whether or not it should approve the next hop. Proxies are
RECOMMENDED to use domain names to identify themselves in URIs and in
the certificates. If the certificate is accepted, the client has to
again send the RTSP request. In that request, the client has to
include the Accept-Credentials header including the hash over the
DER-encoded certificate for all trusted proxies in the chain.
Example:
C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
CSeq: 2
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
"192.0.2.5:4589"
Accept-Ranges: npt, smpte, clock
Accept-Credentials: User
P->C: RTSP/2.0 470 Connection Authorization Required
CSeq: 2
Connection-Credentials: "rtsps://test.example.org";
MIIDNTCCAp...
C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
CSeq: 3
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
"192.0.2.5:4589"
Accept-Credentials: User "rtsps://test.example.org";sha-256;
dPYD7txpoGTbAqZZQJ+vaeOkyH4=
Accept-Ranges: npt, smpte, clock
P->S: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
CSeq: 3
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
"192.0.2.5:4589"
Via: RTSP/2.0 proxy.example.org
Accept-Credentials: User "rtsps://test.example.org";sha-256;
dPYD7txpoGTbAqZZQJ+vaeOkyH4=
Accept-Ranges: npt, smpte, clock
One implication of this process is that the connection for secured
RTSP messages may take significantly more round-trip times for the
first message. A complete extra message exchange between the proxy
connecting to the next hop and the client results because of the
process for approval for each hop. However, if each message contains
the chain of proxies that the requester accepts, the remaining
message exchange should not be delayed. The procedure of including
the credentials in each request rather than building state in each
proxy avoids the need for revocation procedures.
20. Syntax
The RTSP syntax is described in an Augmented Backus-Naur Form (ABNF)
as defined in RFC 5234 [RFC5234]. It uses the basic definitions
present in RFC 5234.
Please note that ABNF strings, e.g., "Accept", are case insensitive
as specified in Section 2.3 of RFC 5234.
The RTSP syntax makes use of the ISO 10646 character set in UTF-8
encoding [RFC3629].
20.1. Base Syntax
RTSP header values can be folded onto multiple lines if the
continuation line begins with a space or horizontal tab. All linear
whitespace, including folding, has the same semantics as SP. A
recipient MAY replace any linear whitespace with a single SP before
interpreting the field-value or forwarding the message downstream.
The SWS construct is used when linear whitespace is optional,
generally between tokens and separators.
To separate the header name from the rest of value, a colon is used,
which, by the above rule, allows whitespace before, but no line
break, and whitespace after, including a line break. The HCOLON
defines this construct.
OCTET = %x00-FF ; any 8-bit sequence of data
CHAR = %x01-7F ; any US-ASCII character (octets 1 - 127)
UPALPHA = %x41-5A ; any US-ASCII uppercase letter "A".."Z"
LOALPHA = %x61-7A ; any US-ASCII lowercase letter "a".."z"
ALPHA = UPALPHA / LOALPHA
DIGIT = %x30-39 ; any US-ASCII digit "0".."9"
CTL = %x00-1F / %x7F ; any US-ASCII control character
; (octets 0 - 31) and DEL (127)
CR = %x0D ; US-ASCII CR, carriage return (13)
LF = %x0A ; US-ASCII LF, linefeed (10)
SP = %x20 ; US-ASCII SP, space (32)
HT = %x09 ; US-ASCII HT, horizontal-tab (9)
BACKSLASH = %x5C ; US-ASCII backslash (92)
CRLF = CR LF
LWS = [CRLF] 1*( SP / HT ) ; Line-breaking whitespace
SWS = [LWS] ; Separating whitespace
HCOLON = *( SP / HT ) ":" SWS
TEXT = %x20-7E / %x80-FF ; any OCTET except CTLs
tspecials = "(" / ")" / "<" / ">" / "@"
/ "," / ";" / ":" / BACKSLASH / DQUOTE
/ "/" / "[" / "]" / "?" / "="
/ "{" / "}" / SP / HT
token = 1*(%x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39
/ %x41-5A / %x5E-7A / %x7C / %x7E)
; 1*<any CHAR except CTLs or tspecials>
quoted-string = ( DQUOTE *qdtext DQUOTE )
qdtext = %x20-21 / %x23-5B / %x5D-7E / quoted-pair
/ UTF8-NONASCII
; No DQUOTE and no "\"
quoted-pair = "\\" / ( "\" DQUOTE )
ctext = %x20-27 / %x2A-7E
/ %x80-FF ; any OCTET except CTLs, "(" and ")"
generic-param = token [ EQUAL gen-value ]
gen-value = token / host / quoted-string
safe = "$" / "-" / "_" / "." / "+"
extra = "!" / "*" / "'" / "(" / ")" / ","
rtsp-extra = "!" / "*" / "'" / "(" / ")"
HEX = DIGIT / "A" / "B" / "C" / "D" / "E" / "F"
/ "a" / "b" / "c" / "d" / "e" / "f"
LHEX = DIGIT / "a" / "b" / "c" / "d" / "e" / "f"
; lowercase "a-f" Hex
reserved = ";" / "/" / "?" / ":" / "@" / "&" / "="
unreserved = ALPHA / DIGIT / safe / extra
rtsp-unreserved = ALPHA / DIGIT / safe / rtsp-extra
base64 = *base64-unit [base64-pad]
base64-unit = 4base64-char
base64-pad = (2base64-char "==") / (3base64-char "=")
base64-char = ALPHA / DIGIT / "+" / "/"
SLASH = SWS "/" SWS ; slash
EQUAL = SWS "=" SWS ; equal
LPAREN = SWS "(" SWS ; left parenthesis
RPAREN = SWS ")" SWS ; right parenthesis
COMMA = SWS "," SWS ; comma
SEMI = SWS ";" SWS ; semicolon
COLON = SWS ":" SWS ; colon
MINUS = SWS "-" SWS ; minus/dash
LDQUOT = SWS DQUOTE ; open double quotation mark
RDQUOT = DQUOTE SWS ; close double quotation mark
RAQUOT = ">" SWS ; right angle quote
LAQUOT = SWS "<" ; left angle quote
TEXT-UTF8char = %x21-7E / UTF8-NONASCII
UTF8-NONASCII = UTF8-2 / UTF8-3 / UTF8-4
UTF8-1 = <As defined in RFC 3629>
UTF8-2 = <As defined in RFC 3629>
UTF8-3 = <As defined in RFC 3629>
UTF8-4 = <As defined in RFC 3629>
UTF8-tail = <As defined in RFC 3629>
POS-FLOAT = 1*12DIGIT ["." 1*9DIGIT]
FLOAT = ["-"] POS-FLOAT
20.2. RTSP Protocol Definition
20.2.1. Generic Protocol Elements
RTSP-IRI = schemes ":" IRI-rest
IRI-rest = ihier-part [ "?" iquery ]
ihier-part = "//" iauthority ipath-abempty
RTSP-IRI-ref = RTSP-IRI / irelative-ref
irelative-ref = irelative-part [ "?" iquery ]
irelative-part = "//" iauthority ipath-abempty
/ ipath-absolute
/ ipath-noscheme
/ ipath-empty
iauthority = < As defined in RFC 3987>
ipath = ipath-abempty ; begins with "/" or is empty
/ ipath-absolute ; begins with "/" but not "//"
/ ipath-noscheme ; begins with a non-colon segment
/ ipath-rootless ; begins with a segment
/ ipath-empty ; zero characters
ipath-abempty = *( "/" isegment )
ipath-absolute = "/" [ isegment-nz *( "/" isegment ) ]
ipath-noscheme = isegment-nz-nc *( "/" isegment )
ipath-rootless = isegment-nz *( "/" isegment )
ipath-empty = 0<ipchar>
isegment = *ipchar [";" *ipchar]
isegment-nz = 1*ipchar [";" *ipchar]
/ ";" *ipchar
isegment-nz-nc = (1*ipchar-nc [";" *ipchar-nc])
/ ";" *ipchar-nc
; non-zero-length segment without any colon ":"
; No parameter (; delimited) inside path.
ipchar = iunreserved / pct-encoded / sub-delims / ":" / "@"
ipchar-nc = iunreserved / pct-encoded / sub-delims / "@"
; sub-delims is different from RFC 3987
; not including ";"
iquery = < As defined in RFC 3987>
iunreserved = < As defined in RFC 3987>
pct-encoded = < As defined in RFC 3987>
RTSP-URI = schemes ":" URI-rest
RTSP-REQ-URI = schemes ":" URI-req-rest
RTSP-URI-Ref = RTSP-URI / RTSP-Relative
RTSP-REQ-Ref = RTSP-REQ-URI / RTSP-REQ-Rel
schemes = "rtsp" / "rtsps" / scheme
scheme = < As defined in RFC 3986>
URI-rest = hier-part [ "?" query ]
URI-req-rest = hier-part [ "?" query ]
; Note fragment part not allowed in requests
hier-part = "//" authority path-abempty
RTSP-Relative = relative-part [ "?" query ]
RTSP-REQ-Rel = relative-part [ "?" query ]
relative-part = "//" authority path-abempty
/ path-absolute
/ path-noscheme
/ path-empty
authority = < As defined in RFC 3986>
query = < As defined in RFC 3986>
path = path-abempty ; begins with "/" or is empty
/ path-absolute ; begins with "/" but not "//"
/ path-noscheme ; begins with a non-colon segment
/ path-rootless ; begins with a segment
/ path-empty ; zero characters
path-abempty = *( "/" segment )
path-absolute = "/" [ segment-nz *( "/" segment ) ]
path-noscheme = segment-nz-nc *( "/" segment )
path-rootless = segment-nz *( "/" segment )
path-empty = 0<pchar>
segment = *pchar [";" *pchar]
segment-nz = ( 1*pchar [";" *pchar]) / (";" *pchar)
segment-nz-nc = ( 1*pchar-nc [";" *pchar-nc]) / (";" *pchar-nc)
; non-zero-length segment without any colon ":"
; No parameter (; delimited) inside path.
pchar = unreserved / pct-encoded / sub-delims / ":" / "@"
pchar-nc = unreserved / pct-encoded / sub-delims / "@"
sub-delims = "!" / "$" / "&" / "'" / "(" / ")"
/ "*" / "+" / "," / "="
; sub-delims is different from RFC 3986/3987
; not including ";"
smpte-range = smpte-type [EQUAL smpte-range-spec]
; See section 4.4
smpte-range-spec = ( smpte-time "-" [ smpte-time ] )
/ ( "-" smpte-time )
smpte-type = "smpte" / "smpte-30-drop"
/ "smpte-25" / smpte-type-extension
; other timecodes may be added
smpte-type-extension = "smpte" token
smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT
[ ":" 1*2DIGIT [ "." 1*2DIGIT ] ]
npt-range = "npt" [EQUAL npt-range-spec]
npt-range-spec = ( npt-time "-" [ npt-time ] ) / ( "-" npt-time )
npt-time = "now" / npt-sec / npt-hhmmss / npt-hhmmss-comp
npt-sec = 1*19DIGIT [ "." 1*9DIGIT ]
npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." 1*9DIGIT ]
npt-hh = 2*19DIGIT ; any positive number
npt-mm = 2*2DIGIT ; 0-59
npt-ss = 2*2DIGIT ; 0-59
npt-hhmmss-comp = npt-hh-comp ":" npt-mm-comp ":" npt-ss-comp
[ "." 1*9DIGIT ] ; Compatibility format
npt-hh-comp = 1*19DIGIT ; any positive number
npt-mm-comp = 1*2DIGIT ; 0-59
npt-ss-comp = 1*2DIGIT ; 0-59
utc-range = "clock" [EQUAL utc-range-spec]
utc-range-spec = ( utc-time "-" [ utc-time ] ) / ( "-" utc-time )
utc-time = utc-date "T" utc-clock "Z"
utc-date = 8DIGIT
utc-clock = 6DIGIT [ "." 1*9DIGIT ]
feature-tag = token
session-id = 1*256( ALPHA / DIGIT / safe )
extension-header = header-name HCOLON header-value
header-name = token
header-value = *(TEXT-UTF8char / LWS)
20.2.2. Message Syntax
RTSP-message = Request / Response ; RTSP/2.0 messages
Request = Request-Line
*((general-header
/ request-header
/ message-body-header) CRLF)
CRLF
[ message-body-data ]
Response = Status-Line
*((general-header
/ response-header
/ message-body-header) CRLF)
CRLF
[ message-body-data ]
Request-Line = Method SP Request-URI SP RTSP-Version CRLF
Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF
Method = "DESCRIBE"
/ "GET_PARAMETER"
/ "OPTIONS"
/ "PAUSE"
/ "PLAY"
/ "PLAY_NOTIFY"
/ "REDIRECT"
/ "SETUP"
/ "SET_PARAMETER"
/ "TEARDOWN"
/ extension-method
extension-method = token
Request-URI = "*" / RTSP-REQ-URI
RTSP-Version = "RTSP/" 1*DIGIT "." 1*DIGIT
message-body-data = 1*OCTET
Status-Code = "100" ; Continue
/ "200" ; OK
/ "301" ; Moved Permanently
/ "302" ; Found
/ "303" ; See Other
/ "304" ; Not Modified
/ "305" ; Use Proxy
/ "400" ; Bad Request
/ "401" ; Unauthorized
/ "402" ; Payment Required
/ "403" ; Forbidden
/ "404" ; Not Found
/ "405" ; Method Not Allowed
/ "406" ; Not Acceptable
/ "407" ; Proxy Authentication Required
/ "408" ; Request Timeout
/ "410" ; Gone
/ "412" ; Precondition Failed
/ "413" ; Request Message Body Too Large
/ "414" ; Request-URI Too Long
/ "415" ; Unsupported Media Type
/ "451" ; Parameter Not Understood
/ "452" ; reserved
/ "453" ; Not Enough Bandwidth
/ "454" ; Session Not Found
/ "455" ; Method Not Valid In This State
/ "456" ; Header Field Not Valid for Resource
/ "457" ; Invalid Range
/ "458" ; Parameter Is Read-Only
/ "459" ; Aggregate Operation Not Allowed
/ "460" ; Only Aggregate Operation Allowed
/ "461" ; Unsupported Transport
/ "462" ; Destination Unreachable
/ "463" ; Destination Prohibited
/ "464" ; Data Transport Not Ready Yet
/ "465" ; Notification Reason Unknown
/ "466" ; Key Management Error
/ "470" ; Connection Authorization Required
/ "471" ; Connection Credentials Not Accepted
/ "472" ; Failure to Establish Secure Connection
/ "500" ; Internal Server Error
/ "501" ; Not Implemented
/ "502" ; Bad Gateway
/ "503" ; Service Unavailable
/ "504" ; Gateway Timeout
/ "505" ; RTSP Version Not Supported
/ "551" ; Option Not Supported
/ "553" ; Proxy Unavailable
/ extension-code
extension-code = 3DIGIT
Reason-Phrase = 1*(TEXT-UTF8char / HT / SP)
rtsp-header = general-header
/ request-header
/ response-header
/ message-body-header
general-header = Accept-Ranges
/ Cache-Control
/ Connection
/ CSeq
/ Date
/ Media-Properties
/ Media-Range
/ Pipelined-Requests
/ Proxy-Supported
/ Range
/ RTP-Info
/ Scale
/ Seek-Style
/ Server
/ Session
/ Speed
/ Supported
/ Timestamp
/ Transport
/ User-Agent
/ Via
/ extension-header
request-header = Accept
/ Accept-Credentials
/ Accept-Encoding
/ Accept-Language
/ Authorization
/ Bandwidth
/ Blocksize
/ From
/ If-Match
/ If-Modified-Since
/ If-None-Match
/ Notify-Reason
/ Proxy-Authorization
/ Proxy-Require
/ Referrer
/ Request-Status
/ Require
/ Terminate-Reason
/ extension-header
response-header = Authentication-Info
/ Connection-Credentials
/ Location
/ MTag
/ Proxy-Authenticate
/ Proxy-Authentication-Info
/ Public
/ Retry-After
/ Unsupported
/ WWW-Authenticate
/ extension-header
message-body-header = Allow
/ Content-Base
/ Content-Encoding
/ Content-Language
/ Content-Length
/ Content-Location
/ Content-Type
/ Expires
/ Last-Modified
/ extension-header
20.2.3. Header Syntax
Accept = "Accept" HCOLON
[ accept-range *(COMMA accept-range) ]
accept-range = media-type-range [SEMI accept-params]
media-type-range = ( "*/*"
/ ( m-type SLASH "*" )
/ ( m-type SLASH m-subtype )
) *( SEMI m-parameter )
accept-params = "q" EQUAL qvalue *(SEMI generic-param )
qvalue = ( "0" [ "." *3DIGIT ] )
/ ( "1" [ "." *3("0") ] )
Accept-Credentials = "Accept-Credentials" HCOLON cred-decision
cred-decision = ("User" [LWS cred-info])
/ "Proxy"
/ "Any"
/ (token [LWS 1*header-value])
; For future extensions
cred-info = cred-info-data *(COMMA cred-info-data)
cred-info-data = DQUOTE RTSP-REQ-URI DQUOTE SEMI hash-alg
SEMI base64
hash-alg = "sha-256" / extension-alg
extension-alg = token
Accept-Encoding = "Accept-Encoding" HCOLON
[ encoding *(COMMA encoding) ]
encoding = codings [SEMI accept-params]
codings = content-coding / "*"
content-coding = "identity" / token
Accept-Language = "Accept-Language" HCOLON
language *(COMMA language)
language = language-range [SEMI accept-params]
language-range = language-tag / "*"
language-tag = primary-tag *( "-" subtag )
primary-tag = 1*8ALPHA
subtag = 1*8ALPHA
Accept-Ranges = "Accept-Ranges" HCOLON acceptable-ranges
acceptable-ranges = (range-unit *(COMMA range-unit))
range-unit = "npt" / "smpte" / "smpte-30-drop" / "smpte-25"
/ "clock" / extension-format
extension-format = token
Allow = "Allow" HCOLON Method *(COMMA Method)
Authentication-Info = "Authentication-Info" HCOLON auth-param-list
auth-param-list = <As the Authentication-Info element in RFC 7615>
Authorization = "Authorization" HCOLON credentials
credentials = <As defined by RFC 7235>
Bandwidth = "Bandwidth" HCOLON 1*19DIGIT
Blocksize = "Blocksize" HCOLON 1*9DIGIT
Cache-Control = "Cache-Control" HCOLON cache-directive
*(COMMA cache-directive)
cache-directive = cache-rqst-directive
/ cache-rspns-directive
cache-rqst-directive = "no-cache"
/ "max-stale" [EQUAL delta-seconds]
/ "min-fresh" EQUAL delta-seconds
/ "only-if-cached"
/ cache-extension
cache-rspns-directive = "public"
/ "private"
/ "no-cache"
/ "no-transform"
/ "must-revalidate"
/ "proxy-revalidate"
/ "max-age" EQUAL delta-seconds
/ cache-extension
cache-extension = token [EQUAL (token / quoted-string)]
delta-seconds = 1*19DIGIT
Connection = "Connection" HCOLON connection-token
*(COMMA connection-token)
connection-token = "close" / token
Connection-Credentials = "Connection-Credentials" HCOLON cred-chain
cred-chain = DQUOTE RTSP-REQ-URI DQUOTE SEMI base64
Content-Base = "Content-Base" HCOLON RTSP-URI
Content-Encoding = "Content-Encoding" HCOLON
content-coding *(COMMA content-coding)
Content-Language = "Content-Language" HCOLON
language-tag *(COMMA language-tag)
Content-Length = "Content-Length" HCOLON 1*19DIGIT
Content-Location = "Content-Location" HCOLON RTSP-REQ-Ref
Content-Type = "Content-Type" HCOLON media-type
media-type = m-type SLASH m-subtype *(SEMI m-parameter)
m-type = discrete-type / composite-type
discrete-type = "text" / "image" / "audio" / "video"
/ "application" / extension-token
composite-type = "message" / "multipart" / extension-token
extension-token = ietf-token / x-token
ietf-token = token
x-token = "x-" token
m-subtype = extension-token / iana-token
iana-token = token
m-parameter = m-attribute EQUAL m-value
m-attribute = token
m-value = token / quoted-string
CSeq = "CSeq" HCOLON cseq-nr
cseq-nr = 1*9DIGIT
Date = "Date" HCOLON RTSP-date
RTSP-date = date-time ;
date-time = <As defined in RFC 5322>
Expires = "Expires" HCOLON RTSP-date
From = "From" HCOLON from-spec
from-spec = ( name-addr / addr-spec ) *( SEMI from-param )
name-addr = [ display-name ] LAQUOT addr-spec RAQUOT
addr-spec = RTSP-REQ-URI / absolute-URI
absolute-URI = < As defined in RFC 3986>
display-name = *(token LWS) / quoted-string
from-param = tag-param / generic-param
tag-param = "tag" EQUAL token
If-Match = "If-Match" HCOLON ("*" / message-tag-list)
message-tag-list = message-tag *(COMMA message-tag)
message-tag = [ weak ] opaque-tag
weak = "W/"
opaque-tag = quoted-string
If-Modified-Since = "If-Modified-Since" HCOLON RTSP-date
If-None-Match = "If-None-Match" HCOLON ("*" / message-tag-list)
Last-Modified = "Last-Modified" HCOLON RTSP-date
Location = "Location" HCOLON RTSP-REQ-URI
Media-Properties = "Media-Properties" HCOLON [media-prop-list]
media-prop-list = media-prop-value *(COMMA media-prop-value)
media-prop-value = ("Random-Access" [EQUAL POS-FLOAT])
/ "Beginning-Only"
/ "No-Seeking"
/ "Immutable"
/ "Dynamic"
/ "Time-Progressing"
/ "Unlimited"
/ ("Time-Limited" EQUAL utc-time)
/ ("Time-Duration" EQUAL POS-FLOAT)
/ ("Scales" EQUAL scale-value-list)
/ media-prop-ext
media-prop-ext = token [EQUAL (1*rtsp-unreserved / quoted-string)]
scale-value-list = DQUOTE scale-entry *(COMMA scale-entry) DQUOTE
scale-entry = scale-value / (scale-value COLON scale-value)
scale-value = FLOAT
Media-Range = "Media-Range" HCOLON [ranges-list]
ranges-list = ranges-spec *(COMMA ranges-spec)
MTag = "MTag" HCOLON message-tag
Notify-Reason = "Notify-Reason" HCOLON Notify-Reas-val
Notify-Reas-val = "end-of-stream"
/ "media-properties-update"
/ "scale-change"
/ Notify-Reason-extension
Notify-Reason-extension = token
Pipelined-Requests = "Pipelined-Requests" HCOLON startup-id
startup-id = 1*8DIGIT
Proxy-Authenticate = "Proxy-Authenticate" HCOLON challenge-list
challenge-list = <As defined by the WWW-Authenticate in RFC 7235>
Proxy-Authentication-Info = "Proxy-Authentication-Info" HCOLON
auth-param-list
Proxy-Authorization = "Proxy-Authorization" HCOLON credentials
Proxy-Require = "Proxy-Require" HCOLON feature-tag-list
feature-tag-list = feature-tag *(COMMA feature-tag)
Proxy-Supported = "Proxy-Supported" HCOLON [feature-tag-list]
Public = "Public" HCOLON Method *(COMMA Method)
Range = "Range" HCOLON ranges-spec
ranges-spec = npt-range / utc-range / smpte-range
/ range-ext
range-ext = extension-format [EQUAL range-value]
range-value = 1*(rtsp-unreserved / quoted-string / ":" )
Referrer = "Referrer" HCOLON (absolute-URI / RTSP-URI-Ref)
Request-Status = "Request-Status" HCOLON req-status-info
req-status-info = cseq-info LWS status-info LWS reason-info
cseq-info = "cseq" EQUAL cseq-nr
status-info = "status" EQUAL Status-Code
reason-info = "reason" EQUAL DQUOTE Reason-Phrase DQUOTE
Require = "Require" HCOLON feature-tag-list
RTP-Info = "RTP-Info" HCOLON [rtsp-info-spec
*(COMMA rtsp-info-spec)]
rtsp-info-spec = stream-url 1*ssrc-parameter
stream-url = "url" EQUAL DQUOTE RTSP-REQ-Ref DQUOTE
ssrc-parameter = LWS "ssrc" EQUAL ssrc HCOLON
ri-parameter *(SEMI ri-parameter)
ri-parameter = ("seq" EQUAL 1*5DIGIT)
/ ("rtptime" EQUAL 1*10DIGIT)
/ generic-param
Retry-After = "Retry-After" HCOLON (RTSP-date / delta-seconds)
Scale = "Scale" HCOLON scale-value
Seek-Style = "Seek-Style" HCOLON Seek-S-values
Seek-S-values = "RAP"
/ "CoRAP"
/ "First-Prior"
/ "Next"
/ Seek-S-value-ext
Seek-S-value-ext = token
Server = "Server" HCOLON ( product / comment )
*(LWS (product / comment))
product = token [SLASH product-version]
product-version = token
comment = LPAREN *( ctext / quoted-pair) RPAREN
Session = "Session" HCOLON session-id
[ SEMI "timeout" EQUAL delta-seconds ]
Speed = "Speed" HCOLON lower-bound MINUS upper-bound
lower-bound = POS-FLOAT
upper-bound = POS-FLOAT
Supported = "Supported" HCOLON [feature-tag-list]
Terminate-Reason = "Terminate-Reason" HCOLON TR-Info
TR-Info = TR-Reason *(SEMI TR-Parameter)
TR-Reason = "Session-Timeout"
/ "Server-Admin"
/ "Internal-Error"
/ token
TR-Parameter = TR-time / TR-user-msg / generic-param
TR-time = "time" EQUAL utc-time
TR-user-msg = "user-msg" EQUAL quoted-string
Timestamp = "Timestamp" HCOLON timestamp-value [LWS delay]
timestamp-value = *19DIGIT [ "." *9DIGIT ]
delay = *9DIGIT [ "." *9DIGIT ]
Transport = "Transport" HCOLON transport-spec
*(COMMA transport-spec)
transport-spec = transport-id *trns-parameter
transport-id = trans-id-rtp / other-trans
trans-id-rtp = "RTP/" profile ["/" lower-transport]
; no LWS is allowed inside transport-id
other-trans = token *("/" token)
profile = "AVP" / "SAVP" / "AVPF" / "SAVPF" / token
lower-transport = "TCP" / "UDP" / token
trns-parameter = (SEMI ( "unicast" / "multicast" ))
/ (SEMI "interleaved" EQUAL channel ["-" channel])
/ (SEMI "ttl" EQUAL ttl)
/ (SEMI "layers" EQUAL 1*DIGIT)
/ (SEMI "ssrc" EQUAL ssrc *(SLASH ssrc))
/ (SEMI "mode" EQUAL mode-spec)
/ (SEMI "dest_addr" EQUAL addr-list)
/ (SEMI "src_addr" EQUAL addr-list)
/ (SEMI "setup" EQUAL contrans-setup)
/ (SEMI "connection" EQUAL contrans-con)
/ (SEMI "RTCP-mux")
/ (SEMI "MIKEY" EQUAL MIKEY-Value)
/ (SEMI trn-param-ext)
contrans-setup = "active" / "passive" / "actpass"
contrans-con = "new" / "existing"
trn-param-ext = par-name [EQUAL trn-par-value]
par-name = token
trn-par-value = *(rtsp-unreserved / quoted-string)
ttl = 1*3DIGIT ; 0 to 255
ssrc = 8HEX
channel = 1*3DIGIT ; 0 to 255
MIKEY-Value = base64
mode-spec = ( DQUOTE mode *(COMMA mode) DQUOTE )
mode = "PLAY" / token
addr-list = quoted-addr *(SLASH quoted-addr)
quoted-addr = DQUOTE (host-port / extension-addr) DQUOTE
host-port = ( host [":" port] )
/ ( ":" port )
extension-addr = 1*qdtext
host = < As defined in RFC 3986>
port = < As defined in RFC 3986>
Unsupported = "Unsupported" HCOLON feature-tag-list
User-Agent = "User-Agent" HCOLON ( product / comment )
*(LWS (product / comment))
Via = "Via" HCOLON via-parm *(COMMA via-parm)
via-parm = sent-protocol LWS sent-by *( SEMI via-params )
via-params = via-ttl / via-maddr
/ via-received / via-extension
via-ttl = "ttl" EQUAL ttl
via-maddr = "maddr" EQUAL host
via-received = "received" EQUAL (IPv4address / IPv6address)
IPv4address = < As defined in RFC 3986>
IPv6address = < As defined in RFC 3986>
via-extension = generic-param
sent-protocol = protocol-name SLASH protocol-version
SLASH transport-prot
protocol-name = "RTSP" / token
protocol-version = token
transport-prot = "UDP" / "TCP" / "TLS" / other-transport
other-transport = token
sent-by = host [ COLON port ]
WWW-Authenticate = "WWW-Authenticate" HCOLON challenge-list
20.3. SDP Extension Syntax
This section defines in ABNF the SDP extensions defined for RTSP.
See Appendix D for the definition of the extensions in text.
control-attribute = "a=control:" *SP RTSP-REQ-Ref CRLF
a-range-def = "a=range:" ranges-spec CRLF
a-mtag-def = "a=mtag:" message-tag CRLF
21. Security Considerations
The security considerations and threats around RTSP and its usage can
be divided into considerations around the signaling protocol itself
and the issues related to the media-stream delivery. However, when
it comes to mitigation of security threats, a threat depending on the
media-stream delivery may in fact be mitigated by a mechanism in the
signaling protocol.
There are several chapters and an appendix in this document that
define security solutions for the protocol. These sections will be
referenced when discussing the threats below. However, the reader
should take special notice of the Security Framework (Section 19) and
the specification of how to use SRTP and its key-management
(Appendix C.1.4) to achieve certain aspects of the media security.
21.1. Signaling Protocol Threats
This section focuses on issues related to the signaling protocol.
Because of the similarity in syntax and usage between RTSP servers
and HTTP servers, the security considerations outlined in [RFC7230],
[RFC7231], [RFC7232], [RFC7233], [RFC7234], and [RFC7235] apply as
well.
Specifically, please note the following:
Abuse of Server Log Information: A server is in the position to save
personal data about a user's requests that might identify their
media consumption patterns or subjects of interest. This
information is clearly confidential in nature, and its handling
can be constrained by law in certain countries. Log
information needs to be securely stored and appropriate
guidelines followed for its analysis. See Section 9.8 of
[RFC7230] for additional guidelines.
Transfer of Sensitive Information: There is no reason to believe
that information transferred in RTSP message, such as the URI
and the content of headers, especially the Server, Via,
Referrer, and From headers, may be any less sensitive than when
used in HTTP. Therefore, all of the precautions regarding the
protection of data privacy and user privacy apply to
implementers of RTSP clients, servers, and proxies. See
Sections 9.3-9.6 of [RFC7231] for further details.
The RTSP methods defined in this document are primarily used to
establish and control the delivery of the media data
represented by the URI; thus, the RTSP message bodies are
generally less sensitive than the ones in HTTP. Where HTTP
bodies could contain, for example, your medical records, in
RTSP, the sensitive video of your medical operation would be in
the media stream over the media-transport protocol, not in the
RTSP message. Still, one has to take note of what potential
sensitive information is included in RTSP. The protection of
the media data is separate, can be applied directly between
client and server, and is dependent on the media-transport
protocol in use. See Section 21.2 for further discussion.
This possibility for separation of security between media-
resource content and the signaling protocol mitigates the risk
of exposing the media content when using hop-by-hop security
for RTSP signaling using proxies (Section 19.3).
Attacks Based On File and Path Names: Though RTSP URIs are opaque
handles that do not necessarily have file-system semantics, it
is anticipated that many implementations will translate
portions of the Request-URIs directly to file-system calls. In
such cases, file systems SHOULD follow the precautions outlined
in Section 9.1 of [RFC7231], such as checking for ".." in path
components.
Personal Information: RTSP clients are often privy to the same
information that HTTP clients are (username, location, etc.)
and thus should be equally sensitive. See Section 9.8 of
[RFC7230], Sections 9.3-9.7 of [RFC7231], and Section 8 of
[RFC7234] for further recommendations.
Privacy Issues Connected to Accept Headers: Since similar usages of
the "Accept" headers exist in RTSP as in HTTP, the same caveats
outlined in Section 9.4 of [RFC7231] with regard to their use
should be followed.
Establishing Authority: RTSP shares with HTTP the question of how a
client communicates with the authoritative source for media
streams (Section 9.1 of [RFC7230]). The used DNS servers, the
security of the communication, and any possibility of a man in
the middle, and the trust in any RTSP proxies all affect the
possibility that a client has received a non-authoritative
response to a request. Ensuring that a client receives an
authoritative response is challenging, although using the
secure communication for RTSP signaling (rtsps) simplifies it
significantly as the server can provide a hostname identity
assertion in the TLS handshake.
Location Headers and Spoofing: If a single server supports multiple
organizations that do not trust each another, then it MUST
check the values of the Content-Location header fields in
responses that are generated under control of said
organizations to make sure that they do not attempt to
invalidate resources over which they have no authority (see
Section 15.4 of [RFC2616]).
In addition to the recommendations in the current HTTP specifications
([RFC7230], [RFC7231], [RFC7232], [RFC7233], [RFC7234], and [RFC7235]
as of this writing) and also those of the previous relevant RFCs
[RFC2068] [RFC2616], future HTTP specifications may provide
additional guidance on security issues.
The following are added considerations for RTSP implementations.
Session Hijacking: Since there is no or little relation between a
transport-layer connection and an RTSP session, it is possible
for a malicious client to issue requests with random session
identifiers that could affect other clients of an unsuspecting
server. To mitigate this, the server SHALL use a large, random
and non-sequential session identifier to minimize the
possibility of this kind of attack. However, unless the RTSP
signaling is always confidentiality protected, e.g., using TLS,
an on-path attacker will be able to hijack a session. Another
choice for preventing session hijacking is to use client
authentication and only allow the authenticated client creating
the session to access that session.
Authentication: Servers SHOULD implement both basic and Digest
[RFC2617] authentication. In environments requiring tighter
security for the control messages, the transport-layer
mechanism TLS [RFC5246] SHOULD be used.
Suspicious Behavior: Upon detecting instances of behavior that is
deemed a security risk, RTSP servers SHOULD return error code
403 (Forbidden). RTSP servers SHOULD also be aware of attempts
to probe the server for weaknesses and entry points and MAY
arbitrarily disconnect and ignore further requests from clients
that are deemed to be in violation of local security policy.
TLS through Proxies: If one uses the possibility to connect TLS in
multiple legs (Section 19.3), one really needs to be aware of
the trust model. This procedure requires trust in all proxies
part of the path to the server. The proxies one connects
through are identified, assuming the proxies so far connected
through are well behaved and fulfilling the trust. The
accepted proxies are men in the middle and have access to all
that goes on over the TLS connection. Thus, it is important to
consider if that trust model is acceptable in the actual
application. Further discussion of the actual trust model is
in Section 19.3. It is important to note what difference in
security properties, if any, may exist with the used media-
transport protocol and its security mechanism. Using SRTP and
the MIKEY-based key-establishment defined in Appendix C.1.4.1
enables media key-establishment to be done end-to-end without
revealing the keys to the proxies.
Resource Exhaustion: As RTSP is a stateful protocol and establishes
resource usage on the server, there is a clear possibility to
attack the server by trying to overbook these resources to
perform a DoS attack. This attack can be both against ongoing
sessions and to prevent others from establishing sessions.
RTSP agents will need to have mechanisms to prevent single
peers from consuming extensive amounts of resources. The
methods for guarding against this are varied and depend on the
agent's role and capabilities and policies. Each
implementation has to carefully consider its methods and
policies to mitigate this threat. There are recommendations
regarding the handling of connections in Section 10.7.
The above threats and considerations have resulted in a set of
security functions and mechanisms built into or used by the protocol.
The signaling protocol relies on two security features defined in the
Security Framework (Section 19): namely client authentication using
HTTP authentication and TLS-based transport protection of the
signaling messages. Both of these mechanisms are required to be
implemented by any RTSP agent.
A number of different security mitigations have been designed into
the protocol and will be instantiated if the specification is
implemented as written, for example, by ensuring sufficient amounts
of entropy in the randomly generated session identifiers when not
using client authentication to minimize the risk of session
hijacking. When client authentication is used, protection against
hijacking will be greatly improved by scoping the accessible sessions
to the one this client identity has created. Some of the above
threats are such that the implementation of the RTSP functionality
itself needs to consider which policy and strategy it uses to
mitigate them.
21.2. Media Stream Delivery Threats
The fact that RTSP establishes and controls a media-stream delivery
results in a set of security issues related to the media streams.
This section will attempt to analyze general threats; however, the
choice of media-stream transport protocol, such as RTP, will result
in some differences in threats and what mechanisms exist to mitigate
them. Thus, it becomes important that each specification of a new
media-stream transport and delivery protocol usable by RTSP requires
its own security analysis. This section includes one for RTP.
The set of general threats from or by the media-stream delivery
itself are:
Concentrated Denial-of-Service Attack: The protocol offers the
opportunity for a remote-controlled DoS attack, where the media
stream is the hammer in that DoS attack. See Section 21.2.1.
Media Confidentiality: The media delivery may contain content of any
type, and it is not possible, in general, to determine how
sensitive this content is from a confidentiality point. Thus, it
is a strong requirement that any media delivery protocol supply a
method for providing confidentiality of the actual media content.
In addition to the media-level confidentiality, it becomes
critical that no resource identifiers used in the signaling be
exposed to an attacker as they may have human-understandable names
or may be available to the attacker, allowing it to determine the
content the user received. Thus, the signaling protocol must also
provide confidentiality protection of any information related to
the media resource.
Media Integrity and Authentication: There are several reasons why an
attacker will be interested in substituting the media stream sent
out from the RTSP server with one of the attacker's creation or
selection, such as discrediting the target and misinformation
about the target. Therefore, it is important that the media
protocol provide mechanisms to verify the source authentication
and integrity and to prevent replay attacks on the media stream.
Scope of Multicast: If RTSP is used to control the transmission of
media onto a multicast network, the scope of the delivery must be
considered. RTSP supports the TTL Transport header parameter to
indicate this scope for IPv4. IPv6 has a different mechanism for
the scope boundary. However, such scope control has risks, as it
may be set too large and distribute media beyond the intended
scope.
Below (Section 21.2.2) a protocol-specific analysis of security
considerations for RTP-based media transport is included. In that
section, the requirements on implementing security functions for RTSP
agents supporting media delivery over RTP are made clear.
21.2.1. Remote DoS Attack
The attacker may initiate traffic flows to one or more IP addresses
by specifying them as the destination in SETUP requests. While the
attacker's IP address may be known in this case, this is not always
useful in the prevention of more attacks or ascertaining the
attacker's identity. Thus, an RTSP server MUST only allow client-
specified destinations for RTSP-initiated traffic flows if the server
has ensured that the specified destination address accepts receiving
media through different security mechanisms. Security mechanisms
that are acceptable in order of increasing generality are:
o Verification of the client's identity against a database of known
users using RTSP authentication mechanisms (preferably Digest
authentication or stronger)
o A list of addresses that have consented to be media destinations,
especially considering user identity
o Verification based on media path
The server SHOULD NOT allow the destination field to be set unless a
mechanism exists in the system to authorize the request originator to
direct streams to the recipient. It is preferred that this
authorization be performed by the media recipient (destination)
itself and the credentials be passed along to the server. However,
in certain cases, such as when the recipient address is a multicast
group or when the recipient is unable to communicate with the server
in an out-of-band manner, this may not be possible. In these cases,
the server may choose another method such as a server-resident
authorization list to ensure that the request originator has the
proper credentials to request stream delivery to the recipient.
One solution that performs the necessary verification of acceptance
of media suitable for unicast-based delivery is the NAT traversal
method based on Interactive Connectivity Establishment (ICE)
[RFC5245] described in [RFC7825]. This mechanism uses random
passwords and a username so that the probability of unintended
indication as a valid media destination is very low. In addition,
the server includes in its Session Traversal Utilities for NAT (STUN)
[RFC5389] requests a cookie (consisting of random material) that the
destination echoes back; thus, the solution also safeguards against
having an off-path attacker being able to spoof the STUN checks.
This leaves this solution vulnerable only to on-path attackers that
can see the STUN requests go to the target of attack and thus forge a
response.
For delivery to multicast addresses, there is a need for another
solution that is not specified in this memo.
21.2.2. RTP Security Analysis
RTP is a commonly used media-transport protocol and has been the most
common choice for RTSP 1.0 implementations. The core RTP protocol
has been in use for a long time, and it has well-known security
properties and the RTP security consideration (Section 9 of
[RFC3550]) needs to be reviewed. In perspective of the usage of RTP
in the context of RTSP, the following properties should be noted:
Stream Additions: RTP has support for multiple simultaneous media
streams in each RTP session. As some use cases require support
for non-synchronized adding and removal of media streams and their
identifiers, an attacker can easily insert additional media
streams into a session context that, according to protocol design,
is intended to be played out. Another threat vector is one of DoS
by exhausting the resources of the RTP session receiver, for
example, by using a large number of SSRC identifiers
simultaneously. The strong mitigation of this is to ensure that
one cryptographically authenticates any incoming packet flow to
the RTP session. Weak mitigations like blocking additional media
streams in session contexts easily lead to a DoS vulnerability in
addition to preventing certain RTP extensions or use cases that
rely on multiple media streams, such as RTP retransmission
[RFC4588] to function.
Forged Feedback: The built-in RTCP also offers a large attack
surface for a couple of different types of attacks. One venue is
to send RTCP feedback to the media sender indicating large amounts
of packet loss and thus trigger a media bitrate adaptation
response from the sender resulting in lowered media quality and
potentially a shutdown of the media stream. Another attack is to
perform a resource-exhaustion attack on the receiver by using many
SSRC identifiers to create large state tables and increase the
RTCP-related processing demands.
RTP/RTCP Extensions: RTP and RTCP extensions generally provide
additional and sometimes extremely powerful tools for DoS attacks
or service disruption. For example, the Code Control Message
[RFC5104] RTCP extensions enables both the lock down of the
bitrate to low values and disruption of video quality by
requesting intra-frames.
Taking into account the above general discussion in Section 21.2 and
the RTP-specific discussion in this section, it is clear that it is
necessary that a strong security mechanism be supported to protect
RTP. Therefore, this specification has the following requirements on
RTP security functions for all RTSP agents that handle media streams
and where media-stream transport is completed using RTP.
RTSP agents supporting RTP MUST implement Secure RTP (SRTP) [RFC3711]
and, thus, SAVP. In addition, SAVPF [RFC5124] MUST also be supported
if AVPF is implemented. This specification requires no additional
cryptographic transforms or configuration values beyond those
specified as mandatory to implement in RFC 3711, i.e., AES-CM and
HMAC-SHA1. The default key-management mechanism that MUST be
implemented is the one defined in MIKEY Key Establishment
(Appendix C.1.4.1). The MIKEY implementation MUST implement the
necessary functions for MIKEY-RSA-R mode [RFC4738] and the SRTP
parameter negotiation necessary to negotiate the supported SRTP
transforms and parameters.
22. IANA Considerations
This section describes a number of registries for RTSP 2.0 that have
been established and are maintained by IANA. These registries are
separate from any registries existing for RTSP 1.0. For each
registry, there is a description of the required content, the
registration procedures, and the entries that this document
registers. For more information on extending RTSP, see Section 2.7.
In addition, this document registers three SDP attributes.
Registries or entries in registries that have been made for RTSP 1.0
are not moved to RTSP 2.0: the registries and entries of RTSP 1.0 and
RTSP 2.0 are independent. If any registry or entry in a registry is
also required in RTSP 2.0, it MUST follow the procedure defined below
to allocate the registry or entry in a registry.
The sections describing how to register an item use some of the
registration policies described in [RFC5226] -- namely, "First Come
First Served", "Expert Review", "Specification Required", and
"Standards Action".
In case a registry requires a contact person, the authors (with
Magnus Westerlund <magnus.westerlund@ericsson.com> as primary) are
the contact persons for any entries created by this document.
IANA will request the following information for any registration
request:
o A name of the item to register according to the rules specified by
the intended registry
o Indication of who has change control over the feature (for
example, the IETF, ISO, ITU-T, other international standardization
bodies, a consortium, a particular company or group of companies,
or an individual)
o A reference to a further description, if available, for example
(in decreasing order of preference), an RFC, a published standard,
a published paper, a patent filing, a technical report, documented
source code or a computer manual
o For proprietary features, contact information (postal and email
address)
22.1. Feature Tags
22.1.1. Description
When a client and server try to determine what part and functionality
of the RTSP specification and any future extensions that its
counterpart implements, there is need for a namespace. This registry
contains named entries representing certain functionality.
The usage of feature tags is explained in Section 11 and
Section 13.1.
22.1.2. Registering New Feature Tags with IANA
The registering of feature tags is done on a First Come, First Served
[RFC5226] basis.
The registry entry for a feature tag has the following information:
o The name of the feature tag
* If the registrant indicates that the feature is proprietary,
IANA should request a vendor "prefix" portion of the name. The
name will then be the vendor prefix followed by a "." followed
by the rest of the provided feature name.
* If the feature is not proprietary, then IANA need not collect a
prefix for the name.
o A one-paragraph description of what the feature tag represents
o The applicability (server, client, proxy, or some combination)
o A reference to a specification, if applicable
Feature tag names (including the vendor prefix) may contain any non-
space and non-control characters. There is no length limit on
feature tags.
Examples for a vendor tag describing a proprietary feature are:
vendorA.specfeat01
vendorA.specfeat02
22.1.3. Registered Entries
The following feature tags are defined in this specification and
hereby registered. The change control belongs to the IETF.
play.basic: The implementation for delivery and playback operations
according to the core RTSP specification, as defined in this
memo. Applies for clients, servers, and proxies. See
Section 11.1.
play.scale: Support of scale operations for media playback. Applies
only for servers. See Section 18.46.
play.speed: Support of the speed functionality for media delivery.
Applies only for servers. See Section 18.50.
setup.rtp.rtcp.mux: Support of the RTP and RTCP multiplexing as
discussed in Appendix C.1.6.4. Applies for both client and
servers and any media caching proxy.
The IANA registry is a table with the name, description, and
reference for each feature tag.
22.2. RTSP Methods
22.2.1. Description
Methods are described in Section 13. Extending the protocol with new
methods allows for totally new functionality.
22.2.2. Registering New Methods with IANA
A new method is registered through a Standards Action [RFC5226]
because new methods may radically change the protocol's behavior and
purpose.
A specification for a new RTSP method consists of the following
items:
o A method name that follows the ABNF rules for methods.
o A clear specification of what a request using the method does and
what responses are expected. In which directions the method is
used: C->S, S->C, or both. How the use of headers, if any,
modifies the behavior and effect of the method.
o A list or table specifying which of the IANA-registered headers
that are allowed to be used with the method in the request or/and
response. The list or table SHOULD follow the format of tables in
Section 18.
o Describe how the method relates to network proxies.
22.2.3. Registered Entries
This specification, RFC 7826, registers 10 methods: DESCRIBE,
GET_PARAMETER, OPTIONS, PAUSE, PLAY, PLAY_NOTIFY, REDIRECT, SETUP,
SET_PARAMETER, and TEARDOWN. The initial table of the registry is
provided below.
Method Directionality Reference
-----------------------------------------------------
DESCRIBE C->S RFC 7826
GET_PARAMETER C->S, S->C RFC 7826
OPTIONS C->S, S->C RFC 7826
PAUSE C->S RFC 7826
PLAY C->S RFC 7826
PLAY_NOTIFY S->C RFC 7826
REDIRECT S->C RFC 7826
SETUP C->S RFC 7826
SET_PARAMETER C->S, S->C RFC 7826
TEARDOWN C->S, S->C RFC 7826
22.3. RTSP Status Codes
22.3.1. Description
A status code is the three-digit number used to convey information in
RTSP response messages; see Section 8. The number space is limited,
and care should be taken not to fill the space.
22.3.2. Registering New Status Codes with IANA
A new status code registration follows the policy of IETF Review
[RFC5226]. New RTSP functionality requiring Status Codes should
first be registered in the range of x50-x99. Only when the range is
full should registrations be made in the x00-x49 range, unless it is
to adopt an HTTP extension to RTSP. This is done to enable any HTTP
extension to be adopted to RTSP without needing to renumber any
related status codes. A specification for a new status code must
include the following:
o The registered number.
o A description of what the status code means and the expected
behavior of the sender and receiver of the code.
22.3.3. Registered Entries
RFC 7826 (this document) registers the numbered status code defined
in the ABNF entry "Status-Code", except "extension-code" (that
defines the syntax allowed for future extensions) in Section 20.2.2.
22.4. RTSP Headers
22.4.1. Description
By specifying new headers, one or more methods can be enhanced in
many different ways. An unknown header will be ignored by the
receiving agent. If the new header is vital for certain
functionality, a feature tag for the functionality can be created and
demanded to be used by the counterpart with the inclusion of a
Require header carrying the feature tag.
22.4.2. Registering New Headers with IANA
Registrations can be made following the Expert Review policy
[RFC5226]. A specification is recommended to be provided, preferably
an RFC or other specification from a Standards Developing
Organization. The minimal information in a registration request is
the header name and the contact information.
The expert reviewer verifies that the registration request contains
the following information:
o The name of the header.
o An ABNF specification of the header syntax.
o A list or table specifying when the header may be used,
encompassing all methods, their request or response, and the
direction (C->S or S->C).
o How the header is to be handled by proxies.
o A description of the purpose of the header.
22.4.3. Registered Entries
All headers specified in Section 18 in RFC 7826 have been registered.
The registry includes the header name and reference.
Furthermore, the following legacy RTSP headers defined in other
specifications are registered with header name, and reference
according to below list. Note: these references may not fulfill all
of the above rules for registrations due to their legacy status.
o x-wap-profile defined in [TS-26234]. The x-wap-profile request-
header contains one or more absolute URLs to the requesting
agent's device-capability profile.
o x-wap-profile-diff defined in [TS-26234]. The x-wap-profile-diff
request-header contains a subset of a device-capability profile.
o x-wap-profile-warning defined in [TS-26234]. The x-wap-profile-
warning is a response-header that contains error codes explaining
to what extent the server has been able to match the terminal
request in regard to device-capability profiles, as described
using x-wap-profile and x-wap-profile-diff headers.
o x-predecbufsize defined in [TS-26234]. This response-header
provides an RTSP agent with the TS 26.234 Annex G hypothetical
pre-decoder buffer size.
o x-initpredecbufperiod defined in [TS-26234]. This response-header
provides an RTSP agent with the TS 26.234 Annex G hypothetical
pre-decoder buffering period.
o x-initpostdecbufperiod defined in [TS-26234]. This response-
header provides an RTSP agent with the TS 26.234 Annex G post-
decoder buffering period.
o 3gpp-videopostdecbufsize defined in [TS-26234]. This response-
header provides an RTSP agent with the TS 26.234 defined post-
decoder buffer size usable for H.264 (AVC) video streams.
o 3GPP-Link-Char defined in [TS-26234]. This request-header
provides the RTSP server with the RTSP client's link
characteristics as determined from the radio interface. The
information that can be provided are guaranteed bitrate, maximum
bitrate and maximum transfer delay.
o 3GPP-Adaptation defined in [TS-26234]. This general-header is
part of the bitrate adaptation solution specified for the Packet-
switched Streaming Service (PSS). It provides the RTSP client's
buffer sizes and target buffer levels to the server, and responses
are used to acknowledge the support and values.
o 3GPP-QoE-Metrics defined in [TS-26234]. This general-header is
used by PSS RTSP agents to negotiate the quality of experience
metrics that a client should gather and report to the server.
o 3GPP-QoE-Feedback defined in [TS-26234]. This request-header is
used by RTSP clients supporting PSS to report the actual values of
the metrics gathered in its quality of experience metering.
The use of "x-" is NOT RECOMMENDED, but the above headers in the list
were defined prior to the clarification.
22.5. Accept-Credentials
The security framework's TLS connection mechanism has two
registerable entities.
22.5.1. Accept-Credentials Policies
This registry is for policies for an RTSP proxy's handling and
verification of TLS certificates when establishing an outbound TLS
connection on behalf of a client. In Section 19.3.1, three policies
for how to handle certificates are specified. Further policies may
be defined; registration is made through Standards Action [RFC5226].
A registration request is required to contain the following
information:
o Name of the policy.
o Text that describes how the policy works for handling the
certificates.
o A contact person.
This specification registers the following values:
Any: A policy requiring the proxy to accept any received
certificate.
Proxy: A policy where the proxy applies its own policies to
determine which certificates are accepted.
User: A policy where the certificate is required to be forwarded down
the proxy chain to the client, thus allowing the user to
decided to accept or refuse a certificate.
22.5.2. Accept-Credentials Hash Algorithms
The Accept-Credentials header (see Section 18.2) allows for the usage
of other algorithms for hashing the DER records of accepted entities.
The registration of any future algorithm is expected to be extremely
rare and could also cause interoperability problems. Therefore, the
bar for registering new algorithms is intentionally placed high.
Any registration of a new hash algorithm requires Standards Action
[RFC5226]. The registration needs to fulfill the following
requirement:
o The algorithms identifier meeting the "token" ABNF requirement.
o Provide a definition of the algorithm.
The registered value is:
Hash Alg. ID Reference
------------------------
sha-256 RFC 7826
22.6. Cache-Control Cache Directive Extensions
There exist a number of cache directives that can be sent in the
Cache-Control header. A registry for these cache directives has been
established by IANA. New registrations in this registry require
Standards Action or IESG Approval [RFC5226]. A registration request
needs to contain the following information.
o The name of the cache directive.
o A definition of the parameter value, if any is allowed.
o The specification if it is a request or response directive.
o Text that explains how the cache directive is used for RTSP-
controlled media streams.
o A contact person.
This specification registers the following values:
no-cache:
public:
private:
no-transform:
only-if-cached:
max-stale:
min-fresh:
must-revalidate:
proxy-revalidate:
max-age:
The registry contains the name of the directive and the reference.
22.7. Media Properties
22.7.1. Description
The media streams being controlled by RTSP can have many different
properties. The media properties required to cover the use cases
that were in mind when writing the specification are defined.
However, it can be expected that further innovation will result in
new use cases or media streams with properties not covered by the
ones specified here. Thus, new media properties can be specified.
As new media properties may need a substantial amount of new
definitions to correctly specify behavior for this property, the bar
is intended to be high.
22.7.2. Registration Rules
Registering a new media property is done following the Specification
Required policy [RFC5226]. The expert reviewer verifies that a
registration request fulfills the following requirements.
o An ABNF definition of the media property value name that meets
"media-prop-ext" definition is included.
o A definition of which media property group it belongs to or define
a new group is included.
o A description of all changes to the behavior of RTSP as result of
these changes is included.
o A contact person for the registration is listed.
22.7.3. Registered Values
This specification registers the ten values listed in Section 18.29.
The registry contains the property group, the name of the media
property, and the reference.
22.8. Notify-Reason Values
22.8.1. Description
Notify-Reason values are used to indicate the reason the notification
was sent. Each reason has its associated rules on what headers and
information may or must be included in the notification. New
notification behaviors need to be specified to enable interoperable
usage; thus, a specification of each new value is required.
22.8.2. Registration Rules
Registrations for new Notify-Reason values follow the Specification
Required policy [RFC5226]. The expert reviewer verifies that the
request fulfills the following requirements:
o An ABNF definition of the Notify-Reason value name that meets
"Notify-Reason-extension" definition is included.
o A description of which headers shall be included in the request
and response, when it should be sent, and any effect it has on the
server client state is made clear.
o A contact person for the registration is listed.
22.8.3. Registered Values
This specification registers three values defined in the Notify-Reas-
val ABNF, Section 20.2.3:
end-of-stream: This Notify-Reason value indicates the end of a media
stream.
media-properties-update: This Notify-Reason value allows the server
to indicate that the properties of the media have changed during
the playout.
scale-change: This Notify-Reason value allows the server to notify
the client about a change in the scale of the media.
The registry contains the name, description, and reference.
22.9. Range Header Formats
22.9.1. Description
The Range header (Section 18.40) allows for different range formats.
These range formats also need an identifier to be used in the Accept-
Ranges header (Section 18.5). New range formats may be registered,
but moderation should be applied as it makes interoperability more
difficult.
22.9.2. Registration Rules
A registration follows the Specification Required policy [RFC5226].
The expert reviewer verifies that the request fulfills the following
requirements:
o An ABNF definition of the range format that fulfills the "range-
ext" definition is included.
o The range format identifier used in Accept-Ranges header according
to the "extension-format" definition is defined.
o Rules for how one handles the range when using a negative Scale
are included.
o A contact person for the registration is listed.
22.9.3. Registered Values
The registry contains the Range header format identifier, the name of
the range format, and the reference. This specification registers
the following values.
npt: Normal Play Time
clock: UTC Absolute Time format
smpte: SMPTE Timestamps
smpte-30-drop: SMPTE Timestamps 29.97 Frames/sec (30 Hz with Drop)
smpte-25: SMPTE Timestamps 25 Frames/sec
22.10. Terminate-Reason Header
The Terminate-Reason header (Section 18.52) has two registries for
extensions.
22.10.1. Redirect Reasons
This registry contains reasons for session termination that can be
included in a Terminate-Reason header (Section 18.52). Registrations
follow the Expert Review policy [RFC5226]. The expert reviewer
verifies that the registration request contains the following
information:
o That the value follows the Terminate-Reason ABNF, i.e., be a
token.
o That the specification provide a definition of what procedures are
to be followed when a client receives this redirect reason.
o A contact person
This specification registers three values:
o Session-Timeout
o Server-Admin
o Internal-Error
The registry contains the name of the Redirect Reason and the
reference.
22.10.2. Terminate-Reason Header Parameters
This registry contains parameters that may be included in the
Terminate-Reason header (Section 18.52) in addition to a reason.
Registrations are made under the Specification Required policy
[RFC5226]. The expert reviewer verifies that the registration
request contains the following:
o A parameter name.
o A parameter following the syntax allowed by the RTSP 2.0
specification.
o A reference to the specification.
o A contact person.
This specification registers:
o time
o user-msg
The registry contains the name of the Terminate Reason and the
reference.
22.11. RTP-Info Header Parameters
22.11.1. Description
The RTP-Info header (Section 18.45) carries one or more parameter
value pairs with information about a particular point in the RTP
stream. RTP extensions or new usages may need new types of
information. As RTP information that could be needed is likely to be
generic enough, and to maximize the interoperability, new
registration is made under the Specification Required policy.
22.11.2. Registration Rules
Registrations for new RTP-Info values follow the policy of
Specification Required [RFC5226]. The expert reviewer verifies that
the registration request contains the following information.
o An ABNF definition that meets the "generic-param" definition.
o A reference to the specification.
o A contact person for the registration.
22.11.3. Registered Values
This specification registers the following parameter value pairs:
o url
o ssrc
o seq
o rtptime
The registry contains the name of the parameter and the reference.
22.12. Seek-Style Policies
22.12.1. Description
Seek-Style policy defines how the RTSP agent seeks in media content
when given a position within the media content. New seek policies
may be registered; however, a large number of these will complicate
implementation substantially. The impact of unknown policies is that
the server will not honor the unknown and will use the server default
policy instead.
22.12.2. Registration Rules
Registrations of new Seek-Style policies follow the Specification
Required policy [RFC5226]. The expert reviewer verifies that the
registration request fulfills the following requirements:
o Has an ABNF definition of the Seek-Style policy name that meets
"Seek-S-value-ext" definition.
o Includes a short description.
o Lists a contact person for the registration.
o Includes a description of which headers shall be included in the
request and response, when it should be sent, and any affect it
has on the server-client state.
22.12.3. Registered Values
This specification registers four values (Name - Short Description):
o RAP - Using the closest Random Access Point prior to or at the
requested start position.
o CoRAP - Conditional Random Access Point is like RAP, but only if
the RAP is closer prior to the requested start position than
current pause point.
o First-Prior - The first-prior policy will start delivery with the
media unit that has a playout time first prior to the requested
start position.
o Next - The next media units after the provided start position.
The registry contains the name of the Seek-Style policy, the
description, and the reference.
22.13. Transport Header Registries
The transport header (Section 18.54) contains a number of parameters
that have possibilities for future extensions. Therefore, registries
for these are defined below.
22.13.1. Transport Protocol Identifier
A Transport Protocol specification consists of a transport protocol
identifier, representing some combination of transport protocols, and
any number of transport header parameters required or optional to use
with the identified protocol specification. This registry contains
the identifiers used by registered transport protocol identifiers.
A registration for the parameter transport protocol identifier
follows the Specification Required policy [RFC5226]. The expert
reviewer verifies that the registration request fulfills the
following requirements:
o A contact person or organization with address and email.
o A value definition that follows the ABNF syntax definition of
"transport-id" Section 20.2.3.
o A descriptive text that explains how the registered values are
used in RTSP, which underlying transport protocols are used, and
any required Transport header parameters.
The registry contains the protocol ID string and the reference.
This specification registers the following values:
RTP/AVP: Use of the RTP [RFC3550] protocol for media transport in
combination with the "RTP Profile for Audio and Video
Conferences with Minimal Control" [RFC3551] over UDP. The
usage is explained in RFC 7826, Appendix C.1.
RTP/AVP/UDP: the same as RTP/AVP.
RTP/AVPF: Use of the RTP [RFC3550] protocol for media transport in
combination with the "Extended RTP Profile for RTCP-based
Feedback (RTP/AVPF)" [RFC4585] over UDP. The usage is
explained in RFC 7826, Appendix C.1.
RTP/AVPF/UDP: the same as RTP/AVPF.
RTP/SAVP: Use of the RTP [RFC3550] protocol for media transport in
combination with the "The Secure Real-time Transport Protocol
(SRTP)" [RFC3711] over UDP. The usage is explained in RFC
7826, Appendix C.1.
RTP/SAVP/UDP: the same as RTP/SAVP.
RTP/SAVPF: Use of the RTP [RFC3550] protocol for media transport in
combination with the "Extended Secure RTP Profile for Real-time
Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)"
[RFC5124] over UDP. The usage is explained in RFC 7826,
Appendix C.1.
RTP/SAVPF/UDP: the same as RTP/SAVPF.
RTP/AVP/TCP: Use of the RTP [RFC3550] protocol for media transport
in combination with the "RTP profile for audio and video
conferences with minimal control" [RFC3551] over TCP. The
usage is explained in RFC 7826, Appendix C.2.2.
RTP/AVPF/TCP: Use of the RTP [RFC3550] protocol for media transport
in combination with the "Extended RTP Profile for Real-time
Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)"
[RFC4585] over TCP. The usage is explained in RFC 7826,
Appendix C.2.2.
RTP/SAVP/TCP: Use of the RTP [RFC3550] protocol for media transport
in combination with the "The Secure Real-time Transport
Protocol (SRTP)" [RFC3711] over TCP. The usage is explained in
RFC 7826, Appendix C.2.2.
RTP/SAVPF/TCP: Use of the RTP [RFC3550] protocol for media transport
in combination with the "Extended Secure RTP Profile for Real-
time Transport Control Protocol (RTCP)-Based Feedback (RTP/
SAVPF)" [RFC5124] over TCP. The usage is explained in RFC
7826, Appendix C.2.2.
22.13.2. Transport Modes
The Transport Mode is a Transport header (Section 18.54) parameter.
It is used to identify the general mode of media transport. The PLAY
value registered defines a PLAYBACK mode, where media flows from
server to client.
A registration for the transport parameter mode follows the Standards
Action policy [RFC5226]. The registration request needs to meet the
following requirements:
o A value definition that follows the ABNF "token" definition
Section 20.2.3.
o Text that explains how the registered value is used in RTSP.
This specification registers one value:
PLAY: See RFC 7826.
The registry contains the transport mode value and the reference.
22.13.3. Transport Parameters
Transport Parameters are different parameters used in a Transport
header's transport specification (Section 18.54) to provide
additional information required beyond the transport protocol
identifier to establish a functioning transport.
A registration for parameters that may be included in the Transport
header follows the Specification Required policy [RFC5226]. The
expert reviewer verifies that the registration request fulfills the
following requirements:
o A Transport Parameter Name following the "token" ABNF definition.
o A value definition, if the parameter takes a value, that follows
the ABNF definition of "trn-par-value" Section 20.2.3.
o Text that explains how the registered value is used in RTSP.
This specification registers all the transport parameters defined in
Section 18.54. This is a copy of that list:
o unicast
o multicast
o interleaved
o ttl
o layers
o ssrc
o mode
o dest_addr
o src_addr
o setup
o connection
o RTCP-mux
o MIKEY
The registry contains the transport parameter name and the reference.
22.14. URI Schemes
This specification updates two URI schemes: one previously
registered, "rtsp", and one missing in the registry, "rtspu"
(previously only defined in RTSP 1.0 [RFC2326]). One new URI scheme,
"rtsps", is also registered. These URI schemes are registered in an
existing registry ("Uniform Resource Identifier (URI) Schemes") not
created by this memo. Registrations follow [RFC7595].
22.14.1. The "rtsp" URI Scheme
URI scheme name: rtsp
Status: Permanent
URI scheme syntax: See Section 20.2.1 of RFC 7826.
URI scheme semantics: The rtsp scheme is used to indicate resources
accessible through the usage of the Real-Time Streaming
Protocol (RTSP). RTSP allows different operations on the
resource identified by the URI, but the primary purpose is the
streaming delivery of the resource to a client. However, the
operations that are currently defined are DESCRIBE,
GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE, REDIRECT,
SETUP, SET_PARAMETER, and TEARDOWN.
Encoding considerations: IRIs in this scheme are defined and need to
be encoded as RTSP URIs when used within RTSP. That encoding
is done according to RFC 3987.
Applications/protocols that use this URI scheme name: RTSP 1.0 (RFC
2326), RTSP 2.0 (RFC 7826).
Interoperability considerations: The extensions in the URI syntax
performed between RTSP 1.0 and 2.0 can create interoperability
issues. The changes are:
Support for IPv6 literals in the host part and future IP
literals through a mechanism as defined in RFC 3986.
A new relative format to use in RTSP elements that is not
required to start with "/".
The above changes should have no impact on interoperability as
discussed in detail in Section 4.2 of RFC 7826.
Security considerations: All the security threats identified in
Section 7 of RFC 3986 also apply to this scheme. They need to
be reviewed and considered in any implementation utilizing this
scheme.
Contact: Magnus Westerlund, magnus.westerlund@ericsson.com
Author/Change controller: IETF
References: RFC 2326, RFC 3986, RFC 3987, and RFC 7826
22.14.2. The "rtsps" URI Scheme
URI scheme name: rtsps
Status: Permanent
URI scheme syntax: See Section 20.2.1 of RFC 7826.
URI scheme semantics: The rtsps scheme is used to indicate resources
accessible through the usage of the Real-Time Streaming
Protocol (RTSP) over TLS. RTSP allows different operations on
the resource identified by the URI, but the primary purpose is
the streaming delivery of the resource to a client. However,
the operations that are currently defined are DESCRIBE,
GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE, REDIRECT,
SETUP, SET_PARAMETER, and TEARDOWN.
Encoding considerations: IRIs in this scheme are defined and need to
be encoded as RTSP URIs when used within RTSP. That encoding
is done according to RFC 3987.
Applications/protocols that use this URI scheme name: RTSP 1.0 (RFC
2326), RTSP 2.0 (RFC 7826).
Interoperability considerations: The "rtsps" scheme was never
officially defined for RTSP 1.0; however, it has seen
widespread use in actual deployments of RTSP 1.0. Therefore,
this section discusses the believed changes between the
unspecified RTSP 1.0 "rtsps" scheme and RTSP 2.0 definition.
The extensions in the URI syntax performed between RTSP 1.0 and
2.0 can create interoperability issues. The changes are:
Support for IPv6 literals in the host part and future IP
literals through a mechanism as defined by RFC 3986.
A new relative format to use in RTSP elements that is not
required to start with "/".
The above changes should have no impact on interoperability as
discussed in detail in Section 4.2 of RFC 7826.
Security considerations: All the security threats identified in
Section 7 of RFC 3986 also apply to this scheme. They need to
be reviewed and considered in any implementation utilizing this
scheme.
Contact: Magnus Westerlund, magnus.westerlund@ericsson.com
Author/Change controller: IETF
References: RFC 2326, RFC 3986, RFC 3987, and RFC 7826
22.14.3. The "rtspu" URI Scheme
URI scheme name: rtspu
Status: Permanent
URI scheme syntax: See Section 3.2 of RFC 2326.
URI scheme semantics: The rtspu scheme is used to indicate resources
accessible through the usage of the Real-Time Streaming
Protocol (RTSP) over unreliable datagram transport. RTSP
allows different operations on the resource identified by the
URI, but the primary purpose is the streaming delivery of the
resource to a client. However, the operations that are
currently defined are DESCRIBE, GET_PARAMETER, OPTIONS,
REDIRECT,PLAY, PLAY_NOTIFY, PAUSE, SETUP, SET_PARAMETER, and
TEARDOWN.
Encoding considerations: This scheme is not intended to be used with
characters outside the US-ASCII repertoire.
Applications/protocols that use this URI scheme name: RTSP 1.0 (RFC
2326).
Interoperability considerations: The definition of the transport
mechanism of RTSP over UDP has interoperability issues. That
makes the usage of this scheme problematic.
Security considerations: All the security threats identified in
Section 7 of RFC 3986 also apply to this scheme. They need to
be reviewed and considered in any implementation utilizing this
scheme.
Contact: Magnus Westerlund, magnus.westerlund@ericsson.com
Author/Change controller: IETF
References: RFC 2326
22.15. SDP Attributes
This specification defines three SDP [RFC4566] attributes that have
been registered by IANA.
SDP Attribute ("att-field"):
Attribute name: range
Long form: Media Range Attribute
Type of name: att-field
Type of attribute: both session and media level
Subject to charset: No
Purpose: RFC 7826
Reference: RFC 2326, RFC 7826
Values: See ABNF definition.
Attribute name: control
Long form: RTSP control URI
Type of name: att-field
Type of attribute: both session and media level
Subject to charset: No
Purpose: RFC 7826
Reference: RFC 2326, RFC 7826
Values: Absolute or Relative URIs.
Attribute name: mtag
Long form: Message Tag
Type of name: att-field
Type of attribute: both session and media level
Subject to charset: No
Purpose: RFC 7826
Reference: RFC 7826
Values: See ABNF definition
22.16. Media Type Registration for text/parameters
Type name: text
Subtype name: parameters
Required parameters:
Optional parameters: charset: The charset parameter is applicable to
the encoding of the parameter values. The default charset is
UTF-8, if the 'charset' parameter is not present.
Encoding considerations: 8bit
Security considerations: This format may carry any type of
parameters. Some can have security requirements, like privacy,
confidentiality, or integrity requirements. The format has no
built-in security protection. For the usage, the transport can be
protected between server and client using TLS. However, care must
be taken to consider if the proxies are also trusted with the
parameters in case hop-by-hop security is used. If stored as a
file in a file system, the necessary precautions need to be taken
in relation to the parameter requirements including object
security such as S/MIME [RFC5751].
Interoperability considerations: This media type was mentioned as a
fictional example in [RFC2326], but was not formally specified.
This has resulted in usage of this media type that may not match
its formal definition.
Published specification: RFC 7826, Appendix F.
Applications that use this media type: Applications that use RTSP
and have additional parameters they like to read and set using the
RTSP GET_PARAMETER and SET_PARAMETER methods.
Additional information:
Magic number(s): N/A
File extension(s): N/A
Macintosh file type code(s): N/A
Person & email address to contact for further information:
Magnus Westerlund (magnus.westerlund@ericsson.com)
Intended usage: Common
Restrictions on usage: None
Author: Magnus Westerlund (magnus.westerlund@ericsson.com)
Change controller: IETF
Addition Notes:
23. References
23.1. Normative References
[FIPS180-4]
National Institute of Standards and Technology (NIST),
"Federal Information Processing Standards Publication:
Secure Hash Standard (SHS)", DOI 10.6028/NIST.FIPS.180-4,
August 2015, <http://nvlpubs.nist.gov/nistpubs/FIPS/
NIST.FIPS.180-4.pdf>.
[RFC768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
DOI 10.17487/RFC0768, August 1980,
<http://www.rfc-editor.org/info/rfc768>.
[RFC793] Postel, J., "Transmission Control Protocol", STD 7,
RFC 793, DOI 10.17487/RFC0793, September 1981,
<http://www.rfc-editor.org/info/rfc793>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.
[RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6
(IPv6) Specification", RFC 2460, DOI 10.17487/RFC2460,
December 1998, <http://www.rfc-editor.org/info/rfc2460>.
[RFC2616] Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext
Transfer Protocol -- HTTP/1.1", RFC 2616,
DOI 10.17487/RFC2616, June 1999,
<http://www.rfc-editor.org/info/rfc2616>.
[RFC2617] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
Leach, P., Luotonen, A., and L. Stewart, "HTTP
Authentication: Basic and Digest Access Authentication",
RFC 2617, DOI 10.17487/RFC2617, June 1999,
<http://www.rfc-editor.org/info/rfc2617>.
[RFC2818] Rescorla, E., "HTTP Over TLS", RFC 2818,
DOI 10.17487/RFC2818, May 2000,
<http://www.rfc-editor.org/info/rfc2818>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003,
<http://www.rfc-editor.org/info/rfc3551>.
[RFC3629] Yergeau, F., "UTF-8, a transformation format of ISO
10646", STD 63, RFC 3629, DOI 10.17487/RFC3629, November
2003, <http://www.rfc-editor.org/info/rfc3629>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004,
<http://www.rfc-editor.org/info/rfc3711>.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
DOI 10.17487/RFC3830, August 2004,
<http://www.rfc-editor.org/info/rfc3830>.
[RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Syntax", STD 66,
RFC 3986, DOI 10.17487/RFC3986, January 2005,
<http://www.rfc-editor.org/info/rfc3986>.
[RFC3987] Duerst, M. and M. Suignard, "Internationalized Resource
Identifiers (IRIs)", RFC 3987, DOI 10.17487/RFC3987,
January 2005, <http://www.rfc-editor.org/info/rfc3987>.
[RFC4086] Eastlake 3rd, D., Schiller, J., and S. Crocker,
"Randomness Requirements for Security", BCP 106, RFC 4086,
DOI 10.17487/RFC4086, June 2005,
<http://www.rfc-editor.org/info/rfc4086>.
[RFC4291] Hinden, R. and S. Deering, "IP Version 6 Addressing
Architecture", RFC 4291, DOI 10.17487/RFC4291, February
2006, <http://www.rfc-editor.org/info/rfc4291>.
[RFC7595] Thaler, D., Ed., Hansen, T., and T. Hardie, "Guidelines
and Registration Procedures for URI Schemes", BCP 35, RFC
7595, DOI 10.17487/RFC7595, June 2015, <http://www.rfc-
editor.org/info/rfc7595>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <http://www.rfc-editor.org/info/rfc4566>.
[RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
and RTP Control Protocol (RTCP) Packets over Connection-
Oriented Transport", RFC 4571, DOI 10.17487/RFC4571, July
2006, <http://www.rfc-editor.org/info/rfc4571>.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
DOI 10.17487/RFC4585, July 2006,
<http://www.rfc-editor.org/info/rfc4585>.
[RFC4648] Josefsson, S., "The Base16, Base32, and Base64 Data
Encodings", RFC 4648, DOI 10.17487/RFC4648, October 2006,
<http://www.rfc-editor.org/info/rfc4648>.
[RFC4738] Ignjatic, D., Dondeti, L., Audet, F., and P. Lin, "MIKEY-
RSA-R: An Additional Mode of Key Distribution in
Multimedia Internet KEYing (MIKEY)", RFC 4738,
DOI 10.17487/RFC4738, November 2006,
<http://www.rfc-editor.org/info/rfc4738>.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
2008, <http://www.rfc-editor.org/info/rfc5124>.
[RFC5226] Narten, T. and H. Alvestrand, "Guidelines for Writing an
IANA Considerations Section in RFCs", BCP 26, RFC 5226,
DOI 10.17487/RFC5226, May 2008,
<http://www.rfc-editor.org/info/rfc5226>.
[RFC5234] Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", STD 68, RFC 5234,
DOI 10.17487/RFC5234, January 2008,
<http://www.rfc-editor.org/info/rfc5234>.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246,
DOI 10.17487/RFC5246, August 2008,
<http://www.rfc-editor.org/info/rfc5246>.
[RFC5280] Cooper, D., Santesson, S., Farrell, S., Boeyen, S.,
Housley, R., and W. Polk, "Internet X.509 Public Key
Infrastructure Certificate and Certificate Revocation List
(CRL) Profile", RFC 5280, DOI 10.17487/RFC5280, May 2008,
<http://www.rfc-editor.org/info/rfc5280>.
[RFC5322] Resnick, P., Ed., "Internet Message Format", RFC 5322,
DOI 10.17487/RFC5322, October 2008,
<http://www.rfc-editor.org/info/rfc5322>.
[RFC5646] Phillips, A., Ed. and M. Davis, Ed., "Tags for Identifying
Languages", BCP 47, RFC 5646, DOI 10.17487/RFC5646,
September 2009, <http://www.rfc-editor.org/info/rfc5646>.
[RFC5751] Ramsdell, B. and S. Turner, "Secure/Multipurpose Internet
Mail Extensions (S/MIME) Version 3.2 Message
Specification", RFC 5751, DOI 10.17487/RFC5751, January
2010, <http://www.rfc-editor.org/info/rfc5751>.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010,
<http://www.rfc-editor.org/info/rfc5761>.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888,
DOI 10.17487/RFC5888, June 2010,
<http://www.rfc-editor.org/info/rfc5888>.
[RFC6838] Freed, N., Klensin, J., and T. Hansen, "Media Type
Specifications and Registration Procedures", BCP 13,
RFC 6838, DOI 10.17487/RFC6838, January 2013,
<http://www.rfc-editor.org/info/rfc6838>.
[RFC7230] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Message Syntax and Routing",
RFC 7230, DOI 10.17487/RFC7230, June 2014,
<http://www.rfc-editor.org/info/rfc7230>.
[RFC7231] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Semantics and Content", RFC 7231,
DOI 10.17487/RFC7231, June 2014,
<http://www.rfc-editor.org/info/rfc7231>.
[RFC7232] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Conditional Requests", RFC 7232,
DOI 10.17487/RFC7232, June 2014,
<http://www.rfc-editor.org/info/rfc7232>.
[RFC7233] Fielding, R., Ed., Lafon, Y., Ed., and J. Reschke, Ed.,
"Hypertext Transfer Protocol (HTTP/1.1): Range Requests",
RFC 7233, DOI 10.17487/RFC7233, June 2014,
<http://www.rfc-editor.org/info/rfc7233>.
[RFC7234] Fielding, R., Ed., Nottingham, M., Ed., and J. Reschke,
Ed., "Hypertext Transfer Protocol (HTTP/1.1): Caching",
RFC 7234, DOI 10.17487/RFC7234, June 2014,
<http://www.rfc-editor.org/info/rfc7234>.
[RFC7235] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
Protocol (HTTP/1.1): Authentication", RFC 7235,
DOI 10.17487/RFC7235, June 2014,
<http://www.rfc-editor.org/info/rfc7235>.
[RFC7615] Reschke, J., "HTTP Authentication-Info and Proxy-
Authentication-Info Response Header Fields", RFC 7615,
DOI 10.17487/RFC7615, September 2015,
<http://www.rfc-editor.org/info/rfc7615>.
[RFC7616] Shekh-Yusef, R., Ed., Ahrens, D., and S. Bremer, "HTTP
Digest Access Authentication", RFC 7616,
DOI 10.17487/RFC7616, September 2015,
<http://www.rfc-editor.org/info/rfc7616>.
[RFC7617] Reschke, J., "The 'Basic' HTTP Authentication Scheme",
RFC 7617, DOI 10.17487/RFC7617, September 2015,
<http://www.rfc-editor.org/info/rfc7617>.
[RFC7825] Goldberg, J., Westerlund, M., and T. Zeng, "A Network
Address Translator (NAT) Traversal Mechanism for Media
Controlled by Real-Time Streaming Protocol (RTSP)",
RFC 7825, DOI 10.17487/RFC7825, December 2016,
<http://www.rfc-editor.org/info/rfc7825>.
[RTP-CIRCUIT-BREAKERS]
Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", Work in
Progress, draft-ietf-avtcore-rtp-circuit-breakers-13,
February 2016.
[SMPTE-TC] Society of Motion Picture and Television Engineers, "ST
12-1:2008 For Television -- Time and Control Code",
DOI 10.5594/SMPTE.ST12-1.2008, February 2008,
<http://ieeexplore.ieee.org/servlet/
opac?punumber=7289818>.
[TS-26234] 3rd Generation Partnership Project (3GPP), "Transparent
end-to-end Packet-switched Streaming Service (PSS);
Protocols and codecs", Technical Specification 26.234,
Release 13, September 2015,
<http://www.3gpp.org/DynaReport/26234.htm>.
23.2. Informative References
[ISO.13818-6.1995]
International Organization for Standardization,
"Information technology -- Generic coding of moving
pictures and associated audio information - part 6:
Extension for DSM-CC", ISO Draft Standard 13818-6:1998,
October 1998,
<http://www.iso.org/iso/home/store/catalogue_tc/
catalogue_detail.htm?csnumber=25039>.
[ISO.8601.2000]
International Organization for Standardization, "Data
elements and interchange formats - Information interchange
- Representation of dates and times", ISO/IEC Standard
8601, December 2000.
[RFC791] Postel, J., "Internet Protocol", STD 5, RFC 791,
DOI 10.17487/RFC0791, September 1981,
<http://www.rfc-editor.org/info/rfc791>.
[RFC1123] Braden, R., Ed., "Requirements for Internet Hosts -
Application and Support", STD 3, RFC 1123,
DOI 10.17487/RFC1123, October 1989,
<http://www.rfc-editor.org/info/rfc1123>.
[RFC2068] Fielding, R., Gettys, J., Mogul, J., Frystyk, H., and T.
Berners-Lee, "Hypertext Transfer Protocol -- HTTP/1.1",
RFC 2068, DOI 10.17487/RFC2068, January 1997,
<http://www.rfc-editor.org/info/rfc2068>.
[RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
Streaming Protocol (RTSP)", RFC 2326,
DOI 10.17487/RFC2326, April 1998,
<http://www.rfc-editor.org/info/rfc2326>.
[RFC2663] Srisuresh, P. and M. Holdrege, "IP Network Address
Translator (NAT) Terminology and Considerations",
RFC 2663, DOI 10.17487/RFC2663, August 1999,
<http://www.rfc-editor.org/info/rfc2663>.
[RFC2974] Handley, M., Perkins, C., and E. Whelan, "Session
Announcement Protocol", RFC 2974, DOI 10.17487/RFC2974,
October 2000, <http://www.rfc-editor.org/info/rfc2974>.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
DOI 10.17487/RFC3261, June 2002,
<http://www.rfc-editor.org/info/rfc3261>.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
DOI 10.17487/RFC3264, June 2002,
<http://www.rfc-editor.org/info/rfc3264>.
[RFC3339] Klyne, G. and C. Newman, "Date and Time on the Internet:
Timestamps", RFC 3339, DOI 10.17487/RFC3339, July 2002,
<http://www.rfc-editor.org/info/rfc3339>.
[RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in
the Session Description Protocol (SDP)", RFC 4145,
DOI 10.17487/RFC4145, September 2005,
<http://www.rfc-editor.org/info/rfc4145>.
[RFC4567] Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E.
Carrara, "Key Management Extensions for Session
Description Protocol (SDP) and Real Time Streaming
Protocol (RTSP)", RFC 4567, DOI 10.17487/RFC4567, July
2006, <http://www.rfc-editor.org/info/rfc4567>.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
DOI 10.17487/RFC4588, July 2006,
<http://www.rfc-editor.org/info/rfc4588>.
[RFC4855] Casner, S., "Media Type Registration of RTP Payload
Formats", RFC 4855, DOI 10.17487/RFC4855, February 2007,
<http://www.rfc-editor.org/info/rfc4855>.
[RFC4856] Casner, S., "Media Type Registration of Payload Formats in
the RTP Profile for Audio and Video Conferences",
RFC 4856, DOI 10.17487/RFC4856, February 2007,
<http://www.rfc-editor.org/info/rfc4856>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <http://www.rfc-editor.org/info/rfc5104>.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
DOI 10.17487/RFC5245, April 2010,
<http://www.rfc-editor.org/info/rfc5245>.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
DOI 10.17487/RFC5389, October 2008,
<http://www.rfc-editor.org/info/rfc5389>.
[RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding
Dependency in the Session Description Protocol (SDP)",
RFC 5583, DOI 10.17487/RFC5583, July 2009,
<http://www.rfc-editor.org/info/rfc5583>.
[RFC5905] Mills, D., Martin, J., Ed., Burbank, J., and W. Kasch,
"Network Time Protocol Version 4: Protocol and Algorithms
Specification", RFC 5905, DOI 10.17487/RFC5905, June 2010,
<http://www.rfc-editor.org/info/rfc5905>.
[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298,
DOI 10.17487/RFC6298, June 2011,
<http://www.rfc-editor.org/info/rfc6298>.
[Stevens98]
Stevens, W., Fenner, B., and A. Rudoff, "Unix Networking
Programming, Volume 1: The Sockets Networking API (3rd
Edition)", 1998.
Appendix A. Examples
This section contains several different examples trying to illustrate
possible ways of using RTSP. The examples can also help with the
understanding of how functions of RTSP work. However, remember that
these are examples and the normative and syntax descriptions in the
other sections take precedence. Please also note that many of the
examples have been broken into several lines, where following lines
start with whitespace as allowed by the syntax.
A.1. Media on Demand (Unicast)
This is an example of media-on-demand streaming of media stored in a
container file. For the purposes of this example, a container file
is a storage entity in which multiple continuous media types
pertaining to the same end-user presentation are present. In effect,
the container file represents an RTSP presentation, with each of its
components being RTSP-controlled media streams. Container files are
a widely used means to store such presentations. While the
components are transported as independent streams, it is desirable to
maintain a common context for those streams at the server end.
This enables the server to keep a single storage handle open
easily. It also allows treating all the streams equally in case
of any prioritization of streams by the server.
It is also possible that the presentation author may wish to prevent
selective retrieval of the streams by the client in order to preserve
the artistic effect of the combined media presentation. Similarly,
in such a tightly bound presentation, it is desirable to be able to
control all the streams via a single control message using an
aggregate URI.
The following is an example of using a single RTSP session to control
multiple streams. It also illustrates the use of aggregate URIs. In
a container file, it is also desirable not to write any URI parts
that are not kept when the container is distributed, like the host
and most of the path element. Therefore, this example also uses the
"*" and relative URI in the delivered SDP.
Also, this presentation description (SDP) is not cacheable, as the
Expires header is set to an equal value with date indicating
immediate expiration of its validity.
Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URI to
the container file.
C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Server: PhonyServer/1.0
Date: Fri, 20 Dec 2013 10:20:32 +0000
Content-Type: application/sdp
Content-Length: 271
Content-Base: rtsp://example.com/twister.3gp/
Expires: Fri, 20 Dec 2013 12:20:32 +0000
v=0
o=- 2890844256 2890842807 IN IP4 198.51.100.5
s=RTSP Session
i=An Example of RTSP Session Usage
e=adm@example.com
c=IN IP4 0.0.0.0
a=control: *
a=range:npt=00:00:00-00:10:34.10
t=0 0
m=audio 0 RTP/AVP 0
a=control: trackID=1
m=video 0 RTP/AVP 26
a=control: trackID=4
C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"
Accept-Ranges: npt, smpte, clock
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast; ssrc=93CB001E;
dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";
src_addr="198.51.100.5:9000"/"198.51.100.5:9001"
Session: OccldOFFq23KwjYpAnBbUr
Expires: Fri, 20 Dec 2013 12:20:33 +0000
Date: Fri, 20 Dec 2013 10:20:33 +0000
Accept-Ranges: npt
Media-Properties: Random-Access=0.02, Immutable, Unlimited
C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
Session: OccldOFFq23KwjYpAnBbUr
Accept-Ranges: npt, smpte, clock
M->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast; ssrc=A813FC13;
dest_addr="192.0.2.53:8002"/"192.0.2.53:8003";
src_addr="198.51.100.5:9002"/"198.51.100.5:9003";
Session: OccldOFFq23KwjYpAnBbUr
Expires: Fri, 20 Dec 2013 12:20:33 +0000
Date: Fri, 20 Dec 2013 10:20:33 +0000
Accept-Range: NPT
Media-Properties: Random-Access=0.8, Immutable, Unlimited
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 4
User-Agent: PhonyClient/1.2
Range: npt=30-
Seek-Style: RAP
Session: OccldOFFq23KwjYpAnBbUr
M->C: RTSP/2.0 200 OK
CSeq: 4
Server: PhonyServer/1.0
Date: Fri, 20 Dec 2013 10:20:34 +0000
Session: OccldOFFq23KwjYpAnBbUr
Range: npt=30-634.10
Seek-Style: RAP
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12345;rtptime=3450012,
url="rtsp://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=54321;rtptime=2876889
C->M: PAUSE rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 5
User-Agent: PhonyClient/1.2
Session: OccldOFFq23KwjYpAnBbUr
# Pause happens 0.87 seconds after starting to play
M->C: RTSP/2.0 200 OK
CSeq: 5
Server: PhonyServer/1.0
Date: Fri, 20 Dec 2013 10:20:35 +0000
Session: OccldOFFq23KwjYpAnBbUr
Range: npt=30.87-634.10
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 6
User-Agent: PhonyClient/1.2
Range: npt=30.87-634.10
Seek-Style: Next
Session: OccldOFFq23KwjYpAnBbUr
M->C: RTSP/2.0 200 OK
CSeq: 6
Server: PhonyServer/1.0
Date: Fri, 20 Dec 2013 10:22:13 +0000
Session: OccldOFFq23KwjYpAnBbUr
Range: npt=30.87-634.10
Seek-Style: Next
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12555;rtptime=6330012,
url="rtsp://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=55021;rtptime=3132889
C->M: TEARDOWN rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 7
User-Agent: PhonyClient/1.2
Session: OccldOFFq23KwjYpAnBbUr
M->C: RTSP/2.0 200 OK
CSeq: 7
Server: PhonyServer/1.0
Date: Fri, 20 Dec 2013 10:31:53 +0000
A.2. Media on Demand Using Pipelining
This example is basically the example above (Appendix A.1), but now
utilizing pipelining to speed up the setup. It requires only two
round-trip times until the media starts flowing. First of all, the
session description is retrieved to determine what media resources
need to be set up. In the second step, one sends the necessary SETUP
requests and the PLAY request to initiate media delivery.
Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URI to
the container file.
C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Server: PhonyServer/1.0
Date: Fri, 20 Dec 2013 10:20:32 +0000
Content-Type: application/sdp
Content-Length: 271
Content-Base: rtsp://example.com/twister.3gp/
Expires: Fri, 20 Dec 2013 12:20:32 +0000
v=0
o=- 2890844256 2890842807 IN IP4 192.0.2.5
s=RTSP Session
i=An Example of RTSP Session Usage
e=adm@example.com
c=IN IP4 0.0.0.0
a=control: *
a=range:npt=00:00:00-00:10:34.10
t=0 0
m=audio 0 RTP/AVP 0
a=control: trackID=1
m=video 0 RTP/AVP 26
a=control: trackID=4
C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"
Accept-Ranges: npt, smpte, clock
Pipelined-Requests: 7654
C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
Accept-Ranges: npt, smpte, clock
Pipelined-Requests: 7654
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 4
User-Agent: PhonyClient/1.2
Range: npt=0-
Seek-Style: RAP
Pipelined-Requests: 7654
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast;
dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";
src_addr="198.51.100.5:9000"/"198.51.100.5:9001";
ssrc=93CB001E
Session: OccldOFFq23KwjYpAnBbUr
Expires: Fri, 20 Dec 2013 12:20:32 +0000
Date: Fri, 20 Dec 2013 10:20:32 +0000
Accept-Ranges: npt
Pipelined-Requests: 7654
Media-Properties: Random-Access=0.2, Immutable, Unlimited
M->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast;
dest_addr="192.0.2.53:8002"/"192.0.2.53:8003;
src_addr="198.51.100.5:9002"/"198.51.100.5:9003";
ssrc=A813FC13
Session: OccldOFFq23KwjYpAnBbUr
Expires: Sat, 21 Dec 2013 10:20:32 +0000
Date: Fri, 20 Dec 2013 10:20:32 +0000
Accept-Range: NPT
Pipelined-Requests: 7654
Media-Properties: Random-Access=0.8, Immutable, Unlimited
M->C: RTSP/2.0 200 OK
CSeq: 4
Server: PhonyServer/1.0
Date: Fri, 20 Dec 2013 10:20:32 +0000
Session: OccldOFFq23KwjYpAnBbUr
Range: npt=0-623.10
Seek-Style: RAP
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12345;rtptime=3450012,
url="rtsp://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=54321;rtptime=2876889
Pipelined-Requests: 7654
A.3. Secured Media Session for On-Demand Content
This example is basically the above example (Appendix A.2), but now
including establishment of SRTP crypto contexts to get a secured
media delivery. First of all, the client attempts to fetch this
insecurely, but the server redirects to a URI indicating a
requirement on using a secure connection for the RTSP messages. The
client establishes a TCP/TLS connection, and the session description
is retrieved to determine what media resources need to be set up. In
the this session description, secure media (SRTP) is indicated. In
the next step, the client sends the necessary SETUP requests
including MIKEY messages. This is pipelined with a PLAY request to
initiate media delivery.
Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URI to
the container file.
Note: The MIKEY messages below are not valid MIKEY messages and are
Base64-encoded random data to represent where the MIKEY messages
would go.
C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 301 Moved Permanently
CSeq: 1
Server: PhonyServer/1.0
Date: Fri, 20 Dec 2013 10:25:32 +0000
Location: rtsps://example.com/twister.3gp
C->M: Establish TCP/TLS connection and verify server's
certificate that represents example.com.
Used for all below RTSP messages.
C->M: DESCRIBE rtsps://example.com/twister.3gp RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Date: Fri, 20 Dec 2013 10:25:33 +0000
Content-Type: application/sdp
Content-Length: 271
Content-Base: rtsps://example.com/twister.3gp/
Expires: Fri, 20 Dec 2013 12:25:33 +0000
v=0
o=- 2890844256 2890842807 IN IP4 192.0.2.5
s=RTSP Session
i=An Example of RTSP Session Usage
e=adm@example.com
c=IN IP4 0.0.0.0
a=control: *
a=range:npt=00:00:00-00:10:34.10
t=0 0
m=audio 0 RTP/SAVP 0
a=control: trackID=1
m=video 0 RTP/SAVP 26
a=control: trackID=4
C->M: SETUP rtsps://example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/SAVP;unicast;dest_addr=":8000"/":8001";
MIKEY=VGhpcyBpcyB0aGUgZmlyc3Qgc3RyZWFtcyBNSUtFWSBtZXNzYWdl
Accept-Ranges: npt, smpte, clock
Pipelined-Requests: 7654
C->M: SETUP rtsps://example.com/twister.3gp/trackID=4 RTSP/2.0
CSeq: 4
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/SAVP;unicast;dest_addr=":8002"/":8003";
MIKEY=TUlLRVkgZm9yIHN0cmVhbSB0d2lzdGVyLjNncC90cmFja0lEPTQ=
Accept-Ranges: npt, smpte, clock
Pipelined-Requests: 7654
C->M: PLAY rtsps://example.com/twister.3gp/ RTSP/2.0
CSeq: 5
User-Agent: PhonyClient/1.2
Range: npt=0-
Seek-Style: RAP
Pipelined-Requests: 7654
M->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Transport: RTP/SAVP;unicast;
dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";
src_addr="198.51.100.5:9000"/"198.51.100.5:9001";
ssrc=93CB001E;
MIKEY=TUlLRVkgUmVzcG9uc2UgdHdpc3Rlci4zZ3AvdHJhY2tJRD0x
Session: OccldOFFq23KwjYpAnBbUr
Expires: Fri, 20 Dec 2013 12:25:34 +0000
Date: Fri, 20 Dec 2013 10:25:34 +0000
Accept-Ranges: npt
Pipelined-Requests: 7654
Media-Properties: Random-Access=0.2, Immutable, Unlimited
M->C: RTSP/2.0 200 OK
CSeq: 4
Server: PhonyServer/1.0
Transport: RTP/SAVP;unicast;
dest_addr="192.0.2.53:8002"/"192.0.2.53:8003;
src_addr="198.51.100.5:9002"/"198.51.100.5:9003";
ssrc=A813FC13;
MIKEY=TUlLRVkgUmVzcG9uc2UgdHdpc3Rlci4zZ3AvdHJhY2tJRD00
Session: OccldOFFq23KwjYpAnBbUr
Expires: Fri, 20 Dec 2013 12:25:34 +0000
Date: Fri, 20 Dec 2013 10:25:34 +0000
Accept-Range: NPT
Pipelined-Requests: 7654
Media-Properties: Random-Access=0.8, Immutable, Unlimited
M->C: RTSP/2.0 200 OK
CSeq: 5
Server: PhonyServer/1.0
Date: Fri, 20 Dec 2013 10:25:34 +0000
Session: OccldOFFq23KwjYpAnBbUr
Range: npt=0-623.10
Seek-Style: RAP
RTP-Info: url="rtsps://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12345;rtptime=3450012,
url="rtsps://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=54321;rtptime=2876889;
Pipelined-Requests: 7654
A.4. Media on Demand (Unicast)
An alternative example of media on demand with a few more tweaks is
the following. Client C requests a movie distributed from two
different media servers A (audio.example.com) and V
(video.example.com). The media description is stored on a web server
W. The media description contains descriptions of the presentation
and all its streams, including the codecs that are available and the
protocol stack.
In this example, the client is only interested in the last part of
the movie.
C->W: GET /twister.sdp HTTP/1.1
Host: www.example.com
Accept: application/sdp
W->C: HTTP/1.1 200 OK
Date: Wed, 23 Jan 2013 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 278
Expires: Thu, 24 Jan 2013 15:35:06 GMT
v=0
o=- 2890844526 2890842807 IN IP4 198.51.100.5
s=RTSP Session
e=adm@example.com
c=IN IP4 0.0.0.0
a=range:npt=00:00:00-01:49:34
t=0 0
m=audio 0 RTP/AVP 0
a=control:rtsp://audio.example.com/twister/audio.en
m=video 0 RTP/AVP 31
a=control:rtsp://video.example.com/twister/video
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057",
RTP/AVP/TCP;unicast;interleaved=0-1
Accept-Ranges: npt, smpte, clock
A->C: RTSP/2.0 200 OK
CSeq: 1
Session: OccldOFFq23KwjYpAnBbUr
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3056"/"192.0.2.53:3057";
src_addr="198.51.100.5:5000"/"198.51.100.5:5001"
Date: Wed, 23 Jan 2013 15:35:12 +0000
Server: PhonyServer/1.0
Expires: Thu, 24 Jan 2013 15:35:12 +0000
Cache-Control: public
Accept-Ranges: npt, smpte
Media-Properties: Random-Access=0.02, Immutable, Unlimited
C->V: SETUP rtsp://video.example.com/twister/video RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3058"/"192.0.2.53:3059",
RTP/AVP/TCP;unicast;interleaved=0-1
Accept-Ranges: npt, smpte, clock
V->C: RTSP/2.0 200 OK
CSeq: 1
Session: P5it3pMo6xHkjUcDrNkBjf
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3058"/"192.0.2.53:3059";
src_addr="198.51.100.5:5002"/"198.51.100.5:5003"
Date: Wed, 23 Jan 2013 15:35:12 +0000
Server: PhonyServer/1.0
Cache-Control: public
Expires: Thu, 24 Jan 2013 15:35:12 +0000
Accept-Ranges: npt, smpte
Media-Properties: Random-Access=1.2, Immutable, Unlimited
C->V: PLAY rtsp://video.example.com/twister/video RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Session: P5it3pMo6xHkjUcDrNkBjf
Range: smpte=0:10:00-
V->C: RTSP/2.0 200 OK
CSeq: 2
Session: P5it3pMo6xHkjUcDrNkBjf
Range: smpte=0:10:00-1:49:23
Seek-Style: First-Prior
RTP-Info: url="rtsp://video.example.com/twister/video"
ssrc=A17E189D:seq=12312232;rtptime=78712811
Server: PhonyServer/2.0
Date: Wed, 23 Jan 2013 15:35:13 +0000
C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Session: OccldOFFq23KwjYpAnBbUr
Range: smpte=0:10:00-
A->C: RTSP/2.0 200 OK
CSeq: 2
Session: OccldOFFq23KwjYpAnBbUr
Range: smpte=0:10:00-1:49:23
Seek-Style: First-Prior
RTP-Info: url="rtsp://audio.example.com/twister/audio.en"
ssrc=3D124F01:seq=876655;rtptime=1032181
Server: PhonyServer/1.0
Date: Wed, 23 Jan 2013 15:35:13 +0000
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: OccldOFFq23KwjYpAnBbUr
A->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: Wed, 23 Jan 2013 15:36:52 +0000
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: P5it3pMo6xHkjUcDrNkBjf
V->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/2.0
Date: Wed, 23 Jan 2013 15:36:52 +0000
Even though the audio and video track are on two different servers
that may start at slightly different times and may drift with respect
to each other over time, the client can perform initial
synchronization of the two media using RTP-Info and Range received in
the PLAY responses. If the two servers are time synchronized, the
RTCP packets can also be used to maintain synchronization.
A.5. Single-Stream Container Files
Some RTSP servers may treat all files as though they are "container
files", yet other servers may not support such a concept. Because of
this, clients needs to use the rules set forth in the session
description for Request-URIs rather than assuming that a consistent
URI may always be used throughout. Below is an example of how a
multi-stream server might expect a single-stream file to be served:
C->S: DESCRIBE rtsp://foo.example.com/test.wav RTSP/2.0
Accept: application/x-rtsp-mh, application/sdp
CSeq: 1
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 1
Content-base: rtsp://foo.example.com/test.wav/
Content-type: application/sdp
Content-length: 163
Server: PhonyServer/1.0
Date: Wed, 23 Jan 2013 15:36:52 +0000
Expires: Thu, 24 Jan 2013 15:36:52 +0000
v=0
o=- 872653257 872653257 IN IP4 192.0.2.5
s=mu-law wave file
i=audio test
c=IN IP4 0.0.0.0
t=0 0
a=control: *
m=audio 0 RTP/AVP 0
a=control:streamid=0
C->S: SETUP rtsp://foo.example.com/test.wav/streamid=0 RTSP/2.0
Transport: RTP/AVP/UDP;unicast;
dest_addr=":6970"/":6971";mode="PLAY"
CSeq: 2
User-Agent: PhonyClient/1.2
Accept-Ranges: npt, smpte, clock
S->C: RTSP/2.0 200 OK
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:6970"/"192.0.2.53:6971";
src_addr="198.51.100.5:6970"/"198.51.100.5:6971";
mode="PLAY";ssrc=EAB98712
CSeq: 2
Session: NYkqQYKk0bb12BY3goyoyO
Expires: Thu, 24 Jan 2013 15:36:52 +0000
Server: PhonyServer/1.0
Date: Wed, 23 Jan 2013 15:36:52 +0000
Accept-Ranges: npt
Media-Properties: Random-Access=0.5, Immutable, Unlimited
C->S: PLAY rtsp://foo.example.com/test.wav/ RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: NYkqQYKk0bb12BY3goyoyO
S->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: Wed, 23 Jan 2013 15:36:52 +0000
Session: NYkqQYKk0bb12BY3goyoyO
Range: npt=0-600
Seek-Style: RAP
RTP-Info: url="rtsp://foo.example.com/test.wav/streamid=0"
ssrc=0D12F123:seq=981888;rtptime=3781123
Note the different URI in the SETUP command and then the switch back
to the aggregate URI in the PLAY command. This makes complete sense
when there are multiple streams with aggregate control, but it is
less than intuitive in the special case where the number of streams
is one. However, the server has declared the aggregated control URI
in the SDP; therefore, this is legal.
In this case, it is also required that servers accept implementations
that use the non-aggregated interpretation and use the individual
media URI, like this:
C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: NYkqQYKk0bb12BY3goyoyO
A.6. Live Media Presentation Using Multicast
The media server M chooses the multicast address and port. Here, it
is assumed that the web server only contains a pointer to the full
description, while the media server M maintains the full description.
C->W: GET /sessions.html HTTP/1.1
Host: www.example.com
W->C: HTTP/1.1 200 OK
Content-Type: text/html
<html>
...
<a href "rtsp://live.example.com/concert/audio">
Streamed Live Music performance </a>
...
</html>
C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/2.0
CSeq: 1
Supported: play.basic, play.scale
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Content-Type: application/sdp
Content-Length: 183
Server: PhonyServer/1.0
Date: Wed, 23 Jan 2013 15:36:52 +0000
Supported: play.basic
v=0
o=- 2890844526 2890842807 IN IP4 192.0.2.5
s=RTSP Session
t=0 0
m=audio 3456 RTP/AVP 0
c=IN IP4 233.252.0.54/16
a=control: rtsp://live.example.com/concert/audio
a=range:npt=0-
C->M: SETUP rtsp://live.example.com/concert/audio RTSP/2.0
CSeq: 2
Transport: RTP/AVP;multicast;
dest_addr="233.252.0.54:3456"/"233.252.0.54:3457";ttl=16
Accept-Ranges: npt, smpte, clock
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Date: Wed, 23 Jan 2013 15:36:52 +0000
Transport: RTP/AVP;multicast;
dest_addr="233.252.0.54:3456"/"233.252.0.54:3457";ttl=16
;ssrc=4D12AB92/0DF876A3
Session: qHj4jidpmF6zy9v9tNbtxr
Accept-Ranges: npt, clock
Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0
C->M: PLAY rtsp://live.example.com/concert/audio RTSP/2.0
CSeq: 3
Session: qHj4jidpmF6zy9v9tNbtxr
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: Wed, 23 Jan 2013 15:36:52 +0000
Session: qHj4jidpmF6zy9v9tNbtxr
Seek-Style: Next
Range:npt=1256-
RTP-Info: url="rtsp://live.example.com/concert/audio"
ssrc=0D12F123:seq=1473; rtptime=80000
A.7. Capability Negotiation
This example illustrates how the client and server determine their
capability to support a special feature, in this case, "play.scale".
The server, through the client request and the included Supported
header, learns that the client supports RTSP 2.0 and also supports
the playback time scaling feature of RTSP. The server's response
contains the following feature-related information to the client; it
supports the basic media delivery functions (play.basic), the
extended functionality of time scaling of content (play.scale), and
one "example.com" proprietary feature (com.example.flight). The
client also learns the methods supported (Public header) by the
server for the indicated resource.
C->S: OPTIONS rtsp://media.example.com/movie/twister.3gp RTSP/2.0
CSeq: 1
Supported: play.basic, play.scale
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 1
Public:OPTIONS,SETUP,PLAY,PAUSE,TEARDOWN,DESCRIBE,GET_PARAMETER
Allow: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN, DESCRIBE
Server: PhonyServer/2.0
Supported: play.basic, play.scale, com.example.flight
When the client sends its SETUP request, it tells the server that it
requires support of the play.scale feature for this session by
including the Require header.
C->S: SETUP rtsp://media.example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3056"/"192.0.2.53:3057",
RTP/AVP/TCP;unicast;interleaved=0-1
Require: play.scale
Accept-Ranges: npt, smpte, clock
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 3
Session: OccldOFFq23KwjYpAnBbUr
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3056"/"192.0.2.53:3057";
src_addr="198.51.100.5:5000"/"198.51.100.5:5001"
Server: PhonyServer/2.0
Accept-Ranges: npt, smpte
Media-Properties: Random-Access=0.8, Immutable, Unlimited
Appendix B. RTSP Protocol State Machine
The RTSP session state machine describes the behavior of the protocol
from RTSP session initialization through RTSP session termination.
It is probably easiest to think of this as the server's state and
then view the client as needing to track what it believes the
server's state will be based on sent or received RTSP messages.
Thus, in most cases, the state tables below can be read as: if the
client does X, and assuming it fulfills any prerequisite(s), the
(server) state will move to the new state and the indicated response
will returned. However, there are also server-to-client
notifications or requests, where the action describes what
notification or request will occur, its requisites, what new state
will result after the server has received the response, as well as
describing the client's response to the action.
The State machine is defined on a per-session basis, which is
uniquely identified by the RTSP session identifier. The session may
contain one or more media streams depending on state. If a single
media stream is part of the session, it is in non-aggregated control.
If two or more are part of the session, it is in aggregated control.
The below state machine is an informative description of the
protocol's behavior. In case of ambiguity with the earlier parts of
this specification, the description in the earlier parts take
precedence.
B.1. States
The state machine contains three states, described below. For each
state, there exists a table that shows which requests and events are
allowed and whether they will result in a state change.
Init: Initial state, no session exists.
Ready: Session is ready to start playing.
Play: Session is playing, i.e., sending media-stream data in the
direction S->C.
B.2. State Variables
This representation of the state machine needs more than its state to
work. A small number of variables are also needed, and they are
explained below.
NRM: The number of media streams that are part of this session.
RP: Resume point, the point in the presentation time line at which
a request to continue playing will resume from. A time format
for the variable is not mandated.
B.3. Abbreviations
To make the state tables more compact, a number of abbreviations are
used, which are explained below.
IFI: IF Implemented.
md: Media
PP: Pause Point, the point in the presentation timeline at which
the presentation was paused.
Prs: Presentation, the complete multimedia presentation.
RedP: Redirect Point, the point in the presentation timeline at which
a REDIRECT was specified to occur.
SES: Session.
B.4. State Tables
This section contains a table for each state. The table contains all
the requests and events on which this state is allowed to act. The
events that are method names are, unless noted, requests with the
given method in the direction client to server (C->S). In some
cases, there exists one or more requisites. The response column
tells what type of response actions should be performed. Possible
actions that are requested for an event include: response codes,
e.g., 200, headers that need to be included in the response, setting
of state variables, or settings of other session-related parameters.
The new state column tells which state the state machine changes to.
The response to a valid request meeting the requisites is normally a
2xx (SUCCESS) unless otherwise noted in the response column. The
exceptions need to be given a response according to the response
column. If the request does not meet the requisite, is erroneous, or
some other type of error occurs, the appropriate response code is to
be sent. If the response code is a 4xx, the session state is
unchanged. A response code of 3rr will result in that the session
being ended and its state changed to Init. A response code of 304
results in no state change. However, there are restrictions to when
a 3rr response may be used. A 5xx response does not result in any
change of the session state, except if the error is not possible to
recover from. An unrecoverable error results in the ending of the
session. In the general case, if it can't be determined whether or
not it was an unrecoverable error, the client will be required to
test. In the case that the next request after a 5xx is responded to
with a 454 (Session Not Found), the client knows that the session has
ended. For any request message that cannot be responded to within
the time defined in Section 10.4, a 100 response must be sent.
The server will time out the session after the period of time
specified in the SETUP response, if no activity from the client is
detected. Therefore, there exists a timeout event for all states
except Init.
In the case that NRM = 1, the presentation URI is equal to the media
URI or a specified presentation URI. For NRM > 1, the presentation
URI needs to be other than any of the media that are part of the
session. This applies to all states.
+---------------+-----------------+---------------------------------+
| Event | Prerequisite | Response |
+---------------+-----------------+---------------------------------+
| DESCRIBE | Needs REDIRECT | 3rr, Redirect |
| | | |
| DESCRIBE | | 200, Session description |
| | | |
| OPTIONS | Session ID | 200, Reset session timeout |
| | | timer |
| | | |
| OPTIONS | | 200 |
| | | |
| SET_PARAMETER | Valid parameter | 200, change value of parameter |
| | | |
| GET_PARAMETER | Valid parameter | 200, return value of parameter |
+---------------+-----------------+---------------------------------+
Table 9: Non-State-Machine Changing Events
The methods in Table 9 do not have any effect on the state machine or
the state variables. However, some methods do change other session-
related parameters, for example, SET_PARAMETER, which will set the
parameter(s) specified in its body. Also, all of these methods that
allow the Session header will also update the keep-alive timer for
the session.
+------------------+----------------+-----------+-------------------+
| Action | Requisite | New State | Response |
+------------------+----------------+-----------+-------------------+
| SETUP | | Ready | NRM=1, RP=0.0 |
| | | | |
| SETUP | Needs Redirect | Init | 3rr Redirect |
| | | | |
| S -> C: REDIRECT | No Session hdr | Init | Terminate all SES |
+------------------+----------------+-----------+-------------------+
Table 10: State: Init
The initial state of the state machine (Table 10) can only be left by
processing a correct SETUP request. As seen in the table, the two
state variables are also set by a correct request. This table also
shows that a correct SETUP can in some cases be redirected to another
URI or server by a 3rr response.
+-------------+------------------------+---------+------------------+
| Action | Requisite | New | Response |
| | | State | |
+-------------+------------------------+---------+------------------+
| SETUP | New URI | Ready | NRM +=1 |
| | | | |
| SETUP | URI Setup prior | Ready | Change transport |
| | | | param |
| | | | |
| TEARDOWN | Prs URI, | Init | No session hdr, |
| | | | NRM = 0 |
| | | | |
| TEARDOWN | md URI,NRM=1 | Init | No Session hdr, |
| | | | NRM = 0 |
| | | | |
| TEARDOWN | md URI,NRM>1 | Ready | Session hdr, NRM |
| | | | -= 1 |
| | | | |
| PLAY | Prs URI, No range | Play | Play from RP |
| | | | |
| PLAY | Prs URI, Range | Play | According to |
| | | | range |
| | | | |
| PLAY | md URI, NRM=1, Range | Play | According to |
| | | | range |
| | | | |
| PLAY | md URI, NRM=1 | Play | Play from RP |
| | | | |
| PAUSE | Prs URI | Ready | Return PP |
| | | | |
| SC:REDIRECT | Terminate-Reason | Ready | Set RedP |
| | | | |
| SC:REDIRECT | No Terminate-Reason | Init | Session is |
| | time parameter | | removed |
| | | | |
| Timeout | | Init | |
| | | | |
| RedP | | Init | TEARDOWN of |
| reached | | | session |
+-------------+------------------------+---------+------------------+
Table 11: State: Ready
In the Ready state (Table 11), some of the actions depend on the
number of media streams (NRM) in the session, i.e., aggregated or
non-aggregated control. A SETUP request in the Ready state can
either add one more media stream to the session or, if the media
stream (same URI) already is part of the session, change the
transport parameters. TEARDOWN depends on both the Request-URI and
the number of media streams within the session. If the Request-URI
is the presentation URI, the whole session is torn down. If a media
URI is used in the TEARDOWN request and more than one media exists in
the session, the session will remain and a session header is returned
in the response. If only a single media stream remains in the
session when performing a TEARDOWN with a media URI, the session is
removed. The number of media streams remaining after tearing down a
media stream determines the new state.
+----------------+-----------------------+--------+-----------------+
| Action | Requisite | New | Response |
| | | State | |
+----------------+-----------------------+--------+-----------------+
| PAUSE | Prs URI | Ready | Set RP to |
| | | | present point |
| | | | |
| End of media | All media | Play | Set RP = End of |
| | | | media |
| | | | |
| End of range | | Play | Set RP = End of |
| | | | range |
| | | | |
| PLAY | Prs URI, No range | Play | Play from |
| | | | present point |
| | | | |
| PLAY | Prs URI, Range | Play | According to |
| | | | range |
| | | | |
| SC:PLAY_NOTIFY | | Play | 200 |
| | | | |
| SETUP | New URI | Play | 455 |
| | | | |
| SETUP | md URI | Play | 455 |
| | | | |
| SETUP | md URI, IFI | Play | Change |
| | | | transport param.|
| | | | |
| TEARDOWN | Prs URI | Init | No session hdr |
| | | | |
| TEARDOWN | md URI,NRM=1 | Init | No Session hdr, |
| | | | NRM=0 |
| | | | |
| TEARDOWN | md URI | Play | 455 |
| | | | |
| SC:REDIRECT | Terminate Reason with | Play | Set RedP |
| | Time parameter | | |
| | | | |
| SC:REDIRECT | | Init | Session is |
| | | | removed |
| | | | |
| RedP reached | | Init | TEARDOWN of |
| | | | session |
| | | | |
| Timeout | | Init | Stop Media |
| | | | playout |
+----------------+-----------------------+--------+-----------------+
Table 12: State: Play
The Play state table (Table 12) contains a number of requests that
need a presentation URI (labeled as Prs URI) to work on (i.e., the
presentation URI has to be used as the Request-URI). This is due to
the exclusion of non-aggregated stream control in sessions with more
than one media stream.
To avoid inconsistencies between the client and server, automatic
state transitions are avoided. This can be seen at, for example, an
"End of media" event when all media has finished playing but the
session still remains in Play state. An explicit PAUSE request needs
to be sent to change the state to Ready. It may appear that there
exist automatic transitions in "RedP reached" and "PP reached".
However, they are requested and acknowledged before they take place.
The time at which the transition will happen is known by looking at
the Terminate-Reason header's time parameter and Range header,
respectively. If the client sends a request close in time to these
transitions, it needs to be prepared for receiving error messages, as
the state may or may not have changed.
Appendix C. Media-Transport Alternatives
This section defines how certain combinations of protocols, profiles,
and lower transports are used. This includes the usage of the
Transport header's source and destination address parameters:
"src_addr" and "dest_addr".
C.1. RTP
This section defines the interaction of RTSP with respect to the RTP
protocol [RFC3550]. It also defines any necessary media-transport
signaling with regard to RTP.
The available RTP profiles and lower-layer transports are described
below along with rules on signaling the available combinations.
C.1.1. AVP
The usage of the "RTP Profile for Audio and Video Conferences with
Minimal Control" [RFC3551] when using RTP for media transport over
different lower-layer transport protocols is defined below in regard
to RTSP.
One such case is defined within this document: the use of embedded
(interleaved) binary data as defined in Section 14. The usage of
this method is indicated by including the "interleaved" parameter.
When using embedded binary data, "src_addr" and "dest_addr" MUST NOT
be used. This addressing and multiplexing is used as defined with
use of channel numbers and the interleaved parameter.
C.1.2. AVP/UDP
This part describes the sending of RTP [RFC3550] over lower-
transport-layer UDP [RFC768] according to the profile "RTP Profile
for Audio and Video Conferences with Minimal Control" defined in
[RFC3551]. Implementations of RTP/AVP/UDP MUST implement RTCP
(Appendix C.1.6). This profile requires one or two unidirectional or
bidirectional UDP flows per media stream. The first UDP flow is for
RTP and the second is for RTCP. Multiplexing of RTP and RTCP
(Appendix C.1.6.4) MAY be used, in which case, a single UDP flow is
used for both parts. Embedding of RTP data with the RTSP messages,
in accordance with Section 14, SHOULD NOT be performed when RTSP
messages are transported over unreliable transport protocols, like
UDP [RFC768].
The RTP/UDP and RTCP/UDP flows can be established using the Transport
header's "src_addr" and "dest_addr" parameters.
In RTSP PLAY mode, the transmission of RTP packets from client to
server is unspecified. The behavior in regard to such RTP packets
MAY be defined in future.
The "src_addr" and "dest_addr" parameters are used in the following
way for media delivery and playback mode, i.e., Mode=PLAY:
o The "src_addr" and "dest_addr" parameters MUST contain either 1 or
2 address specifications. Note that two address specifications
MAY be provided even if RTP and RTCP multiplexing is negotiated.
o Each address specification for RTP/AVP/UDP or RTP/AVP/TCP MUST
contain either:
* both an address and a port number, or
* a port number without an address.
o The first address specification given in either of the parameters
applies to the RTP stream. The second specification, if present,
applies to the RTCP stream, unless in the case RTP and RTCP
multiplexing is negotiated where both RTP and RTCP will use the
first specification.
o The RTP/UDP packets from the server to the client MUST be sent to
the address and port given by the first address specification of
the "dest_addr" parameter.
o The RTCP/UDP packets from the server to the client MUST be sent to
the address and port given by the second address specification of
the "dest_addr" parameter, unless RTP and RTCP multiplexing has
been negotiated, in which case RTCP MUST be sent to the first
address specification. If no second pair is specified and RTP and
RTCP multiplexing has not been negotiated, RTCP MUST NOT be sent.
o The RTCP/UDP packets from the client to the server MUST be sent to
the address and port given by the second address specification of
the "src_addr" parameter, unless RTP and RTCP multiplexing has
been negotiated, in which case RTCP MUST be sent to the first
address specification. If no second pair is specified and RTP and
RTCP multiplexing has not been negotiated, RTCP MUST NOT be sent.
o The RTP/UDP packets from the client to the server MUST be sent to
the address and port given by the first address specification of
the "src_addr" parameter.
o RTP and RTCP packets SHOULD be sent from the corresponding
receiver port, i.e., RTCP packets from the server should be sent
from the "src_addr" parameters second address port pair, unless
RTP and RTCP multiplexing has been negotiated in which case the
first address port pair is used.
C.1.3. AVPF/UDP
The RTP profile "Extended RTP Profile for RTCP-based Feedback (RTP/
AVPF)" [RFC4585] MAY be used as RTP profiles in sessions using RTP.
All that is defined for AVP MUST also apply for AVPF.
The usage of AVPF is indicated by the media initialization protocol
used. In the case of SDP, it is indicated by media lines ("m=")
containing the profile RTP/AVPF. That SDP MAY also contain further
AVPF-related SDP attributes configuring the AVPF session regarding
reporting interval and feedback messages to be used [RFC4585]. This
configuration MUST be followed.
C.1.4. SAVP/UDP
The RTP profile "The Secure Real-time Transport Protocol (SRTP)"
[RFC3711] is an RTP profile (SAVP) that MAY be used in RTSP sessions
using RTP. All that is defined for AVP MUST also apply for SAVP.
The usage of SRTP requires that a security context be established.
The default key-management unless otherwise signaled SHALL be MIKEY
in RSA-R mode as defined in Appendix C.1.4.1 and not according to the
procedure defined in "Key Management Extensions for Session
Description Protocol (SDP) and Real Time Streaming Protocol (RTSP)"
[RFC4567]. The reason is that RFC 4567 sends the initial MIKEY
message in SDP, thus, both requiring the usage of the DESCRIBE method
and forcing the server to keep state for clients performing DESCRIBE
in anticipation that they might require key management.
MIKEY is selected as the default method for establishing SRTP
cryptographic context within an RTSP session as it can be embedded in
the RTSP messages while still ensuring confidentiality of content of
the keying material, even when using hop-by-hop TLS security for the
RTSP messages. This method also supports pipelining of the RTSP
messages.
C.1.4.1. MIKEY Key Establishment
This method for using MIKEY [RFC3830] to establish the SRTP
cryptographic context is initiated in the client's SETUP request, and
the server's response to the SETUP carries the MIKEY response. This
ensures that the crypto context establishment happens simultaneously
with the establishment of the media stream being protected. By using
MIKEY's RSA-R mode [RFC4738] the client can be the initiator and
still allow the server to set the parameters in accordance with the
actual media stream.
The SRTP cryptographic context establishment is done according to the
following process:
1. The client determines that SAVP or SAVPF shall be used from the
media-description format, e.g., SDP. If no other key-management
method is explicitly signaled, then MIKEY SHALL be used as
defined herein. The use of SRTP with RTSP is only defined with
MIKEY with keys established as defined in this section. Future
documents may define how an RTSP implementation treats SDP that
indicates some other key mechanism to be used. The need for
such specification includes [RFC4567], which is not defined for
use in RTSP 2.0 within this document.
2. The client SHALL establish a TLS connection for RTSP messages,
directly or hop-by-hop with the server. If hop-by-hop TLS
security is used, the User method SHALL be indicated in the
Accept-Credentials header. Note that using hop-by-hop does
allow the proxy to insert itself as a man in the middle. This
can also occur in the MIKEY exchange by the proxy providing one
of its certificates rather than the server's in the Connection-
Credentials header. Therefore, the client SHALL validate the
server certificate.
3. The client retrieves the server's certificate from a direct TLS
connection or hop-by-hop from a Connection-Credentials header.
The client then checks that the server certificate is valid and
belongs to the server.
4. The client forms the MIKEY Initiator message using RSA-R mode in
unicast mode as specified in [RFC4738]. The client SHOULD use
the same certificate for TLS and MIKEY to enable the server to
bind the two together. The client's certificate SHALL be
included in the MIKEY message. The client SHALL indicate its
SRTP capabilities in the message.
5. The MIKEY message from the previous step is base64-encoded
[RFC4648] and becomes the value of the MIKEY parameter that is
included in the transport specification(s) that specifies an
SRTP-based profile (SAVP, SAVPF) in the SETUP request.
6. Any proxy encountering the MIKEY parameter SHALL forward it
without modification. A proxy that is required to understand
the Transport specifications will need to understand SAVP/SAVPF
with MIKEY to enable the default keying for SRTP-protected media
streams. If such a proxy does not support SAVP/SAVPF with
MIKEY, it will discard the whole transport specification. Most
types of proxies can easily support SAVP and SAVPF with MIKEY.
If a client encounters a proxy not supporting SAVP/SAVPF with
MIKEY, the client should attempt bypassing that proxy.
7. The server, upon receiving the SETUP request, will need to
decide upon the transport specification to use, if multiple are
included by the client. In the determination of which transport
specifications are supported and preferred, the server SHOULD
decode the MIKEY message to take the embedded SRTP parameters
into account. If all transport spec require SRTP but no MIKEY
parameter or other supported keying method is included, the
server SHALL respond with 403 (Forbidden).
8. Upon generating a response, the following outcomes can occur:
* A transport spec not using SRTP and MIKEY is selected. Thus,
the response will not contain any MIKEY parameters.
* A transport spec using SRTP and MIKEY is selected but an
error is encountered in the MIKEY processing. In this case,
an RTSP error response code of 466 (Key Management Error)
SHALL be used. A MIKEY message describing the error MAY be
included.
* A transport spec using SRTP and MIKEY is selected and a MIKEY
response message can be created. The server SHOULD use the
same certificate for TLS and in MIKEY to enable the client to
bind the two together. If a different certificate is used,
it SHALL be included in the MIKEY message. It is RECOMMENDED
that the envelope key-cache type be set to 'Cache' and that a
single envelope key is reused for all MIKEY messages to the
client. That message is included in the MIKEY parameter part
of the single selected transport specification in the SETUP
response. The server will set the SRTP parameters as
preferred for this media stream within the supported range by
the client.
9. The server transmits the SETUP response back to the client.
10. The client receives the SETUP response and, if the response code
indicates a successful request, it decodes the MIKEY message and
establishes the SRTP cryptographic context from the parameters
in the MIKEY response.
In the above method, the client's certificate may be self signed in
cases where the client's identity is not necessary to authenticate
and the security goal is only to ensure that the RTSP signaling
client is the same as the one receiving the SRTP security context.
C.1.5. SAVPF/UDP
The RTP profile "Extended Secure RTP Profile for Real-time Transport
Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)" [RFC5124] is an
RTP profile (SAVPF) that MAY be used in RTSP sessions using RTP. All
that is defined for AVPF MUST also apply for SAVPF.
The usage of SRTP requires that a cryptographic context be
established. The default mechanism for establishing that security
association is to use MIKEY[RFC3830] with RTSP as defined in
Appendix C.1.4.1.
C.1.6. RTCP Usage with RTSP
RTCP has several usages when RTP is implemented for media transport
as explained below. Thus, RTCP MUST be supported if an RTSP agent
handles RTP.
C.1.6.1. Media Synchronization
RTCP provides media synchronization and clock-drift compensation.
The initial media synchronization is available from RTP-Info header.
However, to be able to handle any clock drift between the media
streams, RTCP is needed.
C.1.6.2. RTSP Session Keep-Alive
RTCP traffic from the RTSP client to the RTSP server MUST function as
keep-alive. This requires an RTSP server supporting RTP to use the
received RTCP packets as indications that the client desires the
related RTSP session to be kept alive.
C.1.6.3. Bitrate Adaption
RTCP Receiver reports and any additional feedback from the client
MUST be used to adapt the bitrate used over the transport for all
cases when RTP is sent over UDP. An RTP sender without reserved
resources MUST NOT use more than its fair share of the available
resources. This can be determined by comparing on short-to-medium
terms (some seconds) the used bitrate and adapting it so that the RTP
sender sends at a bitrate comparable to what a TCP sender would
achieve on average over the same path.
To ensure that the implementation's adaptation mechanism has a well-
defined outer envelope, all implementations using a non-congestion-
controlled unicast transport protocol, like UDP, MUST implement
"Multimedia Congestion Control: Circuit Breakers for Unicast RTP
Sessions" [RTP-CIRCUIT-BREAKERS].
C.1.6.4. RTP and RTCP Multiplexing
RTSP can be used to negotiate the usage of RTP and RTCP multiplexing
as described in [RFC5761]. This allows servers and client to reduce
the amount of resources required for the session by only requiring
one underlying transport stream per media stream instead of two when
using RTP and RTCP. This lessens the server-port consumption and
also the necessary state and keep-alive work when operating across
NATs [RFC2663].
Content must be prepared with some consideration for RTP and RTCP
multiplexing, mainly ensuring that the RTP payload types used do not
collide with the ones used for RTCP packet types. This option likely
needs explicit support from the content unless the RTP payload types
can be remapped by the server and that is correctly reflected in the
session description. Beyond that, support of this feature should
come at little cost and much gain.
It is recommended that, if the content and server support RTP and
RTCP multiplexing, this is indicated in the session description, for
example, using the SDP attribute "a=rtcp-mux". If the SDP message
contains the "a=rtcp-mux" attribute for a media stream, the server
MUST support RTP and RTCP multiplexing. If indicated or otherwise
desired by the client, it can include the Transport parameter "RTCP-
mux" in any transport specification where it desires to use "RTCP-
mux". The server will indicate if it supports "RTCP-mux". Servers
and Clients SHOULD support RTP and RTCP multiplexing.
For capability exchange, an RTSP feature tag for RTP and RTCP
multiplexing is defined: "setup.rtp.rtcp.mux".
To minimize the risk of negotiation failure while using RTP and RTCP
multiplexing, some recommendations are here provided. If the session
description includes explicit indication of support ("a=rtcp-mux" in
SDP), then an RTSP agent can safely create a SETUP request with a
transport specification with only a single "dest_addr" parameter
address specification. If no such explicit indication is provided,
then even if the feature tag "setup.rtp.rtcp.mux" is provided in a
Supported header by the RTSP server or the feature tag included in
the Required header in the SETUP request, the media resource may not
support RTP and RTCP multiplexing. Thus, to maximize the probability
of successful negotiation, the RTSP agent is recommended to include
two "dest_addr" parameter address specifications in the first or
first set (if pipelining is used) of SETUP request(s) for any media
resource aggregate. That way, the RTSP server can accept RTP and
RTCP multiplexing and only use the first address specification or, if
not, use both specifications. The RTSP agent, after having received
the response for a successful negotiation of the usage of RTP and
RTCP multiplexing, can then release the resources associated with the
second address specification.
C.2. RTP over TCP
Transport of RTP over TCP can be done in two ways: over independent
TCP connections using [RFC4571] or interleaved in the RTSP
connection. In both cases, the protocol MUST be "rtp" and the lower-
layer MUST be TCP. The profile may be any of the above specified
ones: AVP, AVPF, SAVP, or SAVPF.
C.2.1. Interleaved RTP over TCP
The use of embedded (interleaved) binary data transported on the RTSP
connection is possible as specified in Section 14. When using this
declared combination of interleaved binary data, the RTSP messages
MUST be transported over TCP. TLS may or may not be used. If TLS is
used, both RTSP messages and the binary data will be protected by
TLS.
One should, however, consider that this will result in all media
streams going through any proxy. Using independent TCP connections
can avoid that issue.
C.2.2. RTP over Independent TCP
In this section, the sending of RTP [RFC3550] over lower-layer
transport TCP [RFC793] according to "Framing Real-time Transport
Protocol (RTP) and RTP Control Protocol (RTCP) Packets over
Connection-Oriented Transport" [RFC4571] is described. This section
adapts the guidelines for using RTP over TCP within SIP/SDP [RFC4145]
to work with RTSP.
A client codes the support of RTP over independent TCP by specifying
an RTP/AVP/TCP transport option without an interleaved parameter in
the Transport line of a SETUP request. This transport option MUST
include the "unicast" parameter.
If the client wishes to use RTP with RTCP, two address specifications
need to be included in the "dest_addr" parameter. If the client
wishes to use RTP without RTCP, one address specification is included
in the "dest_addr" parameter. If the client wishes to multiplex RTP
and RTCP on a single transport flow (see Appendix C.1.6.4), one or
two address specifications are included in the "dest_addr" parameter
in addition to the "RTCP-mux" transport parameter. Two address
specifications are allowed to facilitate successful negotiation when
the server or content can't support RTP and RTCP multiplexing.
Ordering rules of dest_addr ports follow the rules for RTP/AVP/UDP.
If the client wishes to play the active role in initiating the TCP
connection, it MAY set the setup parameter (see Section 18.54) on the
Transport line to be "active", or it MAY omit the setup parameter, as
active is the default. If the client signals the active role, the
ports in the address specifications in the "dest_addr" parameter MUST
be set to 9 (the discard port).
If the client wishes to play the passive role in TCP connection
initiation, it MUST set the setup parameter on the Transport line to
be "passive". If the client is able to assume the active or the
passive role, it MUST set the setup parameter on the Transport line
to be "actpass". In either case, the "dest_addr" parameter's address
specification port value for RTP MUST be set to the TCP port number
on which the client is expecting to receive the TCP connection for
RTP, and the "dest_addr" address specification port value for RTCP
MUST be set to the TCP port number on which the client is expecting
to receive the TCP connection for RTCP. In the case that the client
wishes to multiplex RTP and RTCP on a single transport flow, the
"RTCP-mux" parameter is included and one or two "dest_addr" parameter
address specifications are included, as mentioned earlier in this
section.
Upon receipt of a non-interleaved RTP/AVP/TCP SETUP request, if a
server decides to accept this requested option, the 2xx reply MUST
contain a Transport option that specifies RTP/AVP/TCP (without using
the interleaved parameter and using the unicast parameter). The
"dest_addr" parameter value MUST be echoed from the parameter value
in the client request unless the destination address (only port) was
not provided; in which case, the server MAY include the source
address of the RTSP TCP connection with the port number unchanged.
In addition, the server reply MUST set the setup parameter on the
Transport line, to indicate the role the server will play in the
connection setup. Permissible values are "active" (if a client set
setup to "passive" or "actpass") and "passive" (if a client set setup
to "active" or "actpass").
If a server sets setup to "passive", the "src_addr" in the reply MUST
indicate the ports on which the server is willing to receive a TCP
connection for RTP and (if the client requested a TCP connection for
RTCP by specifying two "dest_addr" address specifications) a TCP/
RTCP connection. If a server sets setup to "active", the ports
specified in "src_addr" address specifications MUST be set to 9. The
server MAY use the "ssrc" parameter, following the guidance in
Section 18.54. The server sets only one address specification in the
case that the client has indicated only a single address
specification or in case RTP and RTCP multiplexing was requested and
accepted by the server. Port ordering for "src_addr" follows the
rules for RTP/AVP/UDP.
Servers MUST support taking the passive role and MAY support taking
the active role. Servers with a public IP address take the passive
role, thus enabling clients behind NATs and firewalls a better chance
of successful connect to the server by actively connecting outwards.
Therefore, the clients are RECOMMENDED to take the active role.
After sending (receiving) a 2xx reply for a SETUP method for a non-
interleaved RTP/AVP/TCP media stream, the active party SHOULD
initiate the TCP connection as soon as possible. The client MUST NOT
send a PLAY request prior to the establishment of all the TCP
connections negotiated using SETUP for the session. In case the
server receives a PLAY request in a session that has not yet
established all the TCP connections, it MUST respond using the 464
(Data Transport Not Ready Yet) (Section 17.4.28) error code.
Once the PLAY request for a media resource transported over non-
interleaved RTP/AVP/TCP occurs, media begins to flow from server to
client over the RTP TCP connection, and RTCP packets flow
bidirectionally over the RTCP TCP connection. Unless RTP and RTCP
multiplexing has been negotiated; in which case, RTP and RTCP will
flow over a common TCP connection. As in the RTP/UDP case, client-
to-server traffic on an RTP-only TCP session is unspecified by this
memo. The packets that travel on these connections MUST be framed
using the protocol defined in [RFC4571], not by the framing defined
for interleaving RTP over the RTSP connection defined in Section 14.
A successful PAUSE request for media being transported over RTP/AVP/
TCP pauses the flow of packets over the connections, without closing
the connections. A successful TEARDOWN request signals that the TCP
connections for RTP and RTCP are to be closed by the RTSP client as
soon as possible.
Subsequent SETUP requests using a URI already set up in an RTSP
session using an RTP/AVP/TCP transport specification may be ambiguous
in the following way: does the client wish to open up a new TCP
connection for RTP or RTCP for the URI, or does the client wish to
continue using the existing TCP connections? The client SHOULD use
the "connection" parameter (defined in Section 18.54) on the
Transport line to make its intention clear (by setting "connection"
to "new" if new connections are needed, and by setting "connection"
to "existing" if the existing connections are to be used). After a
2xx reply for a SETUP request for a new connection, parties should
close the preexisting connections, after waiting a suitable period
for any stray RTP or RTCP packets to arrive.
The usage of SRTP, i.e., either SAVP or SAVPF profiles, requires that
a security association be established. The default mechanism for
establishing that security association is to use MIKEY[RFC3830] with
RTSP as defined Appendix C.1.4.1.
Below, a rewritten version of the example "Media on Demand"
(Appendix A.1) shows the use of RTP/AVP/TCP non-interleaved:
C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Server: PhonyServer/1.0
Date: Wed, 23 Jan 2013 15:36:52 +0000
Content-Type: application/sdp
Content-Length: 227
Content-Base: rtsp://example.com/twister.3gp/
Expires: Thu, 24 Jan 2013 15:36:52 +0000
v=0
o=- 2890844256 2890842807 IN IP4 198.51.100.34
s=RTSP Session
i=An Example of RTSP Session Usage
e=adm@example.com
c=IN IP4 0.0.0.0
a=control: *
a=range:npt=00:00:00-00:10:34.10
t=0 0
m=audio 0 RTP/AVP 0
a=control: trackID=1
C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP/TCP;unicast;dest_addr=":9"/":9";
setup=active;connection=new
Accept-Ranges: npt, smpte, clock
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Transport: RTP/AVP/TCP;unicast;
dest_addr=":9"/":9";
src_addr="198.51.100.5:53478"/"198.51.100:54091";
setup=passive;connection=new;ssrc=93CB001E
Session: OccldOFFq23KwjYpAnBbUr
Expires: Thu, 24 Jan 2013 15:36:52 +0000
Date: Wed, 23 Jan 2013 15:36:52 +0000
Accept-Ranges: npt
Media-Properties: Random-Access=0.8, Immutable, Unlimited
C->M: TCP Connection Establishment x2
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 4
User-Agent: PhonyClient/1.2
Range: npt=30-
Session: OccldOFFq23KwjYpAnBbUr
M->C: RTSP/2.0 200 OK
CSeq: 4
Server: PhonyServer/1.0
Date: Wed, 23 Jan 2013 15:36:54 +0000
Session: OccldOFFq23KwjYpAnBbUr
Range: npt=30-623.10
Seek-Style: First-Prior
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=54321;rtptime=2876889
C.3. Handling Media-Clock Time Jumps in the RTP Media Layer
RTSP allows media clients to control selected, non-contiguous
sections of media presentations, rendering those streams with an RTP
media layer [RFC3550]. Two cases occur, the first is when a new PLAY
request replaces an old ongoing request and the new request results
in a jump in the media. This should produce continuous media stream
at the RTP layer. A client may also immediately follow a completed
PLAY request with a new PLAY request. This will result in some gap
in the media layer. The below text will look into both cases.
A PLAY request that replaces an ongoing PLAY request allows the media
layer rendering the RTP stream to do so continuously without being
affected by jumps in media-clock time. The RTP timestamps for the
new media range are set so that they become continuous with the
previous media range in the previous request. The RTP sequence
number for the first packet in the new range will be the next
following the last packet in the previous range, i.e., monotonically
increasing. The goal is to allow the media-rendering layer to work
without interruption or reconfiguration across the jumps in media
clock. This should be possible in all cases of replaced PLAY
requests for media that has random access properties. In this case,
care is needed to align frames or similar media-dependent structures.
In cases where jumps in media-clock time are a result of RTSP
signaling operations arriving after a completed PLAY operation, the
request timing will result in that media becoming non-continuous.
The server becomes unable to send the media so that it arrives timely
and still carries timestamps to make the media stream continuous. In
these situations, the server will produce RTP streams where there are
gaps in the RTP timeline for the media. If the media has frame
structure, aligning the timestamp for the next frame with the
previous structure reduces the burden to render this media. The gap
should represent the time the server hasn't been serving media, e.g.,
the time between the end of the media stream or a PAUSE request and
the new PLAY request. In these cases, the RTP sequence number would
normally be monotonically increasing across the gap.
For RTSP sessions with media that lacks random access properties,
such as live streams, any media-clock jump is commonly the result of
a correspondingly long pause of delivery. The RTP timestamp will
have increased in direct proportion to the duration of the paused
delivery. Note also that in this case the RTP sequence number should
be the next packet number. If not, the RTCP packet loss reporting
will indicate as loss all packets not received between the point of
pausing and later resuming. This may trigger congestion avoidance
mechanisms. An allowed exception from the above recommendation on
monotonically increasing RTP sequence number is live media streams,
likely being relayed. In this case, when the client resumes
delivery, it will get the media that is currently being delivered to
the server itself. For this type of basic delivery of live streams
to multiple users over unicast, individual rewriting of RTP sequence
numbers becomes quite a burden. For solutions that already cache
media or perform time shifting, the rewriting should impose only a
minor burden.
The goal when handling jumps in media-clock time is that the provided
stream is continuous without gaps in RTP timestamp or sequence
number. However, when delivery has been halted for some reason, the
RTP timestamp, when resuming, MUST represent the duration that the
delivery was halted. An RTP sequence number MUST generally be the
next number, i.e., monotonically increasing modulo 65536. For media
resources with the properties Time-Progressing and Time-Duration=0.0,
the server MAY create RTP media streams with RTP sequence number
jumps in them due to the client first halting delivery and later
resuming it (PAUSE and then later PLAY). However, servers utilizing
this exception must take into consideration the resulting RTCP
receiver reports that likely contain loss reports for all the packets
that were a part of the discontinuity. A client cannot rely on the
fact that a server will align when resuming play, even if it is
RECOMMENDED. The RTP-Info header will provide information on how the
server acts in each case.
One cannot assume that the RTSP client can communicate with the
RTP media agent, as the two may be independent processes. If the
RTP timestamp shows the same gap as the NPT, the media agent will
assume that there is a pause in the presentation. If the jump in
NPT is large enough, the RTP timestamp may roll over and the media
agent may believe later packets to be duplicates of packets just
played out. Having the RTP timestamp jump will also affect the
RTCP measurements based on this.
As an example, assume an RTP timestamp frequency of 8000 Hz, a
packetization interval of 100 ms, and an initial sequence number and
timestamp of zero.
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 4
Session: ymIqLXufHkMHGdtENdblWK
Range: npt=10-15
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 4
Session: ymIqLXufHkMHGdtENdblWK
Range: npt=10-15
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=0;rtptime=0
The ensuing RTP data stream is depicted below:
S -> C: RTP packet - seq = 0, rtptime = 0, NPT time = 10s
S -> C: RTP packet - seq = 1, rtptime = 800, NPT time = 10.1s
. . .
S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s
Upon the completion of the requested delivery, the server sends a
PLAY_NOTIFY.
S->C: PLAY_NOTIFY rtsp://example.com/fizzle RTSP/2.0
CSeq: 5
Notify-Reason: end-of-stream
Request-Status: cseq=4 status=200 reason="OK"
Range: npt=-15
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=49;rtptime=39200
Session: ymIqLXufHkMHGdtENdblWK
C->S: RTSP/2.0 200 OK
CSeq: 5
User-Agent: PhonyClient/1.2
Upon the completion of the play range, the client follows up with a
request to PLAY from a new NPT.
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 6
Session: ymIqLXufHkMHGdtENdblWK
Range: npt=18-20
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 6
Session: ymIqLXufHkMHGdtENdblWK
Range: npt=18-20
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=50;rtptime=40100
The ensuing RTP data stream is depicted below:
S->C: RTP packet - seq = 50, rtptime = 40100, NPT time = 18s
S->C: RTP packet - seq = 51, rtptime = 40900, NPT time = 18.1s
. . .
S->C: RTP packet - seq = 69, rtptime = 55300, NPT time = 19.9s
In this example, first, NPT 10 through 15 are played, then the client
requests the server to skip ahead and play NPT 18 through 20. The
first segment is presented as RTP packets with sequence numbers 0
through 49 and timestamps 0 through 39,200. The second segment
consists of RTP packets with sequence numbers 50 through 69, with
timestamps 40,100 through 55,200. While there is a gap in the NPT,
there is no gap in the sequence-number space of the RTP data stream.
The RTP timestamp gap is present in the above example due to the time
it takes to perform the second play request, in this case, 12.5 ms
(100/8000).
C.4. Handling RTP Timestamps after PAUSE
During a PAUSE/PLAY interaction in an RTSP session, the duration of
time for which the RTP transmission was halted MUST be reflected in
the RTP timestamp of each RTP stream. The duration can be calculated
for each RTP stream as the time elapsed from when the last RTP packet
was sent before the PAUSE request was received and when the first RTP
packet was sent after the subsequent PLAY request was received. The
duration includes all latency incurred and processing time required
to complete the request.
RFC 3550 [RFC3550] states that: "the RTP timestamp for each unit
[packet] would be related to the wallclock time at which the unit
becomes current on the virtual presentation timeline".
In order to satisfy the requirements of [RFC3550], the RTP
timestamp space needs to increase continuously with real time.
While this is not optimal for stored media, it is required for RTP
and RTCP to function as intended. Using a continuous RTP
timestamp space allows the same timestamp model for both stored
and live media and allows better opportunity to integrate both
types of media under a single control.
As an example, assume a clock frequency of 8000 Hz, a packetization
interval of 100 ms, and an initial sequence number and timestamp of
zero.
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 4
Session: ymIqLXufHkMHGdtENdblWK
Range: npt=10-15
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 4
Session: ymIqLXufHkMHGdtENdblWK
Range: npt=10-15
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=0;rtptime=0
The ensuing RTP data stream is depicted below:
S -> C: RTP packet - seq = 0, rtptime = 0, NPT time = 10s
S -> C: RTP packet - seq = 1, rtptime = 800, NPT time = 10.1s
S -> C: RTP packet - seq = 2, rtptime = 1600, NPT time = 10.2s
S -> C: RTP packet - seq = 3, rtptime = 2400, NPT time = 10.3s
The client then sends a PAUSE request:
C->S: PAUSE rtsp://example.com/fizzle RTSP/2.0
CSeq: 5
Session: ymIqLXufHkMHGdtENdblWK
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 5
Session: ymIqLXufHkMHGdtENdblWK
Range: npt=10.4-15
20 seconds elapse and then the client sends a PLAY request. In
addition, the server requires 15 ms to process the request:
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 6
Session: ymIqLXufHkMHGdtENdblWK
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 6
Session: ymIqLXufHkMHGdtENdblWK
Range: npt=10.4-15
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=4;rtptime=164400
The ensuing RTP data stream is depicted below:
S -> C: RTP packet - seq = 4, rtptime = 164400, NPT time = 10.4s
S -> C: RTP packet - seq = 5, rtptime = 165200, NPT time = 10.5s
S -> C: RTP packet - seq = 6, rtptime = 166000, NPT time = 10.6s
First, NPT 10 through 10.3 is played, then a PAUSE is received by the
server. After 20 seconds, a PLAY is received by the server that
takes 15 ms to process. The duration of time for which the session
was paused is reflected in the RTP timestamp of the RTP packets sent
after this PLAY request.
A client can use the RTSP Range header and RTP-Info header to map NPT
time of a presentation with the RTP timestamp.
Note: in RFC 2326 [RFC2326], this matter was not clearly defined and
was misunderstood commonly. However, for RTSP 2.0, it is expected
that this will be handled correctly and no exception handling will be
required.
Note further: it may be required to reset some of the state to ensure
the correct media decoding and the usual jitter-buffer handling when
issuing a PLAY request.
C.5. RTSP/RTP Integration
For certain data types, tight integration between the RTSP layer and
the RTP layer will be necessary. This by no means precludes the
above restrictions. Combined RTSP/RTP media clients should use the
RTP-Info field to determine whether incoming RTP packets were sent
before or after a seek or before or after a PAUSE.
C.6. Scaling with RTP
For scaling (see Section 18.46), RTP timestamps should correspond to
the rendering timing. For example, when playing video recorded at 30
frames per second at a scale of two and speed (Section 18.50) of one,
the server would drop every second frame to maintain and deliver
video packets with the normal timestamp spacing of 3,000 per frame,
but NPT would increase by 1/15 second for each video frame.
Note: the above scaling puts requirements on the media codec or a
media stream to support it. For example, motion JPEG or other
non-predictive video coding can easier handle the above example.
C.7. Maintaining NPT Synchronization with RTP Timestamps
The client can maintain a correct display of NPT by noting the RTP
timestamp value of the first packet arriving after repositioning.
The sequence parameter of the RTP-Info (Section 18.45) header
provides the first sequence number of the next segment.
C.8. Continuous Audio
For continuous audio, the server SHOULD set the RTP marker bit at the
beginning of serving a new PLAY request or at jumps in timeline.
This allows the client to perform playout delay adaptation.
C.9. Multiple Sources in an RTP Session
Note that more than one SSRC MAY be sent in the media stream. If it
happens, all sources are expected to be rendered simultaneously.
C.10. Usage of SSRCs and the RTCP BYE Message during an RTSP Session
The RTCP BYE message indicates the end of use of a given SSRC. If
all sources leave an RTP session, it can, in most cases, be assumed
to have ended. Therefore, a client or server MUST NOT send an RTCP
BYE message until it has finished using a SSRC. A server SHOULD keep
using an SSRC until the RTP session is terminated. Prolonging the
use of a SSRC allows the established synchronization context
associated with that SSRC to be used to synchronize subsequent PLAY
requests even if the PLAY response is late.
An SSRC collision with the SSRC that transmits media does also have
consequences, as it will normally force the media sender to change
its SSRC in accordance with the RTP specification [RFC3550].
However, an RTSP server may wait and see if the client changes and
thus resolve the conflict to minimize the impact. As media sender,
SSRC change will result in a loss of synchronization context and
require any receiver to wait for RTCP sender reports for all media
requiring synchronization before being able to play out synchronized.
Due to these reasons, a client joining a session should take care not
to select the same SSRC(s) as the server indicates in the ssrc
Transport header parameter. Any SSRC signaled in the Transport
header MUST be avoided. A client detecting a collision prior to
sending any RTP or RTCP messages SHALL also select a new SSRC.
C.11. Future Additions
It is the intention that any future protocol or profile regarding
media delivery and lower transport should be easy to add to RTSP.
This section provides the necessary steps that need to be met.
The following things need to be considered when adding a new protocol
or profile for use with RTSP:
o The protocol or profile needs to define a name tag representing
it. This tag is required to be an ABNF "token" to be possible to
use in the Transport header specification.
o The useful combinations of protocol, profiles, and lower-layer
transport for this extension need to be defined. For each
combination, declare the necessary parameters to use in the
Transport header.
o For new media protocols, the interaction with RTSP needs to be
addressed. One important factor will be the media
synchronization. It may be necessary to have new headers similar
to RTP info to carry this information.
o Discussion needs to occur regarding congestion control for media,
especially if transport without built-in congestion control is
used.
See the IANA Considerations section (Section 22) for information on
how to register new attributes.
Appendix D. Use of SDP for RTSP Session Descriptions
The Session Description Protocol (SDP, [RFC4566]) may be used to
describe streams or presentations in RTSP. This description is
typically returned in reply to a DESCRIBE request on a URI from a
server to a client or received via HTTP from a server to a client.
This appendix describes how an SDP file determines the operation of
an RTSP session. Thus, it is worth pointing out that the
interpretation of the SDP is done in the context of the SDP receiver,
which is the one being configured. This is the same as in SAP
[RFC2974]; this differs from SDP Offer/Answer [RFC3264] where each
SDP is interpreted in the context of the agent providing it.
SDP as is provides no mechanism by which a client can distinguish,
without human guidance, between several media streams to be rendered
simultaneously and a set of alternatives (e.g., two audio streams
spoken in different languages). The SDP extension found in "The
Session Description Protocol (SDP) Grouping Framework" [RFC5888]
provides such functionality to some degree. Appendix D.4 describes
the usage of SDP media line grouping for RTSP.
D.1. Definitions
The terms "session-level", "media-level", and other key/attribute
names and values used in this appendix are to be used as defined in
SDP [RFC4566]:
D.1.1. Control URI
The "a=control" attribute is used to convey the control URI. This
attribute is used both for the session and media descriptions. If
used for individual media, it indicates the URI to be used for
controlling that particular media stream. If found at the session
level, the attribute indicates the URI for aggregate control
(presentation URI). The session-level URI MUST be different from any
media-level URI. The presence of a session-level control attribute
MUST be interpreted as support for aggregated control. The control
attribute MUST be present on the media level unless the presentation
only contains a single media stream; in which case, the attribute MAY
be present on the session level only and then also apply to that
single media stream.
ABNF for the attribute is defined in Section 20.3.
Example:
a=control:rtsp://example.com/foo
This attribute MAY contain either relative or absolute URIs,
following the rules and conventions set out in RFC 3986 [RFC3986].
Implementations MUST look for a base URI in the following order:
1. the RTSP Content-Base field;
2. the RTSP Content-Location field;
3. the RTSP Request-URI.
If this attribute contains only an asterisk (*), then the URI MUST be
treated as if it were an empty embedded URI; thus, it will inherit
the entire base URI.
Note: RFC 2326 was very unclear on the processing of relative URIs
and several RTSP 1.0 implementations at the point of publishing
this document did not perform RFC 3986 processing to determine the
resulting URI; instead, simple concatenation is common. To avoid
this issue completely, it is recommended to use absolute URIs in
the SDP.
The URI handling for SDPs from container files needs special
consideration. For example, let's assume that a container file has
the URI: "rtsp://example.com/container.mp4". Let's further assume
this URI is the base URI and that there is an absolute media-level
URI: "rtsp://example.com/container.mp4/trackID=2". A relative media-
level URI that resolves in accordance with RFC 3986 [RFC3986] to the
above given media URI is "container.mp4/trackID=2". It is usually
not desirable to need to include or modify the SDP stored within the
container file with the server local name of the container file. To
avoid this, one can modify the base URI used to include a trailing
slash, e.g., "rtsp://example.com/container.mp4/". In this case, the
relative URI for the media will only need to be "trackID=2".
However, this will also mean that using "*" in the SDP will result in
the control URI including the trailing slash, i.e.,
"rtsp://example.com/container.mp4/".
Note: the usage of TrackID in the above is not a standardized
form, but one example out of several similar strings such as
TrackID, Track_ID, StreamID that is used by different server
vendors to indicate a particular piece of media inside a container
file.
D.1.2. Media Streams
The "m=" field is used to enumerate the streams. It is expected that
all the specified streams will be rendered with appropriate
synchronization. If the session is over multicast, the port number
indicated SHOULD be used for reception. The client MAY try to
override the destination port, through the Transport header. The
servers MAY allow this: the response will indicate whether or not
this is allowed. If the session is unicast, the port numbers are the
ones RECOMMENDED by the server to the client, about which receiver
ports to use; the client MUST still include its receiver ports in its
SETUP request. The client MAY ignore this recommendation. If the
server has no preference, it SHOULD set the port number value to
zero.
The "m=" lines contain information about which transport protocol,
profile, and possibly lower-layer are to be used for the media
stream. The combination of transport, profile, and lower layer, like
RTP/AVP/UDP, needs to be defined for how to be used with RTSP. The
currently defined combinations are discussed in Appendix C; further
combinations MAY be specified.
Example:
m=audio 0 RTP/AVP 31
D.1.3. Payload Type(s)
The payload type or types are specified in the "m=" line. In case
the payload type is a static payload type from RFC 3551 [RFC3551], no
other information may be required. In case it is a dynamic payload
type, the media attribute "rtpmap" is used to specify what the media
is. The "encoding name" within the "rtpmap" attribute may be one of
those specified in [RFC4856], a media type registered with IANA
according to [RFC4855], or an experimental encoding as specified in
SDP [RFC4566]). Codec-specific parameters are not specified in this
field, but rather in the "fmtp" attribute described below.
The selection of the RTP payload type numbers used may be required to
consider RTP and RTCP Multiplexing [RFC5761], if that is to be
supported by the server.
D.1.4. Format-Specific Parameters
Format-specific parameters are conveyed using the "fmtp" media
attribute. The syntax of the "fmtp" attribute is specific to the
encoding(s) to which the attribute refers. Note that some of the
format-specific parameters may be specified outside of the "fmtp"
parameters, for example, like the "ptime" attribute for most audio
encodings.
D.1.5. Directionality of Media Stream
The SDP attributes "a=sendrecv", "a=recvonly", and "a=sendonly"
provide instructions about the direction the media streams flow
within a session. When using RTSP, the SDP can be delivered to a
client using either RTSP DESCRIBE or a number of RTSP external
methods, like HTTP, FTP, and email. Based on this, the SDP applies
to how the RTSP client will see the complete session. Thus, media
streams delivered from the RTSP server to the client would be given
the "a=recvonly" attribute.
"a=recvonly" in an SDP provided to the RTSP client indicates that
media delivery will only occur in the direction from the RTSP server
to the client. SDP provided to the RTSP client that lacks any of the
directionality attributes ("a=recvonly", "a=sendonly", "a=sendrecv")
would be interpreted as having "a=sendrecv". At the time of writing,
there exists no RTSP mode suitable for media traffic in the direction
from the RTSP client to the server. Thus, all RTSP SDP SHOULD have
an "a=recvonly" attribute when using the PLAY mode defined in this
document. If future modes are defined for media in the client-to-
server direction, then usage of "a=sendonly" or "a=sendrecv" may
become suitable to indicate intended media directions.
D.1.6. Range of Presentation
The "a=range" attribute defines the total time range of the stored
session or an individual media. Live sessions that are not seekable
can be indicated as specified below; whereas the length of live
sessions can be deduced from the "t=" and "r=" SDP parameters.
The attribute is both a session- and a media-level attribute. For
presentations that contain media streams of the same duration, the
range attribute SHOULD only be used at the session level. In case of
different lengths, the range attribute MUST be given at media level
for all media and SHOULD NOT be given at the session level. If the
attribute is present at both media level and session level, the
media-level values MUST be used.
Note: usually one will specify the same length for all media, even if
there isn't media available for the full duration on all media.
However, that requires that the server accept PLAY requests within
that range.
Servers MUST take care to provide RTSP Range (see Section 18.40)
values that are consistent with what is presented in the SDP for the
content. There is no reason for non dynamic content, like media
clips provided on demand to have inconsistent values. Inconsistent
values between the SDP and the actual values for the content handled
by the server is likely to generate some failure, like 457 "Invalid
Range", in case the client uses PLAY requests with a Range header.
In case the content is dynamic in length and it is infeasible to
provide a correct value in the SDP, the server is recommended to
describe this as content that is not seekable (see below). The
server MAY override that property in the response to a PLAY request
using the correct values in the Range header.
The unit is specified first, followed by the value range. The units
and their values are as defined in Section 4.4.1, Section 4.4.2, and
Section 4.4.3 and MAY be extended with further formats. Any open-
ended range (start-), i.e., without stop range, is of unspecified
duration and MUST be considered as content that is not seekable
unless this property is overridden. Multiple instances carrying
different clock formats MAY be included at either session or media
level.
ABNF for the attribute is defined in Section 20.3.
Examples:
a=range:npt=0-34.4368
a=range:clock=19971113T211503Z-19971113T220300Z
Non-seekable stream of unknown duration:
a=range:npt=0-
D.1.7. Time of Availability
The "t=" field defines when the SDP is valid. For on-demand content,
the server SHOULD indicate a stop time value for which it guarantees
the description to be valid and a start time that is equal to or
before the time at which the DESCRIBE request was received. It MAY
also indicate start and stop times of 0, meaning that the session is
always available.
For sessions that are of live type, i.e., specific start time,
unknown stop time, likely not seekable, the "t=" and "r=" field
SHOULD be used to indicate the start time of the event. The stop
time SHOULD be given so that the live event will have ended at that
time, while still not being unnecessary far into the future.
D.1.8. Connection Information
In SDP used with RTSP, the "c=" field contains the destination
address for the media stream. If a multicast address is specified,
the client SHOULD use this address in any SETUP request as
destination address, including any additional parameters, such as
TTL. For on-demand unicast streams and some multicast streams, the
destination address MAY be specified by the client via the SETUP
request, thus overriding any specified address. To identify streams
without a fixed destination address, where the client is required to
specify a destination address, the "c=" field SHOULD be set to a null
value. For addresses of type "IP4", this value MUST be "0.0.0.0";
and for type "IP6", this value MUST be "0:0:0:0:0:0:0:0" (can also be
written as "::"), i.e., the unspecified address according to RFC 4291
[RFC4291].
D.1.9. Message Body Tag
The optional "a=mtag" attribute identifies a version of the session
description. It is opaque to the client. SETUP requests may include
this identifier in the If-Match field (see Section 18.24) to allow
session establishment only if this attribute value still corresponds
to that of the current description. The attribute value is opaque
and may contain any character allowed within SDP attribute values.
ABNF for the attribute is defined in Section 20.3.
Example:
a=mtag:"158bb3e7c7fd62ce67f12b533f06b83a"
One could argue that the "o=" field provides identical
functionality. However, it does so in a manner that would put
constraints on servers that need to support multiple session
description types other than SDP for the same piece of media
content.
D.2. Aggregate Control Not Available
If a presentation does not support aggregate control, no session-
level "a=control" attribute is specified. For an SDP with multiple
media sections specified, each section will have its own control URI
specified via the "a=control" attribute.
Example:
v=0
o=- 2890844256 2890842807 IN IP4 192.0.2.56
s=I came from a web page
e=adm@example.com
c=IN IP4 0.0.0.0
t=0 0
m=video 8002 RTP/AVP 31
a=control:rtsp://audio.example.com/movie.aud
m=audio 8004 RTP/AVP 3
a=control:rtsp://video.example.com/movie.vid
Note that the position of the control URI in the description implies
that the client establishes separate RTSP control sessions to the
servers audio.example.com and video.example.com.
It is recommended that an SDP file contain the complete media-
initialization information even if it is delivered to the media
client through non-RTSP means. This is necessary as there is no
mechanism to indicate that the client should request more detailed
media stream information via DESCRIBE.
D.3. Aggregate Control Available
In this scenario, the server has multiple streams that can be
controlled as a whole. In this case, there are both a media-level
"a=control" attribute, which is used to specify the stream URIs, and
a session-level "a=control" attribute, which is used as the Request-
URI for aggregate control. If the media-level URI is relative, it is
resolved to absolute URIs according to Appendix D.1.1 above.
Example:
C->M: DESCRIBE rtsp://example.com/movie RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Date: Wed, 23 Jan 2013 15:36:52 +0000
Expires: Wed, 23 Jan 2013 16:36:52 +0000
Content-Type: application/sdp
Content-Base: rtsp://example.com/movie/
Content-Length: 227
v=0
o=- 2890844256 2890842807 IN IP4 192.0.2.211
s=I contain
i=<more info>
e=adm@example.com
c=IN IP4 0.0.0.0
a=control:*
t=0 0
m=video 8002 RTP/AVP 31
a=control:trackID=1
m=audio 8004 RTP/AVP 3
a=control:trackID=2
In this example, the client is recommended to establish a single RTSP
session to the server, and it uses the URIs rtsp://example.com/movie/
trackID=1 and rtsp://example.com/movie/trackID=2 to set up the video
and audio streams, respectively. The URI rtsp://example.com/movie/,
which is resolved from the "*", controls the whole presentation
(movie).
A client is not required to issue SETUP requests for all streams
within an aggregate object. Servers should allow the client to ask
for only a subset of the streams.
D.4. Grouping of Media Lines in SDP
For some types of media, it is desirable to express a relationship
between various media components, for instance, for lip
synchronization or Scalable Video Codec (SVC) [RFC5583]. This
relationship is expressed on the SDP level by grouping of media
lines, as described in [RFC5888], and can be exposed to RTSP.
For RTSP, it is mainly important to know how to handle grouped media
received by means of SDP, i.e., if the media are under aggregate
control (see Appendix D.3) or if aggregate control is not available
(see Appendix D.2).
It is RECOMMENDED that grouped media are handled by aggregate
control, to give the client the ability to control either the whole
presentation or single media.
D.5. RTSP External SDP Delivery
There are some considerations that need to be made when the session
description is delivered to the client outside of RTSP, for example
via HTTP or email.
First of all, the SDP needs to contain absolute URIs, since relative
will, in most cases, not work as the delivery will not correctly
forward the base URI.
The writing of the SDP session availability information, i.e., "t="
and "r=", needs to be carefully considered. When the SDP is fetched
by the DESCRIBE method, the probability that it is valid is very
high. However, the same is much less certain for SDPs distributed
using other methods. Therefore, the publisher of the SDP should take
care to follow the recommendations about availability in the SDP
specification [RFC4566] in Section 4.2.
Appendix E. RTSP Use Cases
This appendix describes the most important and considered use cases
for RTSP. They are listed in descending order of importance in
regard to ensuring that all necessary functionality is present. This
specification only fully supports usage of the two first. Also, in
these first two cases, there are special cases or exceptions that are
not supported without extensions, e.g., the redirection of media
delivery to an address other than the controlling agent's (client's).
E.1. On-Demand Playback of Stored Content
An RTSP-capable server stores content suitable for being streamed to
a client. A client desiring playback of any of the stored content
uses RTSP to set up the media transport required to deliver the
desired content. RTSP is then used to initiate, halt, and manipulate
the actual transmission (playout) of the content. RTSP is also
required to provide the necessary description and synchronization
information for the content.
The above high-level description can be broken down into a number of
functions of which RTSP needs to be capable.
Presentation Description: Provide initialization information about
the presentation (content); for example, which media codecs are
needed for the content. Other information that is important
includes the number of media streams the presentation contains,
the transport protocols used for the media streams, and
identifiers for these media streams. This information is
required before setup of the content is possible and to
determine if the client is even capable of using the content.
This information need not be sent using RTSP; other external
protocols can be used to transmit the transport presentation
descriptions. Two good examples are the use of HTTP [RFC7230]
or email to fetch or receive presentation descriptions like SDP
[RFC4566]
Setup: Set up some or all of the media streams in a presentation.
The setup itself consists of selecting the protocol for media
transport and the necessary parameters for the protocol, like
addresses and ports.
Control of Transmission: After the necessary media streams have been
established, the client can request the server to start
transmitting the content. The client must be allowed to start
or stop the transmission of the content at arbitrary times.
The client must also be able to start the transmission at any
point in the timeline of the presentation.
Synchronization: For media-transport protocols like RTP [RFC3550],
it might be beneficial to carry synchronization information
within RTSP. This may be due to either the lack of inter-media
synchronization within the protocol itself or the potential
delay before the synchronization is established (which is the
case for RTP when using RTCP).
Termination: Terminate the established contexts.
For this use case, there are a number of assumptions about how it
works. These are:
On-Demand content: The content is stored at the server and can be
accessed at any time during a time period when it is intended
to be available.
Independent sessions: A server is capable of serving a number of
clients simultaneously, including from the same piece of
content at different points in that presentations timeline.
Unicast Transport: Content for each individual client is transmitted
to them using unicast traffic.
It is also possible to redirect the media traffic to a different
destination than that of the agent controlling the traffic. However,
allowing this without appropriate mechanisms for checking that the
destination approves of this allows for Distributed DoS (DDoS).
E.2. Unicast Distribution of Live Content
This use case is similar to the above on-demand content case (see
Appendix E.1), the difference is the nature of the content itself.
Live content is continuously distributed as it becomes available from
a source; i.e., the main difference from on-demand is that one starts
distributing content before the end of it has become available to the
server.
In many cases, the consumer of live content is only interested in
consuming what actually happens "now"; i.e., very similar to
broadcast TV. However, in this case, it is assumed that there exists
no broadcast or multicast channel to the users, and instead the
server functions as a distribution node, sending the same content to
multiple receivers, using unicast traffic between server and client.
This unicast traffic and the transport parameters are individually
negotiated for each receiving client.
Another aspect of live content is that it often has a very limited
time of availability, as it is only available for the duration of the
event the content covers. An example of such live content could be a
music concert that lasts two hours and starts at a predetermined
time. Thus, there is a need to announce when and for how long the
live content is available.
In some cases, the server providing live content may be saving some
or all of the content to allow clients to pause the stream and resume
it from the paused point, or to "rewind" and play continuously from a
point earlier than the live point. Hence, this use case does not
necessarily exclude playing from other than the live point of the
stream, playing with scales other than 1.0, etc.
E.3. On-Demand Playback Using Multicast
It is possible to use RTSP to request that media be delivered to a
multicast group. The entity setting up the session (the controller)
will then control when and what media is delivered to the group.
This use case has some potential for DoS attacks by flooding a
multicast group. Therefore, a mechanism is needed to indicate that
the group actually accepts the traffic from the RTSP server.
An open issue in this use case is how one ensures that all receivers
listening to the multicast or broadcast receives the session
presentation configuring the receivers. This specification has to
rely on an external solution to solve this issue.
E.4. Inviting an RTSP Server into a Conference
If one has an established conference or group session, it is possible
to have an RTSP server distribute media to the whole group.
Transmission to the group is simplest when controlled by a single
participant or leader of the conference. Shared control might be
possible, but would require further investigation and possibly
extensions.
This use case assumes that there exists either a multicast or a
conference focus that redistributes media to all participants.
This use case is intended to be able to handle the following
scenario: a conference leader or participant (hereafter called the
"controller") has some pre-stored content on an RTSP server that he
wants to share with the group. The controller sets up an RTSP
session at the streaming server for this content and retrieves the
session description for the content. The destination for the media
content is set to the shared multicast group or conference focus.
When desired by the controller, he/she can start and stop the
transmission of the media to the conference group.
There are several issues with this use case that are not solved by
this core specification for RTSP:
DoS: To avoid an RTSP server from being an unknowing participant in
a DoS attack, the server needs to be able to verify the
destination's acceptance of the media. Such a mechanism to
verify the approval of received media does not yet exist;
instead, only policies can be used, which can be made to work
in controlled environments.
Distributing the presentation description to all participants in the
group:
To enable a media receiver to correctly decode the content,
the media configuration information needs to be distributed
reliably to all participants. This will most likely require
support from an external protocol.
Passing control of the session: If it is desired to pass control
of the RTSP session between the participants, some support
will be required by an external protocol to exchange state
information and possibly floor control of who is controlling
the RTSP session.
E.5. Live Content Using Multicast
This use case in its simplest form does not require any use of RTSP
at all; this is what multicast conferences being announced with SAP
[RFC2974] and SDP are intended to handle. However, in use cases
where more advanced features like access control to the multicast
session are desired, RTSP could be used for session establishment.
A client desiring to join a live multicasted media session with
cryptographic (encryption) access control could use RTSP in the
following way. The source of the session announces the session and
gives all interested an RTSP URI. The client connects to the server
and requests the presentation description, allowing configuration for
reception of the media. In this step, it is possible for the client
to use secured transport and any desired level of authentication; for
example, for billing or access control. An RTSP link also allows for
load balancing between multiple servers.
If these were the only goals, they could be achieved by simply using
HTTP. However, for cases where the sender likes to keep track of
each individual receiver of a session, and possibly use the session
as a side channel for distributing key-updates or other information
on a per-receiver basis, and the full set of receivers is not known
prior to the session start, the state establishment that RTSP
provides can be beneficial. In this case, a client would establish
an RTSP session for this multicast group with the RTSP server. The
RTSP server will not transmit any media, but instead will point to
the multicast group. The client and server will be able to keep the
session alive for as long as the receiver participates in the session
thus enabling, for example, the server to push updates to the client.
This use case will most likely not be able to be implemented without
some extensions to the server-to-client push mechanism. Here the
PLAY_NOTIFY method (see Section 13.5) with a suitable extension could
provide clear benefits.
Appendix F. Text Format for Parameters
A resource of type "text/parameters" consists of either 1) a list of
parameters (for a query) or 2) a list of parameters and associated
values (for a response or setting of the parameter). Each entry of
the list is a single line of text. Parameters are separated from
values by a colon. The parameter name MUST only use US-ASCII visible
characters while the values are UTF-8 text strings. The media type
registration form is in Section 22.16.
There is a potential interoperability issue for this format. It was
named in RFC 2326 but never defined, even if used in examples that
hint at the syntax. This format matches the purpose and its syntax
supports the examples provided. However, it goes further by allowing
UTF-8 in the value part; thus, usage of UTF-8 strings may not be
supported. However, as individual parameters are not defined, the
implementing application needs to have out-of-band agreement or using
feature tag anyway to determine if the endpoint supports the
parameters.
The ABNF [RFC5234] grammar for "text/parameters" content is:
file = *((parameter / parameter-value) CRLF)
parameter = 1*visible-except-colon
parameter-value = parameter *WSP ":" value
visible-except-colon = %x21-39 / %x3B-7E ; VCHAR - ":"
value = *(TEXT-UTF8char / WSP)
TEXT-UTF8char = <as defined in Section 20.1>
WSP = <See RFC 5234> ; Space or HTAB
VCHAR = <See RFC 5234>
CRLF = <See RFC 5234>
Appendix G. Requirements for Unreliable Transport of RTSP
This appendix provides guidance for those who want to implement RTSP
messages over unreliable transports as has been defined in RTSP 1.0
[RFC2326]. RFC 2326 defined the "rtspu" URI scheme and provided some
basic information for the transport of RTSP messages over UDP. The
information is being provided here as there has been at least one
commercial implementation and compatibility with that should be
maintained.
The following points should be considered for an interoperable
implementation:
o Requests shall be acknowledged by the receiver. If there is no
acknowledgement, the sender may resend the same message after a
timeout of one round-trip time (RTT). Any retransmissions due to
lack of acknowledgement must carry the same sequence number as the
original request.
o The RTT can be estimated as in TCP (RFC 6298) [RFC6298], with an
initial round-trip value of 500 ms. An implementation may cache
the last RTT measurement as the initial value for future
connections.
o The Timestamp header (Section 18.53) is used to avoid the
retransmission ambiguity problem [Stevens98].
o The registered default port for RTSP over UDP for the server is
554.
o RTSP messages can be carried over any lower-layer transport
protocol that is 8-bit clean.
o RTSP messages are vulnerable to bit errors and should not be
subjected to them.
o Source authentication, or at least validation that RTSP messages
comes from the same entity becomes extremely important, as session
hijacking may be substantially easier for RTSP message transport
using an unreliable protocol like UDP than for TCP.
There are two RTSP headers that are primarily intended for being used
by the unreliable handling of RTSP messages and which will be
maintained:
o CSeq: See Section 18.20. It should be noted that the CSeq header
is also required to match requests and responses independent
whether a reliable or unreliable transport is used.
o Timestamp: See Section 18.53
Appendix H. Backwards-Compatibility Considerations
This section contains notes on issues about backwards compatibility
with clients or servers being implemented according to RFC 2326
[RFC2326]. Note that there exists no requirement to implement RTSP
1.0; in fact, this document recommends against it as it is difficult
to do in an interoperable way.
A server implementing RTSP 2.0 MUST include an RTSP-Version of
"RTSP/2.0" in all responses to requests containing RTSP-Version value
of "RTSP/2.0". If a server receives an RTSP 1.0 request, it MAY
respond with an RTSP 1.0 response if it chooses to support RFC 2326.
If the server chooses not to support RFC 2326, it MUST respond with a
505 (RTSP Version Not Supported) status code. A server MUST NOT
respond to an RTSP 1.0 request with an RTSP 2.0 response.
Clients implementing RTSP 2.0 MAY use an OPTIONS request with an
RTSP-Version of "RTSP/2.0" to determine whether a server supports
RTSP 2.0. If the server responds with either an RTSP-Version of
"RTSP/1.0" or a status code of 505 (RTSP Version Not Supported), the
client will have to use RTSP 1.0 requests if it chooses to support
RFC 2326.
H.1. Play Request in Play State
The behavior in the server when a Play is received in Play state has
changed (Section 13.4). In RFC 2326, the new PLAY request would be
queued until the current Play completed. Any new PLAY request now
takes effect immediately replacing the previous request.
H.2. Using Persistent Connections
Some server implementations of RFC 2326 maintain a one-to-one
relationship between a connection and an RTSP session. Such
implementations require clients to use a persistent connection to
communicate with the server and when a client closes its connection,
the server may remove the RTSP session. This is worth noting if an
RTSP 2.0 client also supporting 1.0 connects to a 1.0 server.
Appendix I. Changes
This appendix briefly lists the differences between RTSP 1.0
[RFC2326] and RTSP 2.0 for an informational purpose. For
implementers of RTSP 2.0, it is recommended to read carefully through
this memo and not to rely on the list of changes below to adapt from
RTSP 1.0 to RTSP 2.0, as RTSP 2.0 is not intended to be backwards
compatible with RTSP 1.0 [RFC2326] other than the version negotiation
mechanism.
I.1. Brief Overview
The following protocol elements were removed in RTSP 2.0 compared to
RTSP 1.0:
o the RECORD and ANNOUNCE methods and all related functionality
(including 201 (Created) and 250 (Low On Storage Space) status
codes);
o the use of UDP for RTSP message transport (due to missing interest
and to broken specification);
o the use of PLAY method for keep-alive in Play state.
The following protocol elements were added or changed in RTSP 2.0
compared to RTSP 1.0:
o RTSP session TEARDOWN from the server to the client;
o IPv6 support;
o extended IANA registries (e.g., transport headers parameters,
transport-protocol, profile, lower-transport, and mode);
o request pipelining for quick session start-up;
o fully reworked state machine;
o RTSP messages now use URIs rather than URLs;
o incorporated much of related HTTP text ([RFC2616]) in this memo,
compared to just referencing the sections in HTTP, to avoid
ambiguities;
o the REDIRECT method was expanded and diversified for different
situations;
o Includes a new section about how to set up different media-
transport alternatives and their profiles in addition to lower-
layer protocols. This caused the appendix on RTP interaction to
be moved to the new section instead of being in the part that
describes RTP. The section also includes guidelines what to
consider when writing usage guidelines for new protocols and
profiles;
o Added an asynchronous notification method PLAY_NOTIFY. This
method is used by the RTSP server to asynchronously notify clients
about session changes while in Play state. To a limited extent,
this is comparable with some implementations of ANNOUNCE in RTSP
1.0 not intended for Recording.
I.2. Detailed List of Changes
The below changes have been made to RTSP 1.0 (RFC 2326) when defining
RTSP 2.0. Note that this list does not reflect minor changes in
wording or correction of typographical errors.
o The section on minimal implementation was deleted. Instead, the
main part of the specification defines the core of RTSP 2.0.
o The Transport header has been changed in the following ways:
* The ABNF has been changed to define that extensions are
possible and that unknown parameters result in servers ignoring
the transport specification.
* To prevent backwards compatibility issues, any extension or new
parameter requires the usage of a feature tag combined with the
Require header.
* Syntax ambiguities with the Mode parameter have been resolved.
* Syntax error with ";" for multicast and unicast has been
resolved.
* Two new addressing parameters have been defined: src_addr and
dest_addr. These replace the parameters "port", "client_port",
"server_port", "destination", and "source".
* Support for IPv6 explicit addresses in all address fields has
been included.
* To handle URI definitions that contain ";" or ",", a quoted-URI
format has been introduced and is required.
* IANA registries for the transport header parameters, transport-
protocol, profile, lower-transport, and mode have been defined.
* The Transport header's interleaved parameter's text was made
more strict and uses formal requirements levels. It was also
clarified that the interleaved channels are symmetric and that
it is the server that sets the channel numbers.
* It has been clarified that the client can't request of the
server to use a certain RTP SSRC, using a request with the
transport parameter SSRC.
* Syntax definition for SSRC has been clarified to require 8HEX.
It has also been extended to allow multiple values for clients
supporting this version.
* Clarified the text on the Transport header's "dest_addr"
parameters regarding what security precautions the server is
required to perform.
o The Range formats have been changed in the following way:
* The NPT format has been given an initial NPT identifier that
must now be used.
* All formats now support initial open-ended formats of type
"npt=-10" and also format only "Range: smpte" ranges for usage
with GET_PARAMETER requests.
* The npt-hhmmss notation now follows ISO 8601 more strictly.
o RTSP message handling has been changed in the following ways:
* RTSP messages now use URIs rather than URLs.
* It has been clarified that a 4xx message due to a missing CSeq
header shall be returned without a CSeq header.
* The 300 (Multiple Choices) response code has been removed.
* Rules for how to handle the timing out RTSP messages have been
added.
* Extended Pipelining rules allowing for quick session startup.
* Sequence numbering and proxy handling of sequence numbers have
been defined, including cases when responses arrive out of
order.
o The HTTP references have been updated to first RFCs 2616 and 2617
and then to RFC 7230-7235. Most of the text has been copied and
then altered to fit RTSP into this specification. The Public and
the Content-Base headers have also been imported from RFC 2068 so
that they are defined in the RTSP specification. Known effects on
RTSP due to HTTP clarifications:
* Content-Encoding header can include encoding of type
"identity".
o The state machine section has been completely rewritten. It now
includes more details and is also more clear about the model used.
o An IANA section has been included that contains a number of
registries and their rules. This will allow us to use IANA to
keep track of RTSP extensions.
o The transport of RTSP messages has seen the following changes:
* The use of UDP for RTSP message transport has been deprecated
due to missing interest and to broken specification.
* The rules for how TCP connections are to be handled have been
clarified. Now it is made clear that servers should not close
the TCP connection unless they have been unused for significant
time.
* Strong recommendations why servers and clients should use
persistent connections have also been added.
* There is now a requirement on the servers to handle non-
persistent connections as this provides fault tolerance.
* Added wording on the usage of Connection:Close for RTSP.
* Specified usage of TLS for RTSP messages, including a scheme to
approve a proxy's TLS connection to the next hop.
o The following header-related changes have been made:
* Accept-Ranges response-header has been added. This header
clarifies which range formats can be used for a resource.
* Fixed the missing definitions for the Cache-Control header.
Also added to the syntax definition the missing delta-seconds
for max-stale and min-fresh parameters.
* Put requirement on CSeq header that the value is increased by
one for each new RTSP request. A recommendation to start at 0
has also been added.
* Added a requirement that the Date header must be used for all
messages with a message body and the Server should always
include it.
* Removed the possibility of using Range header with Scale header
to indicate when it is to be activated, since it can't work as
defined. Also, added a rule that lack of Scale header in a
response indicates lack of support for the header. feature
tags for scaled playback have been defined.
* The Speed header must now be responded to in order to indicate
support and the actual speed going to be used. A feature tag
is defined. Notes on congestion control were also added.
* The Supported header was borrowed from SIP [RFC3261] to help
with the feature negotiation in RTSP.
* Clarified that the Timestamp header can be used to resolve
retransmission ambiguities.
* The Session header text has been expanded with an explanation
on keep-alive and which methods to use. SET_PARAMETER is now
recommended to use if only keep-alive within RTSP is desired.
* It has been clarified how the Range header formats are used to
indicate pause points in the PAUSE response.
* Clarified that RTP-Info URIs that are relative use the Request-
URI as base URI. Also clarified that the used URI must be the
one that was used in the SETUP request. The URIs are now also
required to be quoted. The header also expresses the SSRC for
the provided RTP timestamp and sequence number values.
* Added text that requires the Range to always be present in PLAY
responses. Clarified what should be sent in case of live
streams.
* The headers table has been updated using a structure borrowed
from SIP. Those tables convey much more information and should
provide a good overview of the available headers.
* It has been clarified that any message with a message body is
required to have a Content-Length header. This was the case in
RFC 2326, but could be misinterpreted.
* ETag has changed its name to MTag.
* To resolve functionality around MTag, the MTag and If-None-
Match header have been added from HTTP with necessary
clarification in regard to RTSP operation.
* Imported the Public header from HTTP (RFC 2068 [RFC2068]) since
it has been removed from HTTP due to lack of use. Public is
used quite frequently in RTSP.
* Clarified rules for populating the Public header so that it is
an intersection of the capabilities of all the RTSP agents in a
chain.
* Added the Media-Range header for listing the current
availability of the media range.
* Added the Notify-Reason header for giving the reason when
sending PLAY_NOTIFY requests.
* A new header Seek-Style has been defined to direct and inform
how any seek operation should/have been performed.
o The Protocol Syntax has been changed in the following way:
* All ABNF definitions are updated according to the rules defined
in RFC 5234 [RFC5234] and have been gathered in a separate
section (Section 20).
* The ABNF for the User-Agent and Server headers have been
corrected.
* Some definitions in the introduction regarding the RTSP session
have been changed.
* The protocol has been made fully IPv6 capable.
* The CHAR rule has been changed to exclude NULL.
o The Status codes have been changed in the following ways:
* The use of status code 303 (See Other) has been deprecated as
it does not make sense to use in RTSP.
* The never-defined status code 411 "Length Required" has been
completely removed.
* When sending response 451 (Parameter Not Understood) and 458
(Parameter Is Read-Only), the response body should contain the
offending parameters.
* Clarification on when a 3rr redirect status code can be
received has been added. This includes receiving 3rr as a
result of a request within an established session. This
provides clarification to a previous unspecified behavior.
* Removed the 201 (Created) and 250 (Low On Storage Space) status
codes as they are only relevant to recording, which is
deprecated.
* Several new status codes have been defined: 464 (Data Transport
Not Ready Yet), 465 (Notification Reason Unknown), 470
(Connection Authorization Required), 471 (Connection
Credentials Not Accepted), and 472 (Failure to Establish Secure
Connection).
o The following functionality has been deprecated from the protocol:
* The use of Queued Play.
* The use of PLAY method for keep-alive in Play state.
* The RECORD and ANNOUNCE methods and all related functionality.
Some of the syntax has been removed.
* The possibility to use timed execution of methods with the time
parameter in the Range header.
* The description on how rtspu works is not part of the core
specification and will require external description. Only that
it exists is mentioned here and some requirements for the
transport are provided.
o The following changes have been made in relation to methods:
* The OPTIONS method has been clarified with regard to the use of
the Public and Allow headers.
* Added text clarifying the usage of SET_PARAMETER for keep-alive
and usage without a body.
* PLAY method is now allowed to be pipelined with the pipelining
of one or more SETUP requests following the initial that
generates the session for aggregated control.
* REDIRECT has been expanded and diversified for different
situations.
* Added a new method PLAY_NOTIFY. This method is used by the
RTSP server to asynchronously notify clients about session
changes.
o Wrote a new section about how to set up different media-transport
alternatives and their profiles as well as lower-layer protocols.
This caused the appendix on RTP interaction to be moved to the new
section instead of being in the part that describes RTP. The new
section also includes guidelines what to consider when writing
usage guidelines for new protocols and profiles.
o Setup and usage of independent TCP connections for transport of
RTP has been specified.
o Added a new section describing the available mechanisms to
determine if functionality is supported, called "Capability
Handling". Renamed option-tags to feature tags.
o Added a Contributors section with people who have contributed
actual text to the specification.
o Added a section "Use Cases" that describes the major use cases for
RTSP.
o Clarified the usage of a=range and how to indicate live content
that are not seekable with this header.
o Text specifying the special behavior of PLAY for live content.
o Security features of RTSP have been clarified:
* HTTP-based authorization has been clarified requiring both
Basic and Digest support
* TLS support has been mandated
* If one implements RTP, then SRTP and defined MIKEY-based key-
exchange must be supported
* Various minor mitigations discussed or resulted in protocol
changes.
Acknowledgements
This memorandum defines RTSP version 2.0, which is a revision of the
Proposed Standard RTSP version 1.0 defined in [RFC2326]. The authors
of RFC 2326 are Henning Schulzrinne, Anup Rao, and Robert Lanphier.
Both RTSP version 1.0 and RTSP version 2.0 borrow format and
descriptions from HTTP/1.1.
Robert Sparks and especially Elwyn Davies provided very valuable and
detailed reviews in the IETF Last Call that greatly improved the
document and resolved many issues, especially regarding consistency.
This document has benefited greatly from the comments of all those
participating in the MMUSIC WG. In addition to those already
mentioned, the following individuals have contributed to this
specification:
Rahul Agarwal, Claudio Allocchio, Jeff Ayars, Milko Boic, Torsten
Braun, Brent Browning, Bruce Butterfield, Steve Casner, Maureen
Chesire, Jinhang Choi, Francisco Cortes, Elwyn Davies, Spencer
Dawkins, Kelly Djahandari, Martin Dunsmuir, Adrian Farrel, Stephen
Farrell, Ross Finlayson, Eric Fleischman, Jay Geagan, Andy Grignon,
Christian Groves, V. Guruprasad, Peter Haight, Mark Handley, Brad
Hefta-Gaub, Volker Hilt, John K. Ho, Patrick Hoffman, Go Hori,
Philipp Hoschka, Anne Jones, Ingemar Johansson, Jae-Hwan Kim, Anders
Klemets, Ruth Lang, Barry Leiba, Stephanie Leif, Jonathan Lennox,
Eduardo F. Llach, Chris Lonvick, Xavier Marjou, Thomas Marshall, Rob
McCool, Martti Mela, David Oran, Joerg Ott, Joe Pallas, Maria
Papadopouli, Sujal Patel, Ema Patki, Alagu Periyannan, Colin Perkins,
Pekka Pessi, Igor Plotnikov, Pete Resnick, Peter Saint-Andre, Holger
Schmidt, Jonathan Sergent, Pinaki Shah, David Singer, Lior Sion, Jeff
Smith, Alexander Sokolsky, Dale Stammen, John Francis Stracke, Geetha
Srikantan, Scott Taylor, David Walker, Stephan Wenger, Dale R.
Worley, and Byungjo Yoon, and especially Flemming Andreasen.
Contributors
The following people have made written contributions that were
included in the specification:
o Tom Marshall contributed text on the usage of 3rr status codes.
o Thomas Zheng contributed text on the usage of the Range in PLAY
responses and proposed an earlier version of the PLAY_NOTIFY
method.
o Sean Sheedy contributed text on the timeout behavior of RTSP
messages and connections, the 463 (Destination Prohibited) status
code, and proposed an earlier version of the PLAY_NOTIFY method.
o Greg Sherwood proposed an earlier version of the PLAY_NOTIFY
method.
o Fredrik Lindholm contributed text about the RTSP security
framework.
o John Lazzaro contributed the text for RTP over Independent TCP.
o Aravind Narasimhan contributed by rewriting "Media-Transport
Alternatives" (Appendix C) and making editorial improvements on a
number of places in the specification.
o Torbjorn Einarsson has done some editorial improvements of the
text.
Authors' Addresses
Henning Schulzrinne
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
United States of America
Email: schulzrinne@cs.columbia.edu
Anup Rao
Cisco
United States of America
Email: anrao@cisco.com
Rob Lanphier
San Francisco, CA
United States of America
Email: robla@robla.net
Magnus Westerlund
Ericsson
Faeroegatan 2
Stockholm SE-164 80
Sweden
Email: magnus.westerlund@ericsson.com
Martin Stiemerling (editor)
University of Applied Sciences Darmstadt
Haardtring 100
64295 Darmstadt
Germany
Email: mls.ietf@gmail.com
URI: http://www.stiemerling.org