| Rfc | 4240 |
|---|
| Title | Basic Network Media Services with SIP |
|---|
| Author | E. Burger, Ed., J. Van Dyke,
A. Spitzer |
|---|
| Date | December 2005 |
|---|
| Format: | TXT, HTML |
|---|
| Status: | INFORMATIONAL |
|---|
|
Network Working Group E. Burger, Ed.
Request for Comments: 4240 J. Van Dyke
Category: Informational A. Spitzer
Brooktrout Technology, Inc.
December 2005
Basic Network Media Services with SIP
Status of This Memo
This memo provides information for the Internet community. It does
not specify an Internet standard of any kind. Distribution of this
memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2005).
Abstract
In SIP-based networks, there is a need to provide basic network media
services. Such services include network announcements, user
interaction, and conferencing services. These services are basic
building blocks, from which one can construct interesting
applications. In order to have interoperability between servers
offering these building blocks (also known as Media Servers) and
application developers, one needs to be able to locate and invoke
such services in a well defined manner.
This document describes a mechanism for providing an interoperable
interface between Application Servers, which provide application
services to SIP-based networks, and Media Servers, which provide the
basic media processing building blocks.
Table of Contents
1. Overview ........................................................2
1.1. Conventions Used in This Document ..........................3
2. Mechanism .......................................................3
3. Announcement Service ............................................5
3.1. Operation ..................................................8
3.2. Protocol Diagram ...........................................9
3.3. Formal Syntax ..............................................9
4. Prompt and Collect Service .....................................11
4.1. Formal Syntax for Prompt and Collect Service ..............12
5. Conference Service .............................................13
5.1. Protocol Diagram ..........................................14
5.2. Formal Syntax .............................................16
6. IANA Considerations ............................................17
7. The User Part ..................................................17
8. Security Considerations ........................................20
9. Contributors ...................................................20
10. Acknowledgements ..............................................20
11. References ....................................................21
11.1. Normative References .....................................21
11.2. Informative References ...................................22
1. Overview
In SIP-based media networks (RFC 3261 [10]), there is a need to
provide basic network media services. Such services include playing
announcements, initiating a media mixing session (conference), and
prompting and collecting information with a user.
These services are basic in nature, are few in number, and
fundamentally have not changed in 25 years of enhanced telephony
services. Moreover, given their elemental nature, one would not
expect them to change in the future.
Multifunction media servers provide network media services to clients
using server protocols such as SIP, often in conjunction with markup
languages such as VoiceXML [20] and MSCML [21]. This document
describes how to identify to a multifunction media server what sort
of session the client is requesting, without modifying the SIP
protocol.
It is critically important to note that the mechanism described here
in no way modifies the SIP protocol, the meaning, or definition of a
SIP Request URI, or does it put any restrictions, in any way, on
devices that do not implement this convention.
Announcements are media played to the user. Announcements can be
static media files, media files generated in real-time, media streams
generated in real-time, multimedia objects, or combinations of the
above.
Media mixing is the act of mixing different RTP streams, as described
in RFC 3550 [13]. Note that the service described here suffices for
simple mixing of media for a basic conferencing service. This
service does not address enhanced conferencing services, such as
floor control, gain control, muting, subconferences, etc. MSCML [21]
addresses enhanced conferencing. However, that is beyond the scope
of this document. Interested readers should read conferencing-
framework [22] for details on the IETF SIP conferencing framework.
Prompt and collect is where the server prompts the user for some
information, as in an announcement, and then collects the user's
response. This can be a one-step interaction, for example by playing
an announcement, "Please enter your pass code", followed by
collecting a string of digits. It can also be a more complex
interaction, specified, for example, by VoiceXML [20] or MSCML [21].
1.1. Conventions Used in This Document
RFC 2119 [6] the interpretations for the key words "MUST", "MUST
NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT",
"RECOMMENDED", "MAY", and "OPTIONAL" found in this document.
2. Mechanism
In the context of SIP control of media servers, we take advantage of
the fact that the standard SIP URI has a user part. Multifunction
media servers do not have users. Thus we use the user address, or
the left-hand-side of the URI, as a service indicator.
The use of the user part of the SIP Request URI has a number of
useful properties:
o There is no change to core SIP.
o Only devices that choose to conform to this standard have to
implement it.
o This document only applies to multifunction SIP-controlled media
servers.
o This document has no impact on non-multifunction SIP-controlled
media servers.
o The mechanism described in this document has absolutely no impact
on SIP devices other than media servers.
The last bullet point is crucial. In particular, the user part
convention described here places absolutely no restrictions on any
SIP user agent, proxy, back-to-back user agent (B2BUA), or any future
device. The user parts defined here only apply to multifunction
media servers that chose to implement the convention. With the
exception of a conforming media server, these user names and
conventions have no impact on the user part namespace. They do not
restrict the use of these user names at devices other than a
multifunction media server.
Note that the set of services is small, well defined, and well
contained. The section The User Part (Section 7) discusses the
issues with using a fixed set of user-space names.
For per-service security, the media server SHOULD use the security
protocols described in RFC 3261 [10].
The media server MAY issue 401 challenges for authentication. The
media server SHOULD support the sips: scheme for the announcement
service. The media server MUST support the sips: scheme for the
dialog and conference services. The level of authentication to
require for each service is a matter of local policy.
The media server, upon receiving an INVITE, notes the service
indicator. Depending on the service indicator, the media server will
either honor the request or return a failure response code.
The service indicator is the concatenation of the service name and an
optional service instance identifier, separated by an equal sign.
Per RFC 3261 [10], the service indicator is case insensitive. The
service name MUST be from the set alphanumeric characters plus dash
(US-ASCII %2C). The service name MUST NOT include an equal sign
(US-ASCII %3D).
The service name MAY have long- and short-forms, as SIP does for
headers.
A given service indicator MAY have an associated set of parameters.
Such parameters MUST follow the convention set out for SIP URI
parameters. That is, a semi-colon separated list of keyword=value
pairs.
Certain services may have an association with a unique service
instance on the media server. For example, a given media server can
host multiple, separate conference sessions. To identify unique
service instances, a unique identifier modifies the service name.
The unique identifier MUST meet the rules for a legal user part of a
SIP URI. An equal sign, US-ASCII %3D, MUST separate the service
indicator from the unique identifier.
Note that since the service indicator is case insensitive, the
service instance identifier is also case insensitive.
The requesting client issues a SIP INVITE to the media server,
specifying the requested service and any appropriate parameters.
If the media server can perform the requested service, it does so,
following the processing steps described in the service definition
document.
If the media server cannot perform the requested service or does not
recognize the service indicator, it MUST respond with the response
code 488 NOT ACCEPTABLE HERE. This is appropriate, as 488 refers to
a problem with the user part of the URI. Moreover, 606 is not
appropriate, as some other media server may be able to satisfy the
request. RFC 3261 [10] describes the 488 and 606 response codes.
Some services require a unique identifier. Most services
automatically create a service instance upon the first INVITE with
the given identifier. However, if a service requires an existing
service instance, and no such service instance exists on the media
server, the media server MUST respond with the response code 404 NOT
FOUND. This is appropriate as the service itself exists on the media
server, but the particular service instance does not. It is as if
the user was not home.
3. Announcement Service
A network announcement is the delivery of a multimedia resource, such
as a prompt file, to a terminal device. Note the multimedia resource
may be any multimedia object that the media server supports. This
service can play a single object with multiple streams, such as a
video and audio prompt. However, this service cannot play multiple
objects on the same SIP dialog.
There are two types of network announcements. The differentiating
characteristic between the two types is whether the network fully
sets up the SIP dialog before playing the announcement. The analog
in the Public Switched Telephone Network (PSTN) is whether answer
supervision is supplied (i.e., does the announcement server answer
the call prior to delivering the announcement?).
Playing an announcement after call setup is straightforward. First,
the requesting device issues an INVITE to the media server requesting
the announcement service. The media server negotiates the SDP and
responds with a 200 OK. After receiving the ACK from the requesting
device, the media server plays the requested object and issues a BYE
to the requesting device.
If the media server supports announcements, but it cannot find the
referenced URI, it MUST respond with the 404 response code and SHOULD
send the reason phrase "Announcement content not found".
If the media server receives an INVITE for the announcement service
without a "play=" parameter, it MUST respond with the response code
400 and SHOULD send the reason phrase "Mandatory play parameter
missing".
If there is an error retrieving the announcement, the media server
MUST respond with a 400 response code and SHOULD send the reason
phrase "Announcement content could not be retrieved". In addition
the media server SHOULD include a Warning header with appropriate
explanatory text explaining what failed.
The Request URI fully describes the announcement service through the
use of the user part of the address and additional URI parameters.
The user portion of the address, "annc", specifies the announcement
service on the media server. The service has several associated URI
parameters that control the content and delivery of the announcement.
These parameters are described below:
play
Specifies the resource or announcement sequence to be played.
repeat
Specifies how many times the media server should repeat the
announcement or sequence named by the "play=" parameter. The
value "forever" means the repeat should be effectively unbounded.
In this case, it is RECOMMENDED the media server implements some
local policy, such as limiting what "forever" means, to ensure
errant clients do not create a denial of service attack.
delay
Specifies a delay interval between announcement repetitions. The
delay is measured in milliseconds.
duration
Specifies the maximum duration of the announcement. The media
server will discontinue the announcement and end the call if the
maximum duration has been reached. The duration is measured in
milliseconds.
locale
Specifies the language and optionally country variant of the
announcement sequence named in the "play=" parameter. RFC 3066
[9] specifies the locale tag. The locale tag is usually a two- or
three-letter code per ISO 639-1 [11]. The country variant is also
often a two-letter code per ISO 3166-1 [12]. These elements are
concatenated with a single under bar (%x5F) character, such as
"en_CA". If only the language is specified, such as locale=en,
the choice of country variant is an implementation matter.
Implementations SHOULD provide the best possible match between the
requested locale and the available languages in the event the
media server cannot honor the locale request precisely. For
example, if the request has locale=ca_FR, but the media server
only has fr_FR available, the media server should use the fr_FR
variant. Implementations SHOULD provide a default locale to use
if no language variants are available.
param[n]
Provides a mechanism for passing values that are to be substituted
into an announcement sequence. Up to 9 parameters ("param1="
through "param9=") may be specified. The mechanics of
announcement sequences are beyond the scope of this document.
extension
Provides a mechanism for extending the parameter set. If the
media server receives an extension it does not understand, it MUST
silently ignore the extension parameter and value.
The "play=" parameter is mandatory and MUST be present. All other
parameters are OPTIONAL.
NOTE: Some encodings are not self-describing. Thus, the
implementation relies on filename extension conventions for
determining the media type.
Note that RFC 3261 [10] implies that proxies are supposed to pass
parameters through unchanged. However, be aware that non-conforming
proxies may strip Request-URI parameters. That said, given the
likely scenarios for the mechanisms presented in this document, this
should not be an issue. Most likely, the proxy inserting the
parameters is the last proxy before the media server. If the service
provider deploys a proxy for load balancing or service location
purposes, the service provider should ensure that its choice of proxy
preserves parameters.
The form of the SIP Request URI for announcements is as follows.
Note that the backslash, CRLF, and spacing before the "play=" in the
example is for readability purposes only.
sip:annc@ms2.example.net; \
play=http://audio.example.net/allcircuitsbusy.g711
sip:annc@ms2.example.net; \
play=file://fileserver.example.net//geminii/yourHoroscope.wav
3.1. Operation
The scenarios below assume there is a SIP Proxy, application server,
or media gateway controller between the caller and the media server.
However, the announcement service works as described below even if
the caller invokes the service directly. We chose to discuss the
proxy case, as it will be the most common case.
The caller issues an INVITE to the serving SIP Proxy. The SIP Proxy
determines what audio prompt to play to the caller. The proxy
responds to the caller with 100 TRYING.
It is important to note that the mechanism described here in no way
modifies the behavior of SIP [10]. In particular, this convention
does not modify SDP negotiation [18].
The proxy issues an INVITE to the media server, requesting the
appropriate prompt to play coded in the play= parameter. The media
server responds with 200 OK. The proxy relays the 200 OK to the
caller. The caller then issues an ACK. The proxy then relays the
ACK to the media server.
With the call established, the media server plays the requested
prompt. When the media server completes the play of the prompt, it
issues a BYE to the proxy. The proxy then issues a BYE to the
caller.
3.2. Protocol Diagram
Caller Proxy Media Server
| INVITE | |
|----------------------->| INVITE |
| 100 TRYING |----------------------->|
|<-----------------------| 200 OK |
| 200 OK |<-----------------------|
|<-----------------------| |
| ACK | |
|----------------------->| ACK |
| |----------------------->|
| | |
| Play Announcement (RTP) |
|<================================================|
| | |
| | BYE |
| BYE |<-----------------------|
|<-----------------------| |
| 200 OK | |
|----------------------->| 200 OK |
| |----------------------->|
| | |
3.3. Formal Syntax
The following syntax specification uses the augmented Backus-Naur
Form (BNF) as described in RFC 4234 [7].
ANNC-URL = sip-ind annc-ind "@" hostport
annc-parameters uri-parameters
sip-ind = "sip:" / "sips:"
annc-ind = "annc"
annc-parameters = ";" play-param [ ";" content-param ]
[ ";" delay-param]
[ ";" duration-param ]
[ ";" repeat-param ]
[ ";" locale-param ]
[ ";" variable-params ]
[ ";" extension-params ]
play-param = "play=" prompt-url
content-param = "content-type=" MIME-type
delay-param = "delay=" delay-value
delay-value = 1*DIGIT
duration-param = "duration=" duration-value
duration-value = 1*DIGIT
repeat-param = "repeat=" repeat-value
repeat-value = 1*DIGIT / "forever"
locale-param = "locale=" token
; per RFC 3066, usually
; ISO639-1_ISO3166-1
; e.g., en, en_US, en_UK, etc.
variable-params = param-name "=" variable-value
param-name = "param" DIGIT ; e.g., "param1"
variable-value = 1*(ALPHA / DIGIT)
extension-params = extension-param [ ";" extension-params ]
extension-param = token "=" token
"uri-parameters" is the SIP Request-URI parameter list as described
in RFC 3261 [10]. All parameters of the Request URI are part of the
URI matching algorithm.
The MIME-type is the MIME [1] [2] [3] [4] [5] content type for the
announcement, such as audio/basic, audio/G729, audio/mpeg,
video/mpeg, and so on.
A number of MIME registrations, which could be used here, have
parameters, for instance, video/DV. To accommodate this, and retain
compatibility with the SIP URI structure, the MIME-type parameter
separator (semicolon, %3b) and value separator (equal, %d3) MUST be
escaped. For example:
sip:annc@ms.example.net; \
play=file://fs.example.net//clips/my-intro.dvi; \
content-type=video/mpeg%3bencode%d3314M-25/625-50
The locale-value consists of a tag as specified in RFC 3066 [9].
The definition of hostport is as specified by RFC 3261 [10].
The syntax of prompt-url consists of a URL scheme as specified by RFC
3986 [8] or a special token indicating a provisioned announcement
sequence. For example, the URL scheme MAY include any of the
following.
o http/https
o ftp
o file (referencing a local or NFS (RFC 3530 [16]) object)
o nfs (RFC 2224 [14])
If a provisioned announcement sequence is to be played, the value of
prompt-url will have the following form:
prompt-url = "/provisioned/" announcement-id
announcement-id = 1*(ALPHA / DIGIT)
Note that the scheme "/provisioned/" was chosen because of a
hesitation to register a "provisioned:" URI scheme.
This document is strictly focused on the SIP interface for the
announcement service and, as such, does not detail how announcement
sequences are provisioned or defined.
Note that the media type of the object the prompt-url refers to can
be most anything, including audio file formats, text file formats, or
URI lists. See the Prompt and Collect Service (Section 4) section
for more on this topic.
4. Prompt and Collect Service
This service is also known as a voice dialog. It establishes an
aural dialog with the user.
The dialog service follows the model of the announcement service.
However, the service indicator is "dialog". The dialog service takes
a parameter, voicexml=, indicating the URI of the VoiceXML script to
execute.
sip:dialog@mediaserver.example.net; \
voicexml=http://vxmlserver.example.net/cgi-bin/script.vxml
A Media Server MAY accept additional SIP request URI parameters and
deliver them to the VoiceXML interpreter session as session
variables.
Although not good VoiceXML programming practice, VoiceXML scripts
might contain sensitive information, such as a user's pass code in a
DTMF grammar. Thus, the media server MUST support the https scheme
for the voicexml parameter for secure fetching of scripts. Likewise,
dynamic grammars often do have user-identifying information. As
such, the VoiceXML browser implementation on the media server MUST
support https fetching of grammars and subsequent documents.
Returned information often is sensitive. For example, the
information could be financial information or instructions. Thus,
the media server MUST support https posting of results.
4.1. Formal Syntax for Prompt and Collect Service
The following syntax specification uses the augmented Backus-Naur
Form (BNF) as described in RFC 4234 [7].
DIALOG-URL = sip-ind dialog-ind "@" hostport
dialog-parameters
sip-ind = "sip:" / "sips:"
dialog-ind = "dialog"
dialog-parameters = ";" dialog-param [ vxml-parameters ]
[ uri-parameters ]
dialog-param = "voicexml=" vxml-url
vxml-parameters = vxml-param [ vxml-parameters ]
vxml-param = ";" vxml-keyword "=" vxml-value
vxml-keyword = token
vxml-value = token
The vxml-url is the URI of the VoiceXML script. If present, other
parameters get passed to the VoiceXML interpreter session with the
assigned vxml-keyword vxml-value pairs. Note that all vxml-keywords
MUST have values.
If there is a vxml-keyword without a corresponding vxml-value, the
media server MUST reject the request with a 400 BAD REQUEST response
code. In addition, the media server MUST state "Missing VXML Value"
in the reason phrase.
The media server presents the parameters as environment variables in
the connection object. Specifically, the parameter appears in the
connection.sip tree.
If the Media Server does not support the passing of keyword-value
pairs to the VoiceXML interpreter session, it MUST ignore the
parameters.
"uri-parameters" is the SIP Request-URI parameter list as described
in RFC 3261 [10]. All parameters in the parameter list, whether they
come from uri-parameters or from vxml-keyworks, are part of the URI
matching algorithm.
5. Conference Service
One identifies mixing sessions through their SIP request URIs. To
create a mixing session, one sends an INVITE to a request URI that
represents the session. If the URI does not already exist on the
media server and the requested resources are available, the media
server creates a new mixing session. If there is an existing URI for
the session, then the media server interprets it as a request for the
new session to join the existing session. The form of the SIP
request URI for conferencing is:
sip:conf=uniqueIdentifier@mediaserver.example.net
The left-hand side of the request URI is actually the username of the
request in the request URI and the To header. The host portion of
the URI identifies a particular media server. The "conf" user name
conveys to the media server that this is a request for the mixing
service. The uniqueIdentifier can be any value that is compliant
with the SIP URI specification. It is the responsibility of the
conference control application to ensure the identifier is unique
within the scope of any potential conflict.
In the terminology of the conferencing framework [22], this URI
convention tells the media server that the application server is
requesting it to act as a Focus. The conf-id value identifies the
particular focus instance.
As a focus in the conferencing framework, the media server MUST
support the ";isfocus" parameter in the Request URI. Note, however,
that the presence or absence of the ";isfocus" parameter has no
protocol impact at the media server.
It is worth noting that the conference URI shared between the
application and media servers provides enhanced security, as the SIP
control interface does not have to be exposed to participants. It
also allows the assignment of a specific media server to be delayed
as long as possible, thereby simplifying resource management.
One can add additional legs to the conference by INVITEing them to
the above-mentioned request URI. Per the matching rules of RFC 3261
[10], the conf-id parameter is part of the matching string.
Conversely, one can remove legs by issuing a BYE in the corresponding
dialog. The mixing session, and thus the conference-specific request
URI, remains active so long as there is at least one SIP dialog
associated with the given request URI.
If the Request-URI has "conf" as the user part, but does not have a
conf-id parameter, the media server MUST respond with a 404 NOT
FOUND.
NOTE: The media server could create a unique conference instance
and return the conf-id string to the User Agent Clinet (UAC) if
there is no conf-id present. However, such an operation may have
other operational issues, such as permissions and billing. Thus
an application server or proxy is a better place to do such an
operation. Moreover, such action would make the media server into
a Conference Factory in the terminology of conference-framework
[22]. That is not the appropriate behavior for a media server.
Since some conference use cases, such as business conferencing, have
billing implications, the media server SHOULD authenticate the
application server or proxy. At a minimum, the media server MUST
implement sips:.
5.1. Protocol Diagram
This diagram shows the establishment of a three-way conference. This
section is informative. It is only one method of establishing a
conference. This example shows a simple back-to-back user agent.
The conference-framework [22] describes additional parameters and
behaviors of the Application Server. For example, the first INVITE
from P1 to the Application Server would include the ";isfocus"
parameter; the Application Server would act as a Conference Factory;
and so on. However, none of that protocol machinery has an impact on
the operation of the Application Server to Media Server interface,
which is the focus of this protocol document.
P1 P2 P3 Application Server Media Server
| | | | |
| INVITE sip:public-conf@as.example.net |
|---------------------------------->| |
| | | INVITE sip:conf=123@ms.example.net |
| | | |------------------>|
| | | | 200 OK |
| 200 OK | |<------------------|
|<----------------------------------| |
| ACK | | | |
|---------------------------------->| ACK |
| | | |------------------>|
| | | RTP w/ P1 | |
|<=====================================================>|
| | | | |
| INVITE sip:public-conf@as.example.net |
| |-------------------------->| |
| | | INVITE sip:conf=123@ms.example.net |
| | | |------------------>|
| | | | 200 OK |
| | 200 OK | |<------------------|
| |<--------------------------| |
| | ACK | | |
| |-------------------------->| ACK |
| | | |------------------>|
| | | | |
| | | RTP w/ P1+P2-P2 | |
| |<=============================================>|
| | | RTP w/ P1+P2-P1 | |
|<=====================================================>|
| | | | |
| INVITE sip:public-conf@as.example.net |
| | |----------------->| |
| | | INVITE sip:conf=123@ms.example.net |
| | | |------------------>|
| | | | 200 OK |
| | | 200 OK |<------------------|
| | |<-----------------| |
| | | ACK | |
| | |----------------->| ACK |
| | | |------------------>|
| | | | |
| | | RTP w/ P1+P2+P3-P3 |
| | |<====================================>|
| | | RTP w/ P1+P2+P3-P2 |
| |<=============================================>|
| | | RTP w/ P1+P2+P3-P1 |
|<=====================================================>|
| | | | |
| | | | |
Using the terminology of conference-framework [22], the Application
Server is the Conference Factory, and the Media Server is the
Conference Focus.
Note that the above call flow does not show any 100 TRYING messages
that would typically flow from the Application Server to the UACs;
nor does it show the ACKs from the UACs to the Application Server or
from the Application Server to the Media Server.
Each leg can drop out either under the supervision of the UAC, by the
UAC sending a BYE, or under the supervision of the Application
Server, by the Application Server issuing a BYE. In either case, the
Application Server will either issue a BYE on behalf of the UAC or
issue it directly to the Media Server, corresponding to the
respective disconnect case.
It is left as a trivial exercise to the reader for how the
Application Server can mute legs, create side conferences, and so
forth.
Note that the Application Server is a server to the participants
(UACs). However, the Application Server is a client for mixing
services to the Media Server.
5.2. Formal Syntax
The following syntax specification uses the augmented Backus-Naur
Form (BNF) as described in RFC 4234 [7].
CONF-URL = sip-ind conf-ind "=" instance-id "@" hostport
[ uri-parameters ]
sip-ind = "sip:" / "sips:"
conf-ind = "conf"
instance-id = token
"uri-parameters" is the SIP Request-URI parameter list as described
in RFC 3261 [10]. All parameters in the parameter list are part of
the URI matching algorithm.
6. IANA Considerations
The IANA has registered the following parameters in the SIP/SIPS URI
Parameters registry, following the specification required policy of
RFC 3969 [19]:
Parameter Name Predefined Values Reference
-------------- ----------------- ---------
play no RFC 4240
repeat no RFC 4240
delay no RFC 4240
duration no RFC 4240
locale no RFC 4240
param[n] no RFC 4240
extension no RFC 4240
7. The User Part
There has been considerable discussion about the wisdom of using
fixed user parts in a request URI. The most common objection is that
the user part should be opaque and a local matter. The other
objection is that using a fixed user part removes those specified
user addresses from the user address space.
We address the latter issue first. The common example is the
Postmaster address defined by RFC 2821 [15]. The objection is that
by using the Postmaster token for something special, one removes that
token for anyone. Thus, the Postmaster General of the United States,
for example, cannot have the mail address Postmaster@usps.gov.
However, one may debate whether this is a significant limitation.
This document explicitly addresses this issue. The user names
described in the text (namely annc, ivr, dialog, and conf) are
available for whatever local use a given SIP user agent or proxy
wishes for them. What this document does is give special meaning for
these user names at media servers that implement this specification.
If a media server chooses not to implement this specification,
nothing breaks. If a user wishes to use one of the user names
described in this document at their SIP user agent, nothing breaks
and their user agent will work as expected.
The key point is, one cannot confuse the namespace at a Media Server
with the namespace for an organization. For example, let us take the
case where a network offers services for "Ann Charles". She likes to
use the name "annc", and thus she would like to use
"sip:annc@example.net". We offer there is ABSOLUTELY NO NAME
COLLISION WHATSOEVER. Why is this so? This is so because
sip:annc@example.net will resolve to the specific user at a specific
device for Ann. As an example, example.net's SIP Proxy Server
resolves sip:annc@example.net to annc@anns-phone.example.net.
Conversely, one directs requests for the media service annc directly
to the Media Server, e.g., sip:annc@ms21.ap.example.net. Moreover,
by definition, requests for Ann Charles, or anything other than the
announcement service, will NEVER be directly sent to the Media
Server. If that were not true, no phone in the world could use the
user part "eburger", as eburger is a reserved user part in the
Brooktrout domain. Clearly, this is not the case.
If one wishes to make their media server accessible to the global
Internet, but retain one of the Media Server-specific user names in
the domain, a SIP Proxy can easily translate whatever opaque name one
chooses to the Media Server-specific user name. For example, if a
domain wishes to offer services for the above mentioned Ann Charles
at sip:annc@example.com, they can offer the announcement service at
sip:my-special-announcement-service@example.com. The former address,
sip:annc@example.com, would resolve to the actual device where annc
resides. The latter would resolve to the media server announcement
server address, sip:annc@mediaserver.example.com, as an example.
Note that this convention makes it easier to provision this service.
With a fixed mapping at the multifunction media server, there are
less provisioning data elements to get wrong.
Here is another way of looking at this issue. Unix reserves the
special user "root". Just about all Unix machines have a user root,
who has an address "root@a-specific-machine.example.com", where
"a-specific-machine" is the fully-qualified domain name (FQDN) of a
particular instance of a machine. There are very well-defined
semantics for the "root" user.
Even though most every Unix machine has a "root" user, often there is
no mapping for a "root" user in a domain, such as "root@example.com".
Conversely, there is no restriction on creating an MX record for
"root@example.com". That choice is fully up to the administrative
authority for the domain.
The "users" proposed by this document, "annc", "conf", and "dialog"
are all users at a Media Server, just as the "root", "bin", and
"nobody" users are "users" at a Unix host.
After much discussion, with input from the W3C URI work group, we
considered obfuscating the user name by prepending "__sip-" to the
user name. However, as explained above, this obfuscation is not
necessary. There is a fundamental difference between a user name at
a device and a user name at an MX record (SMTP) or Address-of-Record
(SIP). Again, there is no possibility that the name on the device
may "leak out" into the SIP routing network.
The most important thing to note about this convention is that the
left-hand side of the request URI is opaque to the network. The only
network elements that need to know about the convention are the Media
Server and client. Even proxies doing mapping resolution, as in the
example above for public announcement services, do not need to be
aware of the convention. The convention is purely a matter of
provisioning.
Some have proposed that such naming be a pure matter of local
convention. For example, the thesis of the informational RFC RFC
3087 [17] is that you can address services using a request URI.
However, some have taken the examples in the document to an extreme.
Namely, that the only way to address services is via arbitrary,
opaque, long user parts. Clearly, it is possible to provision the
service names, rather than fixed names. While this can work in a
closed network, where the Application Servers and Media Servers are
in the same administrative domain, this does not work across domains,
such as in the Internet. This is because the client of the media
service has to know the local name for each service / domain pair.
This is particularly onerous for situations where there is an ad hoc
relationship between the application and the media service. Without
a well-known relationship between service and service address, how
would the client locate the service?
One very important result of using the user part as the service
descriptor is that we can use all of the standard SIP machinery,
without modification. For example, Media Servers with different
capabilities can SIP Register their capabilities as users. For
example, a VoiceXML-only device will register the "dialog" user,
while a multi-purpose Media Server will register all of the users.
Note that this is why the URI to play is a parameter. Doing
otherwise would overburden a normal SIP proxy or redirect server.
Conversely, having the conference ID be part of the user part gives
an indication that requests get routed similarly (as opposed to
requiring a Globally Routable User Agent URI (GRUU), which would
restrict routing to the same device).
Likewise, this scheme lets us leverage the standard SIP proxy
behavior of using an intelligent redirect server or proxy server to
provide high-available services. For example, two Media Servers can
register with a SIP redirect server for the annc user. If one of the
Media Servers fails, the registration will expire and all requests
for the announcement service ("calls to the annc user") will get sent
to the surviving Media Server.
8. Security Considerations
Exposing network services with well-known addresses may not be
desirable. The Media Server SHOULD authenticate and authorize
requesting endpoints per local policy.
Some interactions in this document result in the transfer of
confidential information. Moreover, many of the interactions require
integrity protection. Thus, the Media Server MUST implement the
sips: scheme. In addition, application developers are RECOMMENDED to
use the security services offered by the Media Server to ensure the
integrity and confidentiality of their user's data, as appropriate.
Untrusted network elements could use the convention described here
for providing information services. Many extant billing arrangements
are for completed calls. Successful call completion occurs with a
2xx result code. This can be an issue for the early media
announcement service. This is one of the reasons why the early media
announcement service is deprecated.
Services such as repeating an announcement forever create the
possibility for denial of service attacks. The media server SHOULD
have local policies to deal with this, such as time-limiting how long
"forever" is, analyzing where multiple requests come from,
implementing white-lists for such a service, and so on.
9. Contributors
Jeff Van Dyke and Andy Spitzer of SnowShore did just about all of the
work developing netann, in conjunction with many application
developers, media server manufacturers, and service providers, some
of whom are listed in the Acknowledgements section. All I did was do
the theory and write it up. That also means all of the mistakes are
mine, as well.
10. Acknowledgements
We would like to thank Kevin Summers and Ravindra Kabre of Sonus
Networks for their constructive comments, as well as Jonathan
Rosenberg of Dynamicsoft and Tim Melanchuk of Convedia for their
encouragement. In addition, the discussion at the Las Vegas Interim
Workgroup Meeting in 2002 was invaluable for clearing up the issues
surrounding the left-hand-side of the request URI. Christer Holmberg
helped tune the language of the multimedia announcement service.
Orit Levin from Radvision gave a close read on the most recent
version of the document. Pete Danielsen from Lucent has consistently
provided excellent reviews of the many different versions of this
document.
Pascal Jalet provided the theoretical underpinning and David Rio
provided the experimental evidence for why the conference identifier
belongs in the user part of the request-URI.
I am particularly indebted to Alan Johnston for his review of this
document and ensuring its conformance with the SIP conference control
work in the IETF.
Mary Barnes, as usual, found the holes and showed how to fix them.
The authors would like to give a special thanks to Walter O'Connor
for doing much of the initial implementation.
Note that at the time of this writing, there are 7 known independent
server implementations that are interoperable with 23 known client
implementations. Our apologies if we did not count your
implementation.
11. References
11.1. Normative References
[1] Freed, N. and N. Borenstein, "Multipurpose Internet Mail
Extensions (MIME) Part One: Format of Internet Message Bodies",
RFC 2045, November 1996.
[2] Freed, N. and N. Borenstein, "Multipurpose Internet Mail
Extensions (MIME) Part Two: Media Types", RFC 2046,
November 1996.
[3] Moore, K., "MIME (Multipurpose Internet Mail Extensions) Part
Three: Message Header Extensions for Non-ASCII Text", RFC 2047,
November 1996.
[4] Freed, N., Klensin, J., and J. Postel, "Multipurpose Internet
Mail Extensions (MIME) Part Four: Registration Procedures",
BCP 13, RFC 2048, November 1996.
[5] Freed, N. and N. Borenstein, "Multipurpose Internet Mail
Extensions (MIME) Part Five: Conformance Criteria and
Examples", RFC 2049, November 1996.
[6] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[7] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", RFC 4234, October 2005.
[8] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Syntax", STD 66, RFC 3986,
January 2005.
[9] Alvestrand, H., "Tags for the Identification of Languages",
BCP 47, RFC 3066, January 2001.
[10] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
Session Initiation Protocol", RFC 3261, June 2002.
[11] International Organization for Standardization, "Codes for the
representation of names of languages -- Part 1: Alpha-2 code",
ISO Standard 639-1, July 2002.
[12] International Organization for Standardization, "Codes for the
representation of names of countries and their subdivisions --
Part 1: Country codes", ISO Standard 3166-1, October 1997.
11.2. Informative References
[13] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", STD 64,
RFC 3550, July 2003.
[14] Callaghan, B., "NFS URL Scheme", RFC 2224, October 1997.
[15] Klensin, J., "Simple Mail Transfer Protocol", RFC 2821,
April 2001.
[16] Shepler, S., Callaghan, B., Robinson, D., Thurlow, R., Beame,
C., Eisler, M., and D. Noveck, "Network File System (NFS)
version 4 Protocol", RFC 3530, April 2003.
[17] Campbell, B. and R. Sparks, "Control of Service Context using
SIP Request-URI", RFC 3087, April 2001.
[18] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
[19] Camarillo, G., "The Internet Assigned Number Authority (IANA)
Uniform Resource Identifier (URI) Parameter Registry for the
Session Initiation Protocol (SIP)", BCP 99, RFC 3969,
December 2004.
[20] Burnett, D., Hunt, A., McGlashan, S., Porter, B., Lucas, B.,
Ferrans, J., Rehor, K., Carter, J., Danielsen, P., and S.
Tryphonas, "Voice Extensible Markup Language (VoiceXML) Version
2.0", W3C REC REC-voicexml20-20040316, March 2004.
[21] Van Dyke, J., Burger, E., Ed., and A. Spitzer, "Media Server
Control Markup Language (MSCML) and Protocol", Work in
Progress, December 2004.
[22] Rosenberg, J., "A Framework for Conferencing with the Session
Initiation Protocol", Work in Progress, October 2004.
Authors' Addresses
Eric Burger
Brooktrout Technology, Inc.
18 Keewaydin Dr.
Salem, NH 03079
USA
EMail: eburger@brooktrout.com
Jeff Van Dyke
Brooktrout Technology, Inc.
18 Keewaydin Dr.
Salem, NH 03079
USA
EMail: jvandyke@brooktrout.com
Andy Spitzer
Brooktrout Technology, Inc.
18 Keewaydin Dr.
Salem, NH 03079
USA
EMail: woof@brooktrout.com
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