| Rfc | 7866 |
|---|
| Title | Session Recording Protocol |
|---|
| Author | L. Portman, H. Lum, Ed., C. Eckel, A.
Johnston, A. Hutton |
|---|
| Date | May 2016 |
|---|
| Format: | TXT, HTML |
|---|
| Status: | PROPOSED
STANDARD |
|---|
|
Internet Engineering Task Force (IETF) L. Portman
Request for Comments: 7866 NICE Systems
Category: Standards Track H. Lum, Ed.
ISSN: 2070-1721 Genesys
C. Eckel
Cisco
A. Johnston
Illinois Institute of Technology
A. Hutton
Unify
May 2016
Session Recording Protocol
Abstract
This document specifies the use of the Session Initiation Protocol
(SIP), the Session Description Protocol (SDP), and the Real-time
Transport Protocol (RTP) for delivering real-time media and metadata
from a Communication Session (CS) to a recording device. The Session
Recording Protocol specifies the use of SIP, SDP, and RTP to
establish a Recording Session (RS) between the Session Recording
Client (SRC), which is on the path of the CS, and a Session Recording
Server (SRS) at the recording device. This document considers only
active recording, where the SRC purposefully streams media to an SRS
and all participating user agents (UAs) are notified of the
recording. Passive recording, where a recording device detects media
directly from the network (e.g., using port-mirroring techniques), is
outside the scope of this document. In addition, lawful intercept is
outside the scope of this document.
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/rfc7866.
Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction ....................................................4
2. Terminology .....................................................4
3. Definitions .....................................................4
4. Scope ...........................................................4
5. Overview of Operations ..........................................5
5.1. Delivering Recorded Media ..................................5
5.2. Delivering Recording Metadata ..............................8
5.3. Receiving Recording Indications and Providing Recording
Preferences ................................................9
6. SIP Handling ...................................................11
6.1. Procedures at the SRC .....................................11
6.1.1. Initiating a Recording Session .....................11
6.1.2. SIP Extensions for Recording Indications
and Preferences ....................................12
6.2. Procedures at the SRS .....................................12
6.3. Procedures for Recording-Aware User Agents ................12
7. SDP Handling ...................................................13
7.1. Procedures at the SRC .....................................13
7.1.1. SDP Handling in the RS .............................13
7.1.1.1. Handling Media Stream Updates .............14
7.1.2. Recording Indication in the CS .....................15
7.1.3. Recording Preference in the CS .....................16
7.2. Procedures at the SRS .....................................16
7.3. Procedures for Recording-Aware User Agents ................18
7.3.1. Recording Indication ...............................18
7.3.2. Recording Preference ...............................19
8. RTP Handling ...................................................20
8.1. RTP Mechanisms ............................................20
8.1.1. RTCP ...............................................20
8.1.2. RTP Profile ........................................21
8.1.3. SSRC ...............................................21
8.1.4. CSRC ...............................................22
8.1.5. SDES ...............................................22
8.1.5.1. CNAME .....................................22
8.1.6. Keepalive ..........................................22
8.1.7. RTCP Feedback Messages .............................23
8.1.7.1. Full Intra Request ........................23
8.1.7.2. Picture Loss Indication ...................23
8.1.7.3. Temporary Maximum Media Stream Bit
Rate Request ..............................24
8.1.8. Symmetric RTP/RTCP for Sending and Receiving .......24
8.2. Roles .....................................................25
8.2.1. SRC Acting as an RTP Translator ....................26
8.2.1.1. Forwarding Translator .....................26
8.2.1.2. Transcoding Translator ....................26
8.2.2. SRC Acting as an RTP Mixer .........................27
8.2.3. SRC Acting as an RTP Endpoint ......................28
8.3. RTP Session Usage by SRC ..................................28
8.3.1. SRC Using Multiple m-lines .........................28
8.3.2. SRC Using Mixing ...................................29
8.4. RTP Session Usage by SRS ..................................30
9. Metadata .......................................................31
9.1. Procedures at the SRC .....................................31
9.2. Procedures at the SRS .....................................33
10. Persistent Recording ..........................................35
11. IANA Considerations ...........................................36
11.1. Registration of Option Tags ..............................36
11.1.1. "siprec" Option Tag ...............................36
11.1.2. "record-aware" Option Tag .........................36
11.2. Registration of Media Feature Tags .......................36
11.2.1. Feature Tag for the SRC ...........................36
11.2.2. Feature Tag for the SRS ...........................37
11.3. New Content-Disposition Parameter Registrations ..........37
11.4. SDP Attributes ...........................................38
11.4.1. "record" SDP Attribute ............................38
11.4.2. "recordpref" SDP Attribute ........................38
12. Security Considerations .......................................39
12.1. Authentication and Authorization .........................39
12.2. RTP Handling .............................................40
12.3. Metadata .................................................41
12.4. Storage and Playback .....................................41
13. References ....................................................41
13.1. Normative References .....................................41
13.2. Informative References ...................................42
Acknowledgements ..................................................44
Authors' Addresses ................................................45
1. Introduction
This document specifies the mechanism to record a Communication
Session (CS) by delivering real-time media and metadata from the CS
to a recording device. In accordance with the architecture
[RFC7245], the Session Recording Protocol specifies the use of SIP,
the Session Description Protocol (SDP), and RTP to establish a
Recording Session (RS) between the Session Recording Client (SRC),
which is on the path of the CS, and a Session Recording Server (SRS)
at the recording device. SIP is also used to deliver metadata to the
recording device, as specified in [RFC7865]. Metadata is information
that describes recorded media and the CS to which they relate. The
Session Recording Protocol intends to satisfy the SIP-based Media
Recording (SIPREC) requirements listed in [RFC6341]. In addition to
the Session Recording Protocol, this document specifies extensions
for user agents (UAs) that are participants in a CS to receive
recording indications and to provide preferences for recording.
This document considers only active recording, where the SRC
purposefully streams media to an SRS and all participating UAs are
notified of the recording. Passive recording, where a recording
device detects media directly from the network (e.g., using
port-mirroring techniques), is outside the scope of this document.
In addition, lawful intercept is outside the scope of this document,
in accordance with [RFC2804].
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
3. Definitions
This document refers to the core definitions provided in the
architecture document [RFC7245].
Section 8 uses the definitions provided in "RTP: A Transport Protocol
for Real-Time Applications" [RFC3550].
4. Scope
The scope of the Session Recording Protocol includes the
establishment of the RSs and the reporting of the metadata. The
scope also includes extensions supported by UAs participating in the
CS, such as an indication of recording. The UAs need not be
recording aware in order to participate in a CS being recorded.
The items in the following list, which is not exhaustive, do not
represent the protocol itself and are considered out of scope for the
Session Recording Protocol:
o Delivering recorded media in real time as the CS media
o Specifications of criteria to select a specific CS to be recorded
or triggers to record a certain CS in the future
o Recording policies that determine whether the CS should be
recorded and whether parts of the CS are to be recorded
o Retention policies that determine how long a recording is stored
o Searching and accessing the recorded media and metadata
o Policies governing how CS users are made aware of recording
o Delivering additional RS metadata through a non-SIP mechanism
5. Overview of Operations
This section is informative and provides a description of recording
operations.
Section 6 describes the SIP communication in an RS between an SRC and
an SRS, as well as the procedures for recording-aware UAs
participating in a CS. Section 7 describes SDP handling in an RS,
and the procedures for recording indications and recording
preferences. Section 8 describes RTP handling in an RS. Section 9
describes the mechanism to deliver recording metadata from the SRC to
the SRS.
As mentioned in the architecture document [RFC7245], there are a
number of types of call flows based on the location of the SRC. The
sample call flows discussed in Section 5.1 provide a quick overview
of the operations between the SRC and the SRS.
5.1. Delivering Recorded Media
When a SIP Back-to-Back User Agent (B2BUA) with SRC functionality
routes a call from UA A to UA B, the SRC has access to the media path
between the UAs. When the SRC is aware that it should be recording
the conversation, the SRC can cause the B2BUA to relay the media
between UA A and UA B. The SRC then establishes the RS with the SRS
and sends replicated media towards the SRS.
An endpoint may also have SRC functionality, where the endpoint
itself establishes the RS to the SRS. Since the endpoint has access
to the media in the CS, the endpoint can send replicated media
towards the SRS.
The example call flows in Figures 1 and 2 show an SRC establishing an
RS towards an SRS. Figure 1 illustrates UA A acting as the SRC.
Figure 2 illustrates a B2BUA acting as the SRC. Note that the SRC
can choose when to establish the RS independent of the CS, even
though the example call flows suggest that the SRC is establishing
the RS (message (5) in Figure 2) after the CS is established.
UA A/SRC UA B SRS
|(1) CS INVITE | |
|---------------------->| |
| (2) 200 OK | |
|<----------------------| |
| | |
|(3) RS INVITE with SDP | |
|--------------------------------------------->|
| | (4) 200 OK with SDP |
|<---------------------------------------------|
|(5) CS RTP | |
|======================>| |
|<======================| |
|(6) RS RTP | |
|=============================================>|
|=============================================>|
| | |
|(7) CS BYE | |
|---------------------->| |
|(8) RS BYE | |
|--------------------------------------------->|
| | |
Figure 1: Basic Recording Call Flow with UA as SRC
UA A SRC UA B SRS
|(1) CS INVITE | | |
|------------->| | |
| |(2) CS INVITE | |
| |---------------------->| |
| | (3) 200 OK | |
| |<----------------------| |
| (4) 200 OK | | |
|<-------------| | |
| |(5) RS INVITE with SDP | |
| |--------------------------------------------->|
| | | (6) 200 OK with SDP |
| |<---------------------------------------------|
|(7) CS RTP | | |
|=============>|======================>| |
|<=============|<======================| |
| |(8) RS RTP | |
| |=============================================>|
| |=============================================>|
|(9) CS BYE | | |
|------------->| | |
| |(10) CS BYE | |
| |---------------------->| |
| |(11) RS BYE | |
| |--------------------------------------------->|
| | | |
Figure 2: Basic Recording Call Flow with B2BUA as SRC
The call flow shown in Figure 2 can also apply to the case of a
centralized conference with a mixer. For clarity, ACKs to INVITEs
and 200 OKs to BYEs are not shown. The conference focus can provide
the SRC functionality, since the conference focus has access to all
the media from each conference participant. When a recording is
requested, the SRC delivers the metadata and the media streams to the
SRS. Since the conference focus has access to a mixer, the SRC may
choose to mix the media streams from all participants as a single
mixed media stream towards the SRS.
An SRC can use a single RS to record multiple CSs. Every time the
SRC wants to record a new call, the SRC updates the RS with a new SDP
offer to add new recorded streams to the RS and to correspondingly
also update the metadata for the new call.
An SRS can also establish an RS to an SRC, although it is beyond the
scope of this document to define how an SRS would specify which calls
to record.
5.2. Delivering Recording Metadata
The SRC is responsible for the delivery of metadata to the SRS. The
SRC may provide an initial metadata snapshot about recorded media
streams in the initial INVITE content in the RS. Subsequent metadata
updates can be represented as a stream of events in UPDATE [RFC3311]
or re-INVITE requests sent by the SRC. These metadata updates are
normally incremental updates to the initial metadata snapshot to
optimize on the size of updates. However, the SRC may also decide to
send a new metadata snapshot at any time.
Metadata is transported in the body of INVITE or UPDATE messages.
Certain metadata, such as the attributes of the recorded media
stream, is located in the SDP of the RS.
The SRS has the ability to send a request to the SRC to ask for a new
metadata snapshot update from the SRC. This can happen when the SRS
fails to understand the current stream of incremental updates for
whatever reason -- for example, when the SRS loses the current state
due to internal failure. The SRS may optionally attach a reason
along with the snapshot request. This request allows both the SRC
and the SRS to synchronize the states with a new metadata snapshot so
that further incremental metadata updates will be based on the latest
metadata snapshot. Similar to the metadata content, the metadata
snapshot request is transported as content in UPDATE or INVITE
messages sent by the SRS in the RS.
SRC SRS
| |
|(1) INVITE (metadata snapshot 1) |
|---------------------------------------------------->|
| (2) 200 OK |
|<----------------------------------------------------|
|(3) ACK |
|---------------------------------------------------->|
|(4) RTP |
|====================================================>|
|====================================================>|
|(5) UPDATE (metadata update 1) |
|---------------------------------------------------->|
| (6) 200 OK |
|<----------------------------------------------------|
|(7) UPDATE (metadata update 2) |
|---------------------------------------------------->|
| (8) 200 OK |
|<----------------------------------------------------|
| (9) UPDATE (metadata snapshot request) |
|<----------------------------------------------------|
| (10) 200 OK |
|---------------------------------------------------->|
| (11) INVITE (metadata snapshot 2 + SDP offer) |
|---------------------------------------------------->|
| (12) 200 OK (SDP answer) |
|<----------------------------------------------------|
| (13) UPDATE (metadata update 1 based on snapshot 2) |
|---------------------------------------------------->|
| (14) 200 OK |
|<----------------------------------------------------|
Figure 3: Delivering Metadata via SIP UPDATE
5.3. Receiving Recording Indications and Providing Recording
Preferences
The SRC is responsible for providing recording indications to the
participants in the CS. A recording-aware UA supports receiving
recording indications via the SDP "a=record" attribute, and it can
specify a recording preference in the CS by including the SDP
"a=recordpref" attribute. The recording attribute is a declaration
by the SRC in the CS to indicate whether recording is taking place.
The recording preference attribute is a declaration by the recording-
aware UA in the CS to indicate its recording preference. A UA that
does not want to be recorded may still be notified that recording is
occurring, for a number of reasons (e.g., it was not capable of
indicating its preference, its preference was ignored). If this
occurs, the UA's only mechanism to avoid being recorded is to
terminate its participation in the session.
To illustrate how the attributes are used, if UA A is initiating a
call to UA B and UA A is also an SRC that is performing the
recording, then UA A provides the recording indication in the SDP
offer with a=record:on. Since UA A is the SRC, UA A receives the
recording indication from the SRC directly. When UA B receives the
SDP offer, UA B will see that recording is happening on the other
endpoint of this session. Since UA B is not an SRC and does not
provide any recording preference, the SDP answer does not contain
a=record or a=recordpref.
UA A UA B
(SRC) |
| |
| [SRC recording starts] |
|(1) INVITE (SDP offer + a=record:on) |
|---------------------------------------------------->|
| (2) 200 OK (SDP answer) |
|<----------------------------------------------------|
|(3) ACK |
|---------------------------------------------------->|
|(4) RTP |
|<===================================================>|
| |
| [UA B wants to set preference to no recording] |
| (5) INVITE (SDP offer + a=recordpref:off) |
|<----------------------------------------------------|
| [SRC honors the preference and stops recording] |
|(6) 200 OK (SDP answer + a=record:off) |
|---------------------------------------------------->|
| (7) ACK |
|<----------------------------------------------------|
Figure 4: Recording Indication and Recording Preference
After the call is established and recording is in progress, UA B
later decides to change the recording preference to no recording and
sends a re-INVITE with the "a=recordpref" attribute. It is up to the
SRC to honor the preference, and in this case the SRC decides to stop
the recording and updates the recording indication in the SDP answer.
Note that UA B could have explicitly indicated a recording preference
in (2), the 200 OK for the original INVITE. Indicating a preference
of no recording in an initial INVITE or an initial response to an
INVITE may reduce the chance of a user being recorded in the
first place.
6. SIP Handling
6.1. Procedures at the SRC
6.1.1. Initiating a Recording Session
An RS is a SIP session with specific extensions applied, and these
extensions are listed in the procedures below for the SRC and the
SRS. When an SRC or an SRS receives a SIP session that is not an RS,
it is up to the SRC or the SRS to determine what to do with the SIP
session.
The SRC can initiate an RS by sending a SIP INVITE request to the
SRS. The SRC and the SRS are identified in the From and To headers,
respectively.
The SRC MUST include the "+sip.src" feature tag in the Contact URI,
defined in this specification as an extension to [RFC3840], for all
RSs. An SRS uses the presence of the "+sip.src" feature tag in
dialog creating and modifying requests and responses to confirm that
the dialog being created is for the purpose of an RS. In addition,
when an SRC sends a REGISTER request to a registrar, the SRC MAY
include the "+sip.src" feature tag to indicate that it is an SRC.
Since SIP Caller Preferences extensions are optional to implement for
routing proxies, there is no guarantee that an RS will be routed to
an SRC or SRS. A new option tag, "siprec", is introduced. As per
[RFC3261], only an SRC or an SRS can accept this option tag in an RS.
An SRC MUST include the "siprec" option tag in the Require header
when initiating an RS so that UAs that do not support the Session
Recording Protocol extensions will simply reject the INVITE request
with a 420 (Bad Extension) response.
When an SRC receives a new INVITE, the SRC MUST only consider the SIP
session as an RS when both the "+sip.srs" feature tag and the
"siprec" option tag are included in the INVITE request.
6.1.2. SIP Extensions for Recording Indications and Preferences
For the CS, the SRC MUST provide recording indications to all
participants in the CS. A participant UA in a CS can indicate that
it is recording aware by providing the "record-aware" option tag, and
the SRC MUST provide recording indications in the new SDP "a=record"
attribute described in Section 7 below. In the absence of the
"record-aware" option tag -- meaning that the participant UA is not
recording aware -- an SRC MUST provide recording indications through
other means, such as playing a tone in-band or having a signed
participant contract in place.
An SRC in the CS may also indicate itself as a session recording
client by including the "+sip.src" feature tag. A recording-aware
participant can learn that an SRC is in the CS and can set the
recording preference for the CS with the new SDP "a=recordpref"
attribute described in Section 7.
6.2. Procedures at the SRS
When an SRS receives a new INVITE, the SRS MUST only consider the SIP
session as an RS when both the "+sip.src" feature tag and the
"siprec" option tag are included in the INVITE request.
The SRS can initiate an RS by sending a SIP INVITE request to the
SRC. The SRS and the SRC are identified in the From and To headers,
respectively.
The SRS MUST include the "+sip.srs" feature tag in the Contact URI,
as per [RFC3840], for all RSs. An SRC uses the presence of this
feature tag in dialog creation and modification requests and
responses to confirm that the dialog being created is for the purpose
of an RS (REQ-030 in [RFC6341]). In addition, when an SRS sends a
REGISTER request to a registrar, the SRS SHOULD include the
"+sip.srs" feature tag to indicate that it is an SRS.
An SRS MUST include the "siprec" option tag in the Require header as
per [RFC3261] when initiating an RS so that UAs that do not support
the Session Recording Protocol extensions will simply reject the
INVITE request with a 420 (Bad Extension) response.
6.3. Procedures for Recording-Aware User Agents
A recording-aware UA is a participant in the CS that supports the SIP
and SDP extensions for receiving recording indications and for
requesting recording preferences for the call. A recording-aware UA
MUST indicate that it can accept the reporting of recording
indications provided by the SRC with a new "record-aware" option tag
when initiating or establishing a CS; this means including the
"record-aware" option tag in the Supported header in the initial
INVITE request or response.
A recording-aware UA MUST provide a recording indication to the end
user through an appropriate user interface, indicating whether
recording is on, off, or paused for each medium. Appropriate user
interfaces may include real-time notification or previously
established agreements that use of the device is subject to
recording. Some UAs that are automatons (e.g., Interactive Voice
Response (IVR), media server, Public Switched Telephone Network
(PSTN) gateway) may not have a user interface to render a recording
indication. When such a UA indicates recording awareness, the UA
SHOULD render the recording indication through other means, such as
passing an in-band tone on the PSTN gateway, putting the recording
indication in a log file, or raising an application event in a
VoiceXML dialog. These UAs MAY also choose not to indicate recording
awareness, thereby relying on whatever mechanism an SRC chooses to
indicate recording, such as playing a tone in-band.
7. SDP Handling
7.1. Procedures at the SRC
The SRC and SRS follow the SDP offer/answer model described in
[RFC3264]. The procedures for the SRC and SRS describe the
conventions used in an RS.
7.1.1. SDP Handling in the RS
Since the SRC does not expect to receive media from the SRS, the SRC
typically sets each media stream of the SDP offer to only send media,
by qualifying them with the "a=sendonly" attribute, according to the
procedures in [RFC3264].
The SRC sends recorded streams of participants to the SRS, and the
SRC MUST provide a "label" attribute ("a=label"), as per [RFC4574],
on each media stream in order to identify the recorded stream with
the rest of the metadata. The "a=label" attribute identifies each
recorded media stream, and the label name is mapped to the Media
Stream Reference in the metadata as per [RFC7865]. The scope of the
"a=label" attribute only applies to the SDP and metadata conveyed in
the bodies of the SIP request or response that the label appeared in.
Note that a recorded stream is distinct from a CS stream; the
metadata provides a list of participants that contribute to each
recorded stream.
Figure 5 shows an example SDP offer from an SRC with both audio and
video recorded streams. Note that this example contains unfolded
lines longer than 72 characters; these lines are captured between
<allOneLine> tags.
v=0
o=SRC 2890844526 2890844526 IN IP4 198.51.100.1
s=-
c=IN IP4 198.51.100.1
t=0 0
m=audio 12240 RTP/AVP 0 4 8
a=sendonly
a=label:1
m=video 22456 RTP/AVP 98
a=rtpmap:98 H264/90000
<allOneLine>
a=fmtp:98 profile-level-id=42A01E;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
</allOneLine>
a=sendonly
a=label:2
m=audio 12242 RTP/AVP 0 4 8
a=sendonly
a=label:3
m=video 22458 RTP/AVP 98
a=rtpmap:98 H264/90000
<allOneLine>
a=fmtp:98 profile-level-id=42A01E;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
</allOneLine>
a=sendonly
a=label:4
Figure 5: Sample SDP Offer from SRC with Audio and Video Streams
7.1.1.1. Handling Media Stream Updates
Over the lifetime of an RS, the SRC can add and remove recorded
streams to and from the RS for various reasons -- for example, when a
CS stream is added to or removed from the CS, or when a CS is created
or terminated if an RS handles multiple CSs. To remove a recorded
stream from the RS, the SRC sends a new SDP offer where the port of
the media stream to be removed is set to zero, according to the
procedures in [RFC3264]. To add a recorded stream to the RS, the SRC
sends a new SDP offer by adding a new media stream description or by
reusing an old media stream that had been previously disabled,
according to the procedures in [RFC3264].
The SRC can temporarily discontinue streaming and collection of
recorded media from the SRC to the SRS for reasons such as masking
the recording. In this case, the SRC sends a new SDP offer and sets
the media stream to inactive (a=inactive) for each recorded stream to
be paused, as per the procedures in [RFC3264]. To resume streaming
and collection of recorded media, the SRC sends a new SDP offer and
sets the media stream to sendonly (a=sendonly). Note that a CS may
itself change the media stream direction by updating the SDP -- for
example, by setting a=inactive for SDP hold. Media stream direction
changes in the CS are conveyed in the metadata by the SRC. When a CS
media stream is changed to or from inactive, the effect on the
corresponding RS media stream is governed by SRC policy. The SRC MAY
have a local policy to pause an RS media stream when the
corresponding CS media stream is inactive, or it MAY leave the RS
media stream as sendonly.
7.1.2. Recording Indication in the CS
While there are existing mechanisms for providing an indication that
a CS is being recorded, these mechanisms are usually delivered on the
CS media streams, such as playing an in-band tone or an announcement
to the participants. A new "record" SDP attribute is introduced to
allow the SRC to indicate recording state to a recording-aware UA in
a CS.
The "record" SDP attribute appears at the media level or
session level in either an SDP offer or answer. When the attribute
is applied at the session level, the indication applies to all media
streams in the SDP. When the attribute is applied at the
media level, the indication applies to that one media stream only,
and that overrides the indication if also set at the session level.
Whenever the recording indication needs to change, such as
termination of recording, the SRC MUST initiate a re-INVITE or UPDATE
to update the SDP "a=record" attribute.
The following is the ABNF [RFC5234] of the "record" attribute:
attribute =/ record-attr
; attribute defined in RFC 4566
record-attr = "record:" indication
indication = "on" / "off" / "paused"
on: Recording is in progress.
off: No recording is in progress.
paused: Recording is in progress but media is paused.
7.1.3. Recording Preference in the CS
When the SRC receives the "a=recordpref" SDP in an SDP offer or
answer, the SRC chooses to honor the preference to record based on
local policy at the SRC. If the SRC makes a change in recording
state, the SRC MUST report the new recording state in the "a=record"
attribute in the SDP answer or in a subsequent SDP offer.
7.2. Procedures at the SRS
Typically, the SRS only receives RTP streams from the SRC; therefore,
the SDP offer/answer from the SRS normally sets each media stream to
receive media, by setting them with the "a=recvonly" attribute,
according to the procedures of [RFC3264]. When the SRS is not ready
to receive a recorded stream, the SRS sets the media stream as
inactive in the SDP offer or answer by setting it with an
"a=inactive" attribute, according to the procedures of [RFC3264].
When the SRS is ready to receive recorded streams, the SRS sends a
new SDP offer and sets the media streams with an "a=recvonly"
attribute.
Figure 6 shows an example of an SDP answer from the SRS for the SDP
offer from Figure 5. Note that this example contains unfolded lines
longer than 72 characters; these lines are captured between
<allOneLine> tags.
v=0
o=SRS 0 0 IN IP4 198.51.100.20
s=-
c=IN IP4 198.51.100.20
t=0 0
m=audio 10000 RTP/AVP 0
a=recvonly
a=label:1
m=video 10002 RTP/AVP 98
a=rtpmap:98 H264/90000
<allOneLine>
a=fmtp:98 profile-level-id=42A01E;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
</allOneLine>
a=recvonly
a=label:2
m=audio 10004 RTP/AVP 0
a=recvonly
a=label:3
m=video 10006 RTP/AVP 98
a=rtpmap:98 H264/90000
<allOneLine>
a=fmtp:98 profile-level-id=42A01E;
sprop-parameter-sets=Z0IACpZTBYmI,aMljiA==
</allOneLine>
a=recvonly
a=label:4
Figure 6: Sample SDP Answer from SRS with Audio and Video Streams
Over the lifetime of an RS, the SRS can remove recorded streams from
the RS for various reasons. To remove a recorded stream from the RS,
the SRS sends a new SDP offer where the port of the media stream to
be removed is set to zero, according to the procedures in [RFC3264].
The SRS MUST NOT add recorded streams in the RS when the SRS sends a
new SDP offer. Similarly, when the SRS starts an RS, the SRS MUST
initiate the INVITE without an SDP offer to let the SRC generate the
SDP offer with the streams to be recorded.
The sequence diagram in Figure 7 shows an example where the SRS is
initially not ready to receive recorded streams and later updates the
RS when the SRS is ready to record.
SRC SRS
| |
|(1) INVITE (SDP offer) |
|---------------------------------------------------->|
| [not ready to record]
| (2) 200 OK with SDP inactive |
|<----------------------------------------------------|
|(3) ACK |
|---------------------------------------------------->|
| ... |
| [ready to record]
| (4) re-INVITE with SDP recvonly |
|<----------------------------------------------------|
|(5) 200 OK with SDP sendonly |
|---------------------------------------------------->|
| (6) ACK |
|<----------------------------------------------------|
|(7) RTP |
|====================================================>|
| ... |
|(8) BYE |
|---------------------------------------------------->|
| (9) OK |
|<----------------------------------------------------|
Figure 7: SRS Responding to Offer with a=inactive
7.3. Procedures for Recording-Aware User Agents
7.3.1. Recording Indication
When a recording-aware UA receives an SDP offer or answer that
includes the "a=record" attribute, the UA provides to the end user an
indication as to whether the recording is on, off, or paused for each
medium, based on the most recently received "a=record" SDP attribute
for that medium.
When a CS is traversed through multiple UAs such as a B2BUA or a
conference focus, each UA involved in the CS that is aware that the
CS is being recorded MUST provide the recording indication through
the "a=record" attribute to all other parties in the CS.
It is possible that more than one SRC is in the call path of the same
CS, but the recording indication attribute does not provide any hint
as to which SRC or how many SRCs are recording. An endpoint knows
only that the call is being recorded. Furthermore, this attribute is
not used as a request for a specific SRC to start or stop recording.
7.3.2. Recording Preference
A participant in a CS MAY set the recording preference in the CS to
be recorded or not recorded at session establishment or during the
session. A new "recordpref" SDP attribute is introduced, and the
participant in the CS may set this recording preference attribute in
any SDP offer/answer at session establishment time or during the
session. The SRC is not required to honor the recording preference
from a participant, based on local policies at the SRC, and the
participant can learn the recording indication through the "a=record"
SDP attribute as described in Section 7.3.1.
The SDP "a=recordpref" attribute can appear at the media level or
session level and can appear in an SDP offer or answer. When the
attribute is applied at the session level, the recording preference
applies to all media streams in the SDP. When the attribute is
applied at the media level, the recording preference applies to that
one media stream only, and that overrides the recording preference if
also set at the session level. The UA can change the recording
preference by changing the "a=recordpref" attribute in a subsequent
SDP offer or answer. The absence of the "a=recordpref" attribute in
the SDP indicates that the UA has no recording preference.
The following is the ABNF of the "recordpref" attribute:
attribute =/ recordpref-attr
; attribute defined in RFC 4566
recordpref-attr = "a=recordpref:" pref
pref = "on" / "off" / "pause" / "nopreference"
on: Sets the preference to record if it has not already been
started. If the recording is currently paused, the
preference is to resume recording.
off: Sets the preference for no recording. If recording has
already been started, then the preference is to stop the
recording.
pause: If the recording is currently in progress, sets the
preference to pause the recording.
nopreference:
Indicates that the UA has no preference regarding recording.
8. RTP Handling
This section provides recommendations and guidelines for RTP and the
Real-time Transport Control Protocol (RTCP) in the context of SIPREC
[RFC6341]. In order to communicate most effectively, the SRC, the
SRS, and any recording-aware UAs should utilize the mechanisms
provided by RTP in a well-defined and predictable manner. It is the
goal of this document to make the reader aware of these mechanisms
and to provide recommendations and guidelines.
8.1. RTP Mechanisms
This section briefly describes important RTP/RTCP constructs and
mechanisms that are particularly useful within the context of SIPREC.
8.1.1. RTCP
The RTP data transport is augmented by a control protocol (RTCP) to
allow monitoring of the data delivery. RTCP, as defined in
[RFC3550], is based on the periodic transmission of control packets
to all participants in the RTP session, using the same distribution
mechanism as the data packets. Support for RTCP is REQUIRED, per
[RFC3550], and it provides, among other things, the following
important functionality in relation to SIPREC:
1) Feedback on the quality of the data distribution
This feedback from the receivers may be used to diagnose faults in
the distribution. As such, RTCP is a well-defined and efficient
mechanism for the SRS to inform the SRC, and for the SRC to inform
recording-aware UAs, of issues that arise with respect to the
reception of media that is to be recorded.
2) Including a persistent transport-level identifier -- the CNAME, or
canonical name -- for an RTP source
The synchronization source (SSRC) [RFC3550] identifier may change
if a conflict is discovered or a program is restarted, in which
case receivers can use the CNAME to keep track of each
participant. Receivers may also use the CNAME to associate
multiple data streams from a given participant in a set of related
RTP sessions -- for example, to synchronize audio and video.
Synchronization of media streams is also facilitated by the NTP
and RTP timestamps included in RTCP packets by data senders.
8.1.2. RTP Profile
The RECOMMENDED RTP profiles for the SRC, SRS, and recording-aware
UAs are "Extended Secure RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/SAVPF)" [RFC5124] when using
encrypted RTP streams, and "Extended RTP Profile for Real-time
Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)"
[RFC4585] when using non-encrypted media streams. However, as these
are not requirements, some implementations may use "The Secure
Real-time Transport Protocol (SRTP)" [RFC3711] and "RTP Profile for
Audio and Video Conferences with Minimal Control" [RFC3551].
Therefore, it is RECOMMENDED that the SRC, SRS, and recording-aware
UAs not rely entirely on RTP/SAVPF or RTP/AVPF for core functionality
that may be at least partially achievable using RTP/SAVP and RTP/AVP.
AVPF and SAVPF provide an improved RTCP timer model that allows more
flexible transmission of RTCP packets in response to events, rather
than strictly according to bandwidth. AVPF-based codec control
messages provide efficient mechanisms for an SRC, an SRS, and
recording-aware UAs to handle events such as scene changes, error
recovery, and dynamic bandwidth adjustments. These messages are
discussed in more detail later in this document.
SAVP and SAVPF provide media encryption, integrity protection, replay
protection, and a limited form of source authentication. They do not
contain or require a specific keying mechanism.
8.1.3. SSRC
The SSRC, as defined in [RFC3550], is carried in the RTP header and
in various fields of RTCP packets. It is a random 32-bit number that
is required to be globally unique within an RTP session. It is
crucial that the number be chosen with care, in order that
participants on the same network or starting at the same time are not
likely to choose the same number. Guidelines regarding SSRC value
selection and conflict resolution are provided in [RFC3550].
The SSRC may also be used to separate different sources of media
within a single RTP session. For this reason, as well as for
conflict resolution, it is important that the SRC, SRS, and
recording-aware UAs handle changes in SSRC values and properly
identify the reason for the change. The CNAME values carried in RTCP
facilitate this identification.
8.1.4. CSRC
The contributing source (CSRC), as defined in [RFC3550], identifies
the source of a stream of RTP packets that has contributed to the
combined stream produced by an RTP mixer. The mixer inserts a list
of the SSRC identifiers of the sources that contributed to the
generation of a particular packet into the RTP header of that packet.
This list is called the CSRC list. It is RECOMMENDED that an SRC or
recording-aware UA, when acting as a mixer, set the CSRC list
accordingly, and that the SRC and SRS interpret the CSRC list per
[RFC3550] when received.
8.1.5. SDES
The Source Description (SDES), as defined in [RFC3550], contains an
SSRC/CSRC identifier followed by a list of zero or more items that
carry information about the SSRC/CSRC. End systems send one SDES
packet containing their own source identifier (the same as the SSRC
in the fixed RTP header). A mixer sends one SDES packet containing a
chunk for each CSRC from which it is receiving SDES information, or
multiple complete SDES packets if there are more than 31 such
sources.
The ability to identify individual CSRCs is important in the context
of SIPREC. Metadata [RFC7865] provides a mechanism to achieve this
at the signaling level. SDES provides a mechanism at the RTP level.
8.1.5.1. CNAME
The Canonical End-Point Identifier (CNAME), as defined in [RFC3550],
provides the binding from the SSRC identifier to an identifier for
the source (sender or receiver) that remains constant. It is
important that the SRC and recording-aware UAs generate CNAMEs
appropriately and that the SRC and SRS interpret and use them for
this purpose. Guidelines for generating CNAME values are provided in
"Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names
(CNAMEs)" [RFC7022].
8.1.6. Keepalive
It is anticipated that media streams in SIPREC may exist in an
inactive state for extended periods of time for any of a number of
valid reasons. In order for the bindings and any pinholes in
NATs/firewalls to remain active during such intervals, it is
RECOMMENDED that the SRC, SRS, and recording-aware UAs follow the
keepalive procedure recommended in "Application Mechanism for Keeping
Alive the NAT Mappings Associated with RTP / RTP Control Protocol
(RTCP) Flows" [RFC6263] for all RTP media streams.
8.1.7. RTCP Feedback Messages
"Codec Control Messages in the RTP Audio-Visual Profile with Feedback
(AVPF)" [RFC5104] specifies extensions to the messages defined in
AVPF [RFC4585]. Support for and proper usage of these messages are
important to SRC, SRS, and recording-aware UA implementations. Note
that these messages are applicable only when using the AVPF or SAVPF
RTP profiles.
8.1.7.1. Full Intra Request
A Full Intra Request (FIR) command, when received by the designated
media sender, requires that the media sender send a decoder refresh
point at the earliest opportunity. Using a decoder refresh point
implies refraining from using any picture sent prior to that point as
a reference for the encoding process of any subsequent picture sent
in the stream.
Decoder refresh points, especially Intra or Instantaneous Decoding
Refresh (IDR) pictures for H.264 video codecs, are in general several
times larger in size than predicted pictures. Thus, in scenarios in
which the available bit rate is small, the use of a decoder refresh
point implies a delay that is significantly longer than the typical
picture duration.
8.1.7.1.1. Deprecated Usage of SIP INFO Instead of FIR
"XML Schema for Media Control" [RFC5168] defines an Extensible Markup
Language (XML) Schema for video fast update. Implementations are
discouraged from using the method described in [RFC5168], except for
purposes of backward compatibility. Implementations SHOULD use FIR
messages instead.
To make sure that a common mechanism exists between the SRC and SRS,
the SRS MUST support both mechanisms (FIR and SIP INFO), using FIR
messages when negotiated successfully with the SRC and using SIP INFO
otherwise.
8.1.7.2. Picture Loss Indication
Picture Loss Indication (PLI), as defined in [RFC4585], informs the
encoder of the loss of an undefined amount of coded video data
belonging to one or more pictures. [RFC4585] recommends using PLI
instead of FIR messages to recover from errors. FIR is appropriate
only in situations where not sending a decoder refresh point would
render the video unusable for the users. Examples where sending FIR
messages is appropriate include a multipoint conference when a new
user joins the conference and no regular decoder refresh point
interval is established, and a video-switching Multipoint Control
Unit (MCU) that changes streams.
Appropriate use of PLI and FIR is important to ensure, with minimum
overhead, that the recorded video is usable (e.g., the necessary
reference frames exist for a player to render the recorded video).
8.1.7.3. Temporary Maximum Media Stream Bit Rate Request
A receiver, translator, or mixer uses the Temporary Maximum Media
Stream Bit Rate Request (TMMBR) [RFC5104] to request a sender to
limit the maximum bit rate for a media stream to the provided value.
Appropriate use of TMMBR facilitates rapid adaptation to changes in
available bandwidth.
8.1.7.3.1. Renegotiation of SDP Bandwidth Attribute
If it is likely that the new value indicated by TMMBR will be valid
for the remainder of the session, the TMMBR sender is expected to
perform a renegotiation of the session upper limit using the session
signaling protocol. Therefore, for SIPREC, implementations are
RECOMMENDED to use TMMBR for temporary changes and renegotiation of
bandwidth via SDP offer/answer for more permanent changes.
8.1.8. Symmetric RTP/RTCP for Sending and Receiving
Within an SDP offer/answer exchange, RTP entities choose the RTP and
RTCP transport addresses (i.e., IP addresses and port numbers) on
which to receive packets. When sending packets, the RTP entities may
use the same source port or a different source port than those
signaled for receiving packets. When the transport address used to
send and receive RTP is the same, it is termed "symmetric RTP"
[RFC4961]. Likewise, when the transport address used to send and
receive RTCP is the same, it is termed "symmetric RTCP" [RFC4961].
When sending RTP, the use of symmetric RTP is REQUIRED. When sending
RTCP, the use of symmetric RTCP is REQUIRED. Although an SRS will
not normally send RTP, it will send RTCP as well as receive RTP and
RTCP. Likewise, although an SRC will not normally receive RTP from
the SRS, it will receive RTCP as well as send RTP and RTCP.
Note: Symmetric RTP and symmetric RTCP are different from RTP/RTCP
multiplexing [RFC5761].
8.2. Roles
An SRC has the task of gathering media from the various UAs in one or
more CSs and forwarding the information to the SRS within the context
of a corresponding RS. There are numerous ways in which an SRC may
do this, including, but not limited to, appearing as a UA within a
CS, or as a B2BUA between UAs within a CS.
(Recording Session) +---------+
+------------SIP------->| |
| +------RTP/RTCP----->| SRS |
| | +-- Metadata -->| |
| | | +---------+
v v |
+---------+
| SRC |
|---------| (Communication Session) +---------+
| |<----------SIP---------->| |
| UA-A | | UA-B |
| |<-------RTP/RTCP-------->| |
+---------+ +---------+
Figure 8: UA as SRC
(Recording Session) +---------+
+------------SIP------->| |
| +------RTP/RTCP----->| SRS |
| | +-- Metadata -->| |
| | | +---------+
v v |
+---------+
| SRC |
+---------+ |---------| +---------+
| |<----SIP----->| |<----SIP----->| |
| UA-A | | B2BUA | | UA-B |
| |<--RTP/RTCP-->| |<--RTP/RTCP-->| |
+---------+ +---------+ +---------+
|_______________________________________________|
(Communication Session)
Figure 9: B2BUA as SRC
The following subsections define a set of roles an SRC may choose to
play, based on its position with respect to a UA within a CS, and an
SRS within an RS. A CS and a corresponding RS are independent
sessions; therefore, an SRC may play a different role within a CS
than it does within the corresponding RS.
8.2.1. SRC Acting as an RTP Translator
The SRC may act as a translator, as defined in [RFC3550]. A defining
characteristic of a translator is that it forwards RTP packets with
their SSRC identifier intact. There are two types of translators:
one that simply forwards, and another that performs transcoding
(e.g., from one codec to another) in addition to forwarding.
8.2.1.1. Forwarding Translator
When acting as a forwarding translator, RTP received as separate
streams from different sources (e.g., from different UAs with
different SSRCs) cannot be mixed by the SRC and MUST be sent
separately to the SRS. All RTCP reports MUST be passed by the SRC
between the UAs and the SRS, such that the UAs and SRS are able to
detect any SSRC collisions.
RTCP Sender Reports generated by a UA sending a stream MUST be
forwarded to the SRS. RTCP Receiver Reports generated by the SRS
MUST be forwarded to the relevant UA.
UAs may receive multiple sets of RTCP Receiver Reports -- one or more
from other UAs participating in the CS, and one from the SRS
participating in the RS. A UA SHOULD process the RTCP Receiver
Reports from the SRS if it is recording aware.
If SRTP is used on both the CS and the RS, decryption and/or
re-encryption may occur. For example, if different keys are used, it
will occur. If the same keys are used, it need not occur.
Section 12 provides additional information on SRTP and keying
mechanisms.
If packet loss occurs, either from the UA to the SRC or from the SRC
to the SRS, the SRS SHOULD detect and attempt to recover from the
loss. The SRC does not play a role in this, other than forwarding
the associated RTP and RTCP packets.
8.2.1.2. Transcoding Translator
When acting as a transcoding translator, an SRC MAY perform
transcoding (e.g., from one codec to another), and this may result in
a different rate of packets between what the SRC receives on the CS
and what the SRC sends on the RS. As when acting as a forwarding
translator, RTP received as separate streams from different sources
(e.g., from different UAs with different SSRCs) cannot be mixed by
the SRC and MUST be sent separately to the SRS. All RTCP reports
MUST be passed by the SRC between the UAs and the SRS, such that the
UAs and SRS are able to detect any SSRC collisions.
RTCP Sender Reports generated by a UA sending a stream MUST be
forwarded to the SRS. RTCP Receiver Reports generated by the SRS
MUST be forwarded to the relevant UA. The SRC may need to manipulate
the RTCP Receiver Reports to take into account any transcoding that
has taken place.
UAs may receive multiple sets of RTCP Receiver Reports -- one or more
from other UAs participating in the CS, and one from the SRS
participating in the RS. A recording-aware UA SHOULD be prepared to
process the RTCP Receiver Reports from the SRS, whereas a recording-
unaware UA may discard such RTCP packets as irrelevant.
If SRTP is used on both the CS and the RS, decryption and/or
re-encryption may occur. For example, if different keys are used, it
will occur. If the same keys are used, it need not occur.
Section 12 provides additional information on SRTP and keying
mechanisms.
If packet loss occurs, either from the UA to the SRC or from the SRC
to the SRS, the SRS SHOULD detect and attempt to recover from the
loss. The SRC does not play a role in this, other than forwarding
the associated RTP and RTCP packets.
8.2.2. SRC Acting as an RTP Mixer
In the case of the SRC acting as an RTP mixer, as defined in
[RFC3550], the SRC combines RTP streams from different UAs and sends
them towards the SRS using its own SSRC. The SSRCs from the
contributing UA SHOULD be conveyed as CSRC identifiers within this
stream. The SRC may make timing adjustments among the received
streams and generate its own timing on the stream sent to the SRS.
Optionally, an SRC acting as a mixer can perform transcoding and can
even cope with different codings received from different UAs. RTCP
Sender Reports and Receiver Reports are not forwarded by an SRC
acting as a mixer, but there are requirements for forwarding RTCP
Source Description (SDES) packets. The SRC generates its own RTCP
Sender Reports and Receiver Reports toward the associated UAs
and SRS.
The use of SRTP between the SRC and the SRS for the RS is independent
of the use of SRTP between the UAs and the SRC for the CS.
Section 12 provides additional information on SRTP and keying
mechanisms.
If packet loss occurs from the UA to the SRC, the SRC SHOULD detect
and attempt to recover from the loss. If packet loss occurs from
the SRC to the SRS, the SRS SHOULD detect and attempt to recover from
the loss.
8.2.3. SRC Acting as an RTP Endpoint
The case of the SRC acting as an RTP endpoint, as defined in
[RFC3550], is similar to the mixer case, except that the RTP session
between the SRC and the SRS is considered completely independent from
the RTP session that is part of the CS. The SRC can, but need not,
mix RTP streams from different participants prior to sending to the
SRS. RTCP between the SRC and the SRS is completely independent of
RTCP on the CS.
The use of SRTP between the SRC and the SRS for the RS is independent
of the use of SRTP between the UAs and SRC for the CS. Section 12
provides additional information on SRTP and keying mechanisms.
If packet loss occurs from the UA to the SRC, the SRC SHOULD detect
and attempt to recover from the loss. If packet loss occurs from
the SRC to the SRS, the SRS SHOULD detect and attempt to recover from
the loss.
8.3. RTP Session Usage by SRC
There are multiple ways that an SRC may choose to deliver recorded
media to an SRS. In some cases, it may use a single RTP session for
all media within the RS, whereas in others it may use multiple RTP
sessions. The following subsections provide examples of basic RTP
session usage by the SRC, including a discussion of how the RTP
constructs and mechanisms covered previously are used. An SRC may
choose to use one or more of the RTP session usages within a single
RS. For the purpose of base interoperability between SRC and SRS, an
SRC MUST support separate m-lines in SDP, one per CS media direction.
The set of RTP session usages described is not meant to be
exhaustive.
8.3.1. SRC Using Multiple m-lines
When using multiple m-lines, an SRC includes each m-line in an SDP
offer to the SRS. The SDP answer from the SRS MUST include all
m-lines, with any rejected m-lines indicated with a zero port, per
[RFC3264]. Having received the answer, the SRC starts sending media
to the SRS as indicated in the answer. Alternatively, if the SRC
deems the level of support indicated in the answer to be
unacceptable, it may initiate another SDP offer/answer exchange in
which an alternative RTP session usage is negotiated.
In order to preserve the mapping of media to participant within the
CSs in the RS, the SRC SHOULD map each unique CNAME within the CSs to
a unique CNAME within the RS. Additionally, the SRC SHOULD map each
unique combination of CNAME/SSRC within the CSs to a unique
CNAME/SSRC within the RS. In doing so, the SRC may act as an
RTP translator or as an RTP endpoint.
Figure 10 illustrates a case in which each UA represents a
participant contributing two RTP sessions (e.g., one for audio and
one for video), each with a single SSRC. The SRC acts as an RTP
translator and delivers the media to the SRS using four RTP sessions,
each with a single SSRC. The CNAME and SSRC values used by the UAs
within their media streams are preserved in the media streams from
the SRC to the SRS.
+---------+
+------------SSRC Aa--->| |
| + --------SSRC Av--->| |
| | +------SSRC Ba--->| SRS |
| | | +---SSRC Bv--->| |
| | | | +---------+
| | | |
| | | |
+---------+ +----------+ +---------+
| |---SSRC Aa-->| SRC |<--SSRC Ba---| |
| UA-A | |(CNAME-A, | | UA-B |
|(CNAME-A)|---SSRC Av-->| CNAME-B) |<--SSRC Bv---|(CNAME-B)|
+---------+ +----------+ +---------+
Figure 10: SRC Using Multiple m-lines
8.3.2. SRC Using Mixing
When using mixing, the SRC combines RTP streams from different
participants and sends them towards the SRS using its own SSRC. The
SSRCs from the contributing participants SHOULD be conveyed as CSRC
identifiers. The SRC includes one m-line for each RTP session in an
SDP offer to the SRS. The SDP answer from the SRS MUST include all
m-lines, with any rejected m-lines indicated with a zero port, per
[RFC3264]. Having received the answer, the SRC starts sending media
to the SRS as indicated in the answer.
In order to preserve the mapping of media to participant within the
CSs in the RS, the SRC SHOULD map each unique CNAME within the CSs to
a unique CNAME within the RS. Additionally, the SRC SHOULD map each
unique combination of CNAME/SSRC within the CSs to a unique
CNAME/SSRC within the RS. The SRC MUST avoid SSRC collisions,
rewriting SSRCs if necessary when used as CSRCs in the RS. In
doing so, the SRC acts as an RTP mixer.
In the event that the SRS does not support this usage of CSRC values,
it relies entirely on the SIPREC metadata to determine the
participants included within each mixed stream.
Figure 11 illustrates a case in which each UA represents a
participant contributing two RTP sessions (e.g., one for audio and
one for video), each with a single SSRC. The SRC acts as an RTP
mixer and delivers the media to the SRS using two RTP sessions,
mixing media from each participant into a single RTP session
containing a single SSRC and two CSRCs.
SSRC Sa +---------+
+-------CSRC Aa,Ba--->| |
| | |
| SSRC Sv | SRS |
| +---CSRC Av,Bv--->| |
| | +---------+
| |
+----------+
+---------+ | SRC | +---------+
| |---SSRC Aa-->|(CNAME-S, |<--SSRC Ba---| |
| UA-A | | CNAME-A, | | UA-B |
|(CNAME-A)|---SSRC Av-->| CNAME-B) |<--SSRC Bv---|(CNAME-B)|
+---------+ +----------+ +---------+
Figure 11: SRC Using Mixing
8.4. RTP Session Usage by SRS
An SRS that supports recording an audio CS MUST support SRC usage of
separate audio m-lines in SDP, one per CS media direction. An SRS
that supports recording a video CS MUST support SRC usage of separate
video m-lines in SDP, one per CS media direction. Therefore, for an
SRS supporting a typical audio call, the SRS has to support receiving
at least two audio m-lines. For an SRS supporting a typical audio
and video call, the SRS has to support receiving at least four total
m-lines in the SDP -- two audio m-lines and two video m-lines.
These requirements allow an SRS to be implemented that supports video
only, without requiring support for audio recording. They also allow
an SRS to be implemented that supports recording only one direction
of one stream in a CS -- for example, an SRS designed to record
security monitoring cameras that only send (not receive) video
without any audio. These requirements were not written to prevent
other modes from being implemented and used, such as using a single
m-line and mixing the separate audio streams together. Rather, the
requirements were written to provide a common base mode to implement
for the sake of interoperability. It is important to note that an
SRS implementation supporting the common base mode may not record all
media streams in a CS if a participant supports more than one m-line
in a video call, such as one for camera and one for presentation.
SRS implementations may support other modes as well, but they have to
at least support the modes discussed above, such that they
interoperate in the common base mode for basic interoperability.
9. Metadata
Some metadata attributes are contained in SDP, and others are
contained in a new content type called "application/rs-metadata".
The format of the metadata is described as part of the mechanism in
[RFC7865]. A new "disposition-type" of Content-Disposition is
defined for the purpose of carrying metadata. The value is
"recording-session", which indicates that the
"application/rs-metadata" content contains metadata to be handled by
the SRS.
9.1. Procedures at the SRC
The SRC MUST send metadata to the SRS in an RS. The SRC SHOULD send
metadata as soon as it becomes available and whenever it changes.
Cases in which an SRC may be justified in waiting temporarily before
sending metadata include:
o waiting for a previous metadata exchange to complete (i.e., the
SRC cannot send another SDP offer until the previous offer/answer
completes and may also prefer not to send an UPDATE during this
time).
o constraining the signaling rate on the RS.
o sending metadata when key events occur, rather than for every
event that has any impact on metadata.
The SRC may also be configured to suppress certain metadata out of
concern for privacy or perceived lack of need for it to be included
in the recording.
Metadata sent by the SRC is categorized as either a full metadata
snapshot or a partial update. A full metadata snapshot describes all
metadata associated with the RS. The SRC MAY send a full metadata
snapshot at any time. The SRC MAY send a partial update only if a
full metadata snapshot has been sent previously.
The SRC MAY send metadata (either a full metadata snapshot or a
partial update) in an INVITE request, an UPDATE request [RFC3311], or
a 200 response to an offerless INVITE from the SRS. If the metadata
contains a reference to any SDP labels, the request containing the
metadata MUST also contain an SDP offer that defines those labels.
When a SIP message contains both an SDP offer and metadata, the
request body MUST have content type "multipart/mixed", with one
subordinate body part containing the SDP offer and another containing
the metadata. When a SIP message contains only an SDP offer or
metadata, the "multipart/mixed" container is optional.
The SRC SHOULD include a full metadata snapshot in the initial INVITE
request establishing the RS. If metadata is not yet available (e.g.,
an RS established in the absence of a CS), the SRC SHOULD send a full
metadata snapshot as soon as metadata becomes available.
If the SRC receives a snapshot request from the SRS, it MUST
immediately send a full metadata snapshot.
Figure 12 illustrates an example of a full metadata snapshot sent by
the SRC in the initial INVITE request:
INVITE sip:recorder@example.com SIP/2.0
Via: SIP/2.0/TCP src.example.com;branch=z9hG4bKdf6b622b648d9
From: <sip:2000@example.com>;tag=35e195d2-947d-4585-946f-09839247
To: <sip:recorder@example.com>
Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a
CSeq: 101 INVITE
Max-Forwards: 70
Require: siprec
Accept: application/sdp, application/rs-metadata
Contact: <sip:2000@src.example.com>;+sip.src
Content-Type: multipart/mixed;boundary=foobar
Content-Length: [length]
--foobar
Content-Type: application/sdp
v=0
o=SRS 2890844526 2890844526 IN IP4 198.51.100.1
s=-
c=IN IP4 198.51.100.1
t=0 0
m=audio 12240 RTP/AVP 0 4 8
a=sendonly
a=label:1
--foobar
Content-Type: application/rs-metadata
Content-Disposition: recording-session
[metadata content]
Figure 12: Sample INVITE Request for the Recording Session
9.2. Procedures at the SRS
The SRS receives metadata updates from the SRC in INVITE and UPDATE
requests. Since the SRC can send partial updates based on the
previous update, the SRS needs to keep track of the sequence of
updates from the SRC.
In the case of an internal failure at the SRS, the SRS may fail to
recognize a partial update from the SRC. The SRS may be able to
recover from the internal failure by requesting a full metadata
snapshot from the SRC. Certain errors, such as syntax errors or
semantic errors in the metadata information, are likely caused by an
error on the SRC side, and it is likely that the same error will
occur again even when a full metadata snapshot is requested. In
order to avoid repeating the same error, the SRS can simply terminate
the RS when a syntax error or semantic error is detected in the
metadata.
The SRS MAY explicitly request a full metadata snapshot by sending an
UPDATE request. This request MUST contain a body with
Content-Disposition type "recording-session" and MUST NOT contain an
SDP body. The SRS MUST NOT request a full metadata snapshot in an
UPDATE response or in any other SIP transaction. The format of the
content is "application/rs-metadata", and the body is an XML
document, the format of which is defined in [RFC7865]. Figure 13
shows an example:
UPDATE sip:2000@src.example.com SIP/2.0
Via: SIP/2.0/UDP srs.example.com;branch=z9hG4bKdf6b622b648d9
To: <sip:2000@example.com>;tag=35e195d2-947d-4585-946f-098392474
From: <sip:recorder@example.com>;tag=1234567890
Call-ID: d253c800-b0d1ea39-4a7dd-3f0e20a
CSeq: 1 UPDATE
Max-Forwards: 70
Require: siprec
Contact: <sip:recorder@srs.example.com>;+sip.srs
Accept: application/sdp, application/rs-metadata
Content-Disposition: recording-session
Content-Type: application/rs-metadata
Content-Length: [length]
<?xml version="1.0" encoding="UTF-8"?>
<requestsnapshot xmlns='urn:ietf:params:xml:ns:recording:1'>
<requestreason xml:lang="it">SRS internal error</requestreason>
</requestsnapshot>
Figure 13: Metadata Request
Note that UPDATE was chosen for the SRS to request a metadata
snapshot, because it can be sent regardless of the state of the
dialog. This was seen as better than requiring support for both
UPDATE and re-INVITE messages for this operation.
When the SRC receives a request for a metadata snapshot, it MUST
immediately provide a full metadata snapshot in a separate INVITE or
UPDATE transaction. Any subsequent partial updates will not be
dependent on any metadata sent prior to this full metadata snapshot.
The metadata received by the SRS can contain ID elements used to
cross-reference one element to another. An element containing the
definition of an ID and an element containing a reference to that ID
will often be received from the same SRC. It is also valid for those
elements to be received from different SRCs -- for example, when each
endpoint in the same CS acts as an SRC to record the call and a
common ID refers to the same CS. The SRS MUST NOT consider this an
error.
10. Persistent Recording
Persistent recording is a specific use case addressing REQ-005 in
[RFC6341], where an RS can be established in the absence of a CS.
The SRC continuously records media in an RS to the SRS even in the
absence of a CS for all UAs that are part of persistent recording.
By allocating recorded streams and continuously sending recorded
media to the SRS, the SRC does not have to prepare new recorded
streams with a new SDP offer when a new CS is created and also does
not impact the timing of the CS. The SRC only needs to update the
metadata when new CSs are created.
When there is no CS running on the devices with persistent recording,
there is no recorded media to stream from the SRC to the SRS. In
certain environments where a Network Address Translator (NAT) is
used, a minimum amount of flow activity is typically required to
maintain the NAT binding for each port opened. Agents that support
Interactive Connectivity Establishment (ICE) solve this problem. For
non-ICE agents, in order not to lose the NAT bindings for the
RTP/RTCP ports opened for the recorded streams, the SRC and SRS
SHOULD follow the recommendations provided in [RFC6263] to maintain
the NAT bindings.
11. IANA Considerations
11.1. Registration of Option Tags
This specification registers two option tags. The required
information for this registration, as specified in [RFC3261], is as
follows.
11.1.1. "siprec" Option Tag
Name: siprec
Description: This option tag is for identifying that the SIP session
is for the purpose of an RS. This is typically not used in a
Supported header. When present in a Require header in a request,
it indicates that the UA is either an SRC or SRS capable of
handling an RS.
11.1.2. "record-aware" Option Tag
Name: record-aware
Description: This option tag is to indicate the ability of the UA to
receive recording indicators in media-level or session-level SDP.
When present in a Supported header, it indicates that the UA can
receive recording indicators in media-level or session-level SDP.
11.2. Registration of Media Feature Tags
This document registers two new media feature tags in the SIP tree
per the process defined in [RFC2506] and [RFC3840].
11.2.1. Feature Tag for the SRC
Media feature tag name: sip.src
ASN.1 Identifier: 1.3.6.1.8.4.27
Summary of the media feature indicated by this tag: This feature tag
indicates that the UA is a Session Recording Client for the
purpose of an RS.
Values appropriate for use with this feature tag: boolean
The feature tag is intended primarily for use in the following
applications, protocols, services, or negotiation mechanisms:
This feature tag is only useful for an RS.
Examples of typical use: Routing the request to a Session Recording
Server.
Security Considerations: Security considerations for this media
feature tag are discussed in Section 11.1 of RFC 3840.
11.2.2. Feature Tag for the SRS
Media feature tag name: sip.srs
ASN.1 Identifier: 1.3.6.1.8.4.28
Summary of the media feature indicated by this tag: This feature tag
indicates that the UA is a Session Recording Server for the
purpose of an RS.
Values appropriate for use with this feature tag: boolean
The feature tag is intended primarily for use in the following
applications, protocols, services, or negotiation mechanisms:
This feature tag is only useful for an RS.
Examples of typical use: Routing the request to a Session Recording
Client.
Security Considerations: Security considerations for this media
feature tag are discussed in Section 11.1 of RFC 3840.
11.3. New Content-Disposition Parameter Registrations
This document registers a new "disposition-type" value in the
Content-Disposition header: recording-session.
recording-session: The body describes either
* metadata about the RS
or
* the reason for the metadata snapshot request
as determined by the MIME value indicated in the Content-Type.
11.4. SDP Attributes
This document registers the following new SDP attributes.
11.4.1. "record" SDP Attribute
Contact names:
Leon Portman, leon.portman@nice.com;
Henry Lum, henry.lum@genesyslab.com
Attribute name: record
Long-form attribute name: Recording Indication
Type of attribute: session level or media level
Subject to charset: no
This attribute provides the recording indication for the session or
media stream.
Allowed attribute values: on, off, paused
11.4.2. "recordpref" SDP Attribute
Contact names:
Leon Portman, leon.portman@nice.com;
Henry Lum, henry.lum@genesyslab.com
Attribute name: recordpref
Long-form attribute name: Recording Preference
Type of attribute: session level or media level
Subject to charset: no
This attribute provides the recording preference for the session or
media stream.
Allowed attribute values: on, off, pause, nopreference
12. Security Considerations
The RS is fundamentally a standard SIP dialog [RFC3261]; therefore,
the RS can reuse any of the existing SIP security mechanisms
available for securing the session signaling, the recorded media, and
the metadata. The use cases and requirements document [RFC6341]
outlines the general security considerations, and this document
describes specific security recommendations.
The SRC and SRS MUST support SIP with Transport Layer Security (TLS)
version 1.2, SHOULD follow the best practices when using TLS as per
[RFC7525], and MAY use Session Initiation Protocol Secure (SIPS) with
TLS as per [RFC5630]. The RS MUST be at least as secure as the CS;
this means using at least the same strength of cipher suite as the CS
if the CS is secured. For example, if the CS uses SIPS for signaling
and RTP/SAVP for media, then the RS may not use SIP or plain RTP
unless other equivalent security measures are in effect, since doing
so would mean an effective security downgrade. Examples of other
potentially equivalent security mechanisms include mutually
authenticated TLS for the RS signaling channel or an appropriately
protected network path for the RS media component.
12.1. Authentication and Authorization
At the transport level, the RS uses TLS authentication to validate
the authenticity of the SRC and SRS. The SRC and SRS MUST implement
TLS mutual authentication for establishing the RS. Whether the
SRC/SRS chooses to use TLS mutual authentication is a deployment
decision. In deployments where a UA acts as its own SRC, this
requires that the UA have its own certificate as needed for TLS
mutual authentication. In deployments where the SRC and the SRS are
in the same administrative domain and have some other means of
assuring authenticity, the SRC and SRS may choose not to authenticate
each other or to have the SRC authenticate the SRS only. In
deployments where the SRS can be hosted on a different administrative
domain, it is important to perform mutual authentication to ensure
the authenticity of both the SRC and the SRS before transmitting any
recorded media. The risk of not authenticating the SRS is that the
recording may be sent to an entity other than the intended SRS,
allowing a sensitive call recording to be received by an attacker.
On the other hand, the risk of not authenticating the SRC is that an
SRS will accept calls from an unknown SRC and allow potential forgery
of call recordings.
There may be scenarios in which the signaling between the SRC and SRS
is not direct, e.g., a SIP proxy exists between the SRC and the SRS.
In such scenarios, each hop is subject to the TLS mutual
authentication constraint, and transitive trust at each hop is
utilized. Additionally, an SRC or SRS may use other existing SIP
mechanisms available, including, but not limited to, Digest
authentication [RFC3261], asserted identity [RFC3325], and connected
identity [RFC4916].
The SRS may have its own set of recording policies to authorize
recording requests from the SRC. The use of recording policies is
outside the scope of the Session Recording Protocol.
12.2. RTP Handling
In many scenarios, it will be critical for the media transported
between the SRC and the SRS to be protected. Media encryption is an
important element in the overall SIPREC solution; therefore, the SRC
and the SRS MUST support RTP/SAVP [RFC3711] and RTP/SAVPF [RFC5124].
RTP/SAVP and RTP/SAVPF provide media encryption, integrity
protection, replay protection, and a limited form of source
authentication. They do not contain or require a specific keying
mechanism. At a minimum, the SRC and SRS MUST support the SDP
security descriptions key negotiation mechanism [RFC4568]. For cases
in which Datagram Transport Layer Security for Secure RTP (DTLS-SRTP)
is used to encrypt a CS media stream, an SRC may use SRTP Encrypted
Key Transport (EKT) [EKT-SRTP] in order to use SRTP-SDES in the RS
without needing to re-encrypt the media.
Note: When using EKT in this manner, it is possible for
participants in the CS to send traffic that appears to be from
other participants and have this forwarded by the SRC to the SRS
within the RS. If this is a concern (e.g., the RS is intended for
audit or compliance purposes), EKT is not an appropriate choice.
When RTP/SAVP or RTP/SAVPF is used, an SRC can choose to use the same
keys or different keys in the RS than those used in the CS. Some
SRCs are designed to simply replicate RTP packets from a CS media
stream to the SRS, in which case the SRC will use the same key in the
RS as the key used in the CS. In this case, the SRC MUST secure the
SDP containing the keying material in the RS with at least the same
level of security as in the CS. The risk of lowering the level of
security in the RS is that it will effectively become a downgrade
attack on the CS, since the same key is used for both the CS and
the RS.
SRCs that decrypt an encrypted CS media stream and re-encrypt it when
sending it to the SRS MUST use a different key than what is used for
the CS media stream, to ensure that it is not possible for someone
who has the key for the CS media stream to access recorded data they
are not authorized to access. In order to maintain a comparable
level of security, the key used in the RS SHOULD be of equivalent
strength to, or greater strength than, that used in the CS.
12.3. Metadata
Metadata contains sensitive information, such as the address of
record of the participants and other extension data placed by the
SRC. It is essential to protect the content of the metadata in the
RS. Since metadata is a content type transmitted in SIP signaling,
metadata SHOULD be protected at the transport level by SIPS/TLS.
12.4. Storage and Playback
While storage and playback of the call recording are beyond the scope
of this document, it is worthwhile to mention here that it is also
important for the recording storage and playback to provide a level
of security that is comparable to the CS. It would defeat the
purpose of securing both the CS and the RS mentioned in the previous
sections if the recording can be easily played back with a simple,
unsecured HTTP interface without any form of authentication or
authorization.
13. References
13.1. Normative References
[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>.
[RFC2506] Holtman, K., Mutz, A., and T. Hardie, "Media Feature Tag
Registration Procedure", BCP 31, RFC 2506,
DOI 10.17487/RFC2506, March 1999,
<http://www.rfc-editor.org/info/rfc2506>.
[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>.
[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>.
[RFC3840] Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
"Indicating User Agent Capabilities in the Session
Initiation Protocol (SIP)", RFC 3840,
DOI 10.17487/RFC3840, August 2004,
<http://www.rfc-editor.org/info/rfc3840>.
[RFC4574] Levin, O. and G. Camarillo, "The Session Description
Protocol (SDP) Label Attribute", RFC 4574,
DOI 10.17487/RFC4574, August 2006,
<http://www.rfc-editor.org/info/rfc4574>.
[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>.
[RFC7245] Hutton, A., Ed., Portman, L., Ed., Jain, R., and K. Rehor,
"An Architecture for Media Recording Using the Session
Initiation Protocol", RFC 7245, DOI 10.17487/RFC7245,
May 2014, <http://www.rfc-editor.org/info/rfc7245>.
[RFC7865] Ravindranath, R., Ravindran, P., and P. Kyzivat, "Session
Initiation Protocol (SIP) Recording Metadata", RFC 7865,
DOI 10.17487/RFC7865, May 2016,
<http://www.rfc-editor.org/info/rfc7865>.
13.2. Informative References
[EKT-SRTP] Mattsson, J., Ed., McGrew, D., Wing, D., and F. Andreasen,
"Encrypted Key Transport for Secure RTP", Work in
Progress, draft-ietf-avtcore-srtp-ekt-03, October 2014.
[RFC2804] IAB and IESG, "IETF Policy on Wiretapping", RFC 2804,
DOI 10.17487/RFC2804, May 2000,
<http://www.rfc-editor.org/info/rfc2804>.
[RFC3311] Rosenberg, J., "The Session Initiation Protocol (SIP)
UPDATE Method", RFC 3311, DOI 10.17487/RFC3311,
October 2002, <http://www.rfc-editor.org/info/rfc3311>.
[RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private
Extensions to the Session Initiation Protocol (SIP) for
Asserted Identity within Trusted Networks", RFC 3325,
DOI 10.17487/RFC3325, November 2002,
<http://www.rfc-editor.org/info/rfc3325>.
[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>.
[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>.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006,
<http://www.rfc-editor.org/info/rfc4568>.
[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>.
[RFC4916] Elwell, J., "Connected Identity in the Session Initiation
Protocol (SIP)", RFC 4916, DOI 10.17487/RFC4916,
June 2007, <http://www.rfc-editor.org/info/rfc4916>.
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007,
<http://www.rfc-editor.org/info/rfc4961>.
[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>.
[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>.
[RFC5168] Levin, O., Even, R., and P. Hagendorf, "XML Schema for
Media Control", RFC 5168, DOI 10.17487/RFC5168,
March 2008, <http://www.rfc-editor.org/info/rfc5168>.
[RFC5630] Audet, F., "The Use of the SIPS URI Scheme in the Session
Initiation Protocol (SIP)", RFC 5630,
DOI 10.17487/RFC5630, October 2009,
<http://www.rfc-editor.org/info/rfc5630>.
[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>.
[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for
Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263,
DOI 10.17487/RFC6263, June 2011,
<http://www.rfc-editor.org/info/rfc6263>.
[RFC6341] Rehor, K., Ed., Portman, L., Ed., Hutton, A., and R. Jain,
"Use Cases and Requirements for SIP-Based Media Recording
(SIPREC)", RFC 6341, DOI 10.17487/RFC6341, August 2011,
<http://www.rfc-editor.org/info/rfc6341>.
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
September 2013, <http://www.rfc-editor.org/info/rfc7022>.
[RFC7525] Sheffer, Y., Holz, R., and P. Saint-Andre,
"Recommendations for Secure Use of Transport Layer
Security (TLS) and Datagram Transport Layer Security
(DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525,
May 2015, <http://www.rfc-editor.org/info/rfc7525>.
Acknowledgements
We want to thank John Elwell, Paul Kyzivat, Partharsarathi R, Ram
Mohan R, Hadriel Kaplan, Adam Roach, Miguel Garcia, Thomas Stach,
Muthu Perumal, Dan Wing, and Magnus Westerlund for their valuable
comments and inputs to this document.
Authors' Addresses
Leon Portman
NICE Systems
22 Zarhin Street
P.O. Box 690
Ra'anana 4310602
Israel
Email: leon.portman@gmail.com
Henry Lum (editor)
Genesys
1380 Rodick Road, Suite 201
Markham, Ontario L3R4G5
Canada
Email: henry.lum@genesyslab.com
Charles Eckel
Cisco
170 West Tasman Drive
San Jose, CA 95134
United States
Email: eckelcu@cisco.com
Alan Johnston
Illinois Institute of Technology
Bellevue, WA
United States
Email: alan.b.johnston@gmail.com
Andrew Hutton
Unify
Brickhill Street
Milton Keynes MK15 0DJ
United Kingdom
Email: andrew.hutton@unify.com