| Network Working Group | P. Saint-Andre |
| Internet-Draft | Cisco Systems, Inc. |
| Intended status: Standards Track | S. Ibarra |
| Expires: January 16, 2014 | AG Projects |
| E. Ivov | |
| Jitsi | |
| July 15, 2013 |
Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Media Sessions
draft-ietf-stox-media-01
This document defines a bi-directional protocol mapping for use by gateways that enable the exchange of media signalling messages between systems that implement the Jingle extensions to the Extensible Messaging and Presence Protocol (XMPP) and those that implement the Session Initiation Protocol (SIP).
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The Session Initiation Protocol [RFC3261] is a widely-deployed technology for the management of media sessions (such as voice calls) over the Internet. SIP itself provides a signalling channel (typically via the User Datagram Protocol [RFC768]), over which two or more parties can exchange messages for the purpose of negotiating a media session that uses a dedicated media channel such as the Real-time Transport Protocol [RFC3550].
The Extensible Messaging and Presence Protocol [RFC6120] also provides a signalling channel, typically via the Transmission Control Protocol [RFC793]. Given the significant differences between XMPP and SIP, it is difficult to combine the two technologies in a single user agent. Therefore, developers wishing to add media session capabilities to XMPP clients have defined an XMPP-specific negotiation protocol called Jingle [XEP-0166].
However, Jingle was designed to easily map to SIP for communication through gateways or other transformation mechanisms. Therefore, consistent with existing specifications for mapping between SIP and XMPP (see [I-D.ietf-stox-core] and other related specifications), this document describes a bi-directional protocol mapping for use by gateways that enable the exchange of media signalling messages between systems that implement SIP and those that implement the XMPP Jingle extensions.
The discussion venue for this document is the mailing list of the STOX WG; visit https://www.ietf.org/mailman/listinfo/stox for subscription information and discussion archives.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in [RFC2119].
A number of technical terms used here are defined in [RFC3261], [RFC6120], [XEP-0166], and [XEP-0167]. The term "JID" is short for "Jabber Identifier".
Even if Jingle has meny similarities with the model used in SIP, there are some use cases that cannot be achieved the same way due to how the offer-answer model is used in SIP in conjustion with SDP.
When using ICE transport, Jingle endpoints are capable of sending candidates in several transport-info meesages. Since there is no equivalent way to achieve that with SIP, [XEP-0176] defines an offer-answer support mode defined by the "urn:ietf:rfc:3264" feature tag. Implementations conforming to this specification MUST support offfer-answer model with Jingle.
If an implementation which conforms to this specification receives a transport-info message from a Jingle endpoint it MAY choose to ignore it or reply to it with an appropriate error.
As mentioned, Jingle was designed in part to enable straightforward protocol mapping between XMPP and SIP. However, given the significantly different technology assumptions underlying XMPP and SIP, Jingle is naturally different from SIP in several important respects:
Jingle is designed in a modular fashion, so that session description data is generally carried in a payload within the generic Jingle elements, i.e., the <jingle/> element and its <content/> child. The following example illustrates this structure, where the XMPP stanza is a request to initiate an audio session using RTP over a raw UDP transport.
<iq from='romeo@example.net/v3rsch1kk3l1jk'
id='ne91v36s'
to='juliet@example.com/t3hr0zny'
type='set'>
<jingle xmlns='urn:xmpp:jingle:1'
action='session-initiate'
initiator='romeo@example.net/v3rsch1kk3l1jk'
sid='a73sjjvkla37jfea'>
<content creator='initiator'
media='audio'
name='this-is-the-audio-content'
senders='both'>
<description xmlns='urn:xmpp:jingle:app:rtp:1'>
<payload-type id='96' name='speex' clockrate='16000'/>
<payload-type id='97' name='speex' clockrate='8000'/>
<payload-type id='18' name='G729'/>
<payload-type channels='2'
clockrate='16000'
id='103'
name='L16'/>
<payload-type id='98' name='x-ISAC' clockrate='8000'/>
</description>
<transport xmlns='urn:xmpp:jingle:transport:raw-udp'>
<candidate ip='10.1.1.104' port='13540' generation='0'/>
</transport>
</content>
</jingle>
</iq>
In the foregoing example, the syntax and semantics of the <jingle/> and <content/> elements are defined in [XEP-0166], the syntax and semantics of the <description/> element are defined in [XEP-0167], and the syntax and semantics of the <transport/> element are defined in [XEP-0177]. Other <description/> elements are defined in specifications for the appropriate application types (see for example [XEP-0167]) and other <transport/> elements are defined in the specifications for appropriate transport methods (see for example [XEP-0176], which defines an XMPP profile of [RFC5245]).
At the core Jingle layer, the following mappings are defined.
+--------------------------------+--------------------------------+
| Jingle | SIP |
+--------------------------------+--------------------------------+
| <jingle/> 'action' | [ see next table ] |
+--------------------------------+--------------------------------+
| <jingle/> 'initiator' | [ no mapping ] |
+--------------------------------+--------------------------------+
| <jingle/> 'responder' | [ no mapping ] |
+--------------------------------+--------------------------------+
| <jingle/> 'sid' | local-part of Call-ID |
+--------------------------------+--------------------------------+
| local-part of 'initiator' | <username> in SDP o= line |
+--------------------------------+--------------------------------+
| <content/> 'creator' | [ no mapping ] |
+--------------------------------+--------------------------------+
| <content/> 'name' | [ no mapping ] |
+--------------------------------+--------------------------------+
| <content/> 'profile' | <proto> in SDP m= line |
+--------------------------------+--------------------------------+
| <content/> 'senders' value of | a= line of sendrecv, recvonly, |
| both, initiator, or responder | or sendonly |
+--------------------------------+--------------------------------+
The 'senders' attribute is optional in Jingle, thus in case it's absent it's RECOMMENDED that the direction value is considered as 'sendrecv'.
The 'action' attribute of the <jingle/> element has nine allowable values. In general they should be mapped as shown in the following table, with some exceptions as described herein.
+-------------------+-----------------+
| Jingle Action | SIP Method |
+-------------------+-----------------+
| content-accept | INVITE response |
| | (1xx or 2xx) |
+-------------------+-----------------+
| content-add | INVITE request |
+-------------------+-----------------+
| content-modify | INVITE request |
+-------------------+-----------------+
| content-remove | INVITE request |
+-------------------+-----------------+
| session-accept | INVITE response |
| | (1xx or 2xx) |
+-------------------+-----------------+
| session-info | [varies] |
+-------------------+-----------------+
| session-initiate | INVITE request |
+-------------------+-----------------+
| session-terminate | BYE |
+-------------------+-----------------+
| transport-info | unnused |
+-------------------+-----------------+
A Jingle application format for audio exchange via RTP is specified in [XEP-0167]. This application format effectively maps to the "RTP/AVP" profile specified in [RFC3551] and the "RTP/SAVP" profile specified in RFC3711, where the media type is "audio" and the specific mappings to SDP syntax are provided in [XEP-0167]. As stated in [XEP-0167] future versions of this specification might define how to use other RTP profiles such as "RTP/AVPF" and "RTP/SAVPF" as fedined in RFC4585 and RFC5124 respectively.
A Jingle application format for video exchange via RTP is specified in [XEP-0167]. This application format effectively maps to the "RTP/AVP" profile specified in [RFC3551] and the "RTP/SAVP" profile specified in RFC3711, where the media type is "audio" and the specific mappings to SDP syntax are provided in [XEP-0167]. As stated in [XEP-0167] future versions of this specification might define how to use other RTP profiles such as "RTP/AVPF" and "RTP/SAVPF" as fedined in RFC4585 and RFC5124 respectively.
A basic Jingle transport method for exchanging media over UDP is specified in [XEP-0177]. This transport method involves the negotiation of an IP address and port only, and does not provide NAT traversal. The Jingle 'ip' attribute maps to the connection-address parameter of the SDP c= line and the 'port' attribute maps to the port parameter of the SDP m= line.
A more advanced Jingle transport method for exchanging media over UDP is specified in [XEP-0176]. Under ideal conditions this transport method provides NAT traversal by following the Interactive Connectivity Exchange methodology specified in [RFC5245].
The relevant SDP mappings are provided in [XEP-0176], however there are a few syntax incompatibilities which need to be addressed by gateways conforming to this specification:
The following sections provide sample scenarios (or "call flows") that illustrate the principles of interworking from Jingle to SIP. These scenarios are not exhaustive.
The protocol flow for a basic voice chat for which an XMPP user (juliet@example.com) is the iniator and a SIP user (romeo@example.net) is the responder. The voice chat is consummated through a gateway. To simplify the example, the transport method negotiated is "raw user datagram protocol" as specified in [XEP-0177].
INITIATOR ...XMPP... GATEWAY ...SIP... RESPONDER
| | |
| session-initiate | |
|----------------------->| |
| IQ-result (ack) | |
|<-----------------------| |
| | INVITE |
| |---------------------->|
| | 180 Ringing |
| |<----------------------|
| session-info (ringing) | |
|<-----------------------| |
| IQ-result (ack) | |
|----------------------->| |
| | 200 OK |
| |<----------------------|
| session-accept | |
|<-----------------------| |
| IQ-result (ack) | |
|----------------------->| |
| | ACK |
| |---------------------->|
| MEDIA SESSION |
|<==============================================>|
| | BYE |
| |<----------------------|
| session-terminate | |
|<-----------------------| |
| IQ-result (ack) | |
|----------------------->| |
| | 200 OK |
| |---------------------->|
| | |
The packet flow is as follows.
First the XMPP user sends a Jingle session-initiation request to the SIP user.
<iq from='juliet@example.com/t3hr0zny'
id='hu2s61f4'
from='romeo@example.net/v3rsch1kk3l1jk'
type='set'>
<jingle xmlns='urn:xmpp:jingle:1'
action='session-initiate'
initiator='juliet@example.com/t3hr0zny'
sid='a73sjjvkla37jfea'>
<content creator='initiator'
media='audio'
name='this-is-the-audio-content'>
<description xmlns='urn:xmpp:jingle:app:rtp:1'>
<payload-type id='96' name='speex' clockrate='16000'/>
<payload-type id='97' name='speex' clockrate='8000'/>
<payload-type id='18' name='G729'/>
</description>
<transport xmlns='urn:xmpp:jingle:transport:raw-udp'>
<candidate ip='192.0.2.101' port='49172' generation='0'/>
</transport>
</content>
</jingle>
</iq>
The gateway returns an XMPP IQ-result to the initiator on behalf of the responder.
<iq from='juliet@example.com/t3hr0zny'
id='hu2s61f4'
to='romeo@example.net/v3rsch1kk3l1jk'
type='result'/>
The gateway transforms the Jingle session-initiate action into a SIP INVITE.
INVITE sip:romeo@example.net SIP/2.0
Via: SIP/2.0/TCP client.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Juliet Capulet <sip:juliet@example.com>;tag=t3hr0zny
To: Romeo Montague <sip:romeo@example.net>
Call-ID: 3848276298220188511@example.com
CSeq: 1 INVITE
Contact: <sip:juliet@client.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 184
v=0
o=alice 2890844526 2890844526 IN IP4 client.example.com
s=-
c=IN IP4 192.0.2.101
t=0 0
m=audio 49172 RTP/AVP 18 96 97
a=rtpmap:96 sppex/16000
a=rtpmap:97 speex/8000
a=rtpmap:18 G729
The responder returns a SIP 180 Ringing message.
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP client.example.com:5060;branch=z9hG4bK74bf9;received=192.0.2.101
From: Juliet Capulet <sip:juliet@example.com>;tag=t3hr0zny
To: Romeo Montague <sip:romeo@example.net>;tag=v3rsch1kk3l1jk
Call-ID: 3848276298220188511@example.com
CSeq: 1 INVITE
Contact: <sip:romeo@client.example.net;transport=tcp>
Content-Length: 0
The gateway transforms the ringing message into XMPP syntax.
<iq from='romeo@montague.net/v3rsch1kk3l1jk'
id='ol3ba71g'
to='juliet@example.com/t3hr0zny'
type='set'>
<jingle xmlns='urn:xmpp:jingle:1'
action='session-info'
initiator='juliet@example.com/t3hr0zny'
sid='a73sjjvkla37jfea'>
<ringing xmlns='urn:xmpp:jingle:apps:rtp:info:1'/>
</jingle>
</iq>
The initiator returns an IQ-result acknowledging receipt of the ringing message, which is used only by the gateway and not transformed into SIP syntax.
<iq from='juliet@example.com/t3hr0zny'
id='ol3ba71g'
to='romeo@example.net/v3rsch1kk3l1jk'
type='result'/>
The responder sends a SIP 200 OK to the initiator.
SIP/2.0 200 OK
Via: SIP/2.0/TCP client.example.com:5060;branch=z9hG4bK74bf9;received=192.0.2.101
From: Juliet Capulet <sip:juliet@example.com>;tag=t3hr0zny
To: Romeo Montague <sip:romeo@example.net>;tag=v3rsch1kk3l1jk
Call-ID: 3848276298220188511@example.com
CSeq: 1 INVITE
Contact: <sip:romeo@client.example.net;transport=tcp>
Content-Type: application/sdp
Content-Length: 147
v=0
o=romeo 2890844527 2890844527 IN IP4 client.example.net
s=-
c=IN IP4 192.0.2.201
t=0 0
m=audio 3456 RTP/AVP 97
a=rtpmap:97 speex/8000
The gateway transforms the 200 OK into a Jingle session-accept action.
<iq from='romeo@example.net/v3rsch1kk3l1jk'
id='pd1bf839'
to='juliet@example.com/t3hr0zny'
type='set'>
<jingle xmlns='urn:xmpp:jingle:1'
action='session-accept'
initiator='juliet@example.com/t3hr0zny'
responder='romeo@example.net/v3rsch1kk3l1jk'
sid='a73sjjvkla37jfea'>
<content creator='initiator'
media='audio'
name='this-is-the-audio-content'>
<description xmlns='urn:xmpp:jingle:app:rtp:1'>
<payload-type id='97' name='speex' clockrate='8000'/>
</description>
<transport xmlns='urn:xmpp:jingle:transport:raw-udp'>
<candidate ip='192.0.2.101' port='49172' generation='0'/>
</transport>
</content>
</jingle>
</iq>
If the payload types and transport candidate can be successfully used by both parties, then the initiator acknowledges the session-accept action.
<iq from='romeo@example.net/v3rsch1kk3l1jk'
id='pd1bf839'
to='juliet@example.com/t3hr0zny'
type='result'/>
The parties now begin to exchange media. In this case they would exchange audio using the Speex codec at a clockrate of 8000 since that is the highest-priority codec for the responder (as determined by the XML order of the <payloadtype/> children).
The parties may continue the session as long as desired.
Eventually, one of the parties (in this case the responder) terminates the session.
BYE sip:juliet@client.example.com SIP/2.0
Via: SIP/2.0/TCP client.example.net:5060;branch=z9hG4bKnashds7
Max-Forwards: 70
From: Romeo Montague <sip:romeo@example.net>;tag=8321234356
To: Juliet Capulet <sip:juliet@example.com>;tag=9fxced76sl
Call-ID: 3848276298220188511@example.com
CSeq: 1 BYE
Content-Length: 0
The gateway transforms the SIP BYE into XMPP syntax.
<iq from='romeo@example.net/v3rsch1kk3l1jk'
id='rv301b47'
to='juliet@example.com/t3hr0zny'
type='set'>
<jingle xmlns='urn:xmpp:jingle:1'
action='session-terminate'
initiator='juliet@example.com/t3hr0zny'
reasoncode='no-error'
sid='a73sjjvkla37jfea'/>
</iq>
The initiator returns an IQ-result acknowledging receipt of the session termination, which is used only by the gateway and not transformed into SIP syntax.
<iq from='romeo@example.net/v3rsch1kk3l1jk'
id='rv301b47'
to='juliet@example.com/t3hr0zny'
type='result'/>
To follow.
Detailed security considerations for session management are given for SIP in [RFC3261] and for XMPP in [XEP-0166] (see also [RFC6120]).
This document has no actions for the IANA.
| [RFC768] | Postel, J., "User Datagram Protocol", STD 6, RFC 768, August 1980. |
| [RFC793] | Postel, J., "Transmission Control Protocol", STD 7, RFC 793, September 1981. |
| [RFC2616] | Fielding, R., Gettys, J., Mogul, J., Frystyk, H., Masinter, L., Leach, P. and T. Berners-Lee, "Hypertext Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999. |
| [RFC3550] | Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. |