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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).
1.
Introduction
2.
Jingle to SIP
2.1.
Overview
2.2.
Syntax Mappings
2.3.
Sample Scenarios
3.
SIP to Jingle
4.
Security Considerations
5.
References
5.1.
Normative References
5.2.
Informative References
§
Author's Address
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The Session Initiation Protocol [SIP] (Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, “SIP: Session Initiation Protocol,” June 2002.) 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 [UDP] (Postel, J., “User Datagram Protocol,” August 1980.)), 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 [RTP] (Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications,” July 2003.).
The Extensible Messaging and Presence Protocol [XMPP] (Saint-Andre, P., “Extensible Messaging and Presence Protocol (XMPP): Core,” October 2004.) also provides a signalling channel, typically via the Transmission Control Protocol [TCP] (Postel, J., “Transmission Control Protocol,” September 1981.). 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 [JINGLE] (Ludwig, S., Beda, J., Saint-Andre, P., McQueen, R., Egan, S., and J. Hildebrand, “Jingle,” June 2007.).
However, Jingle has been 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 [SIP‑XMPP] (Saint-Andre, P., Houri, A., and J. Hildebrand, “Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Core,” March 2009.) and other specifications in that "series"), 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.
Note: The capitalized 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 RFC 2119 (Bradner, S., “Key words for use in RFCs to Indicate Requirement Levels,” March 1997.) [TERMS].
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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:
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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 [JINGLE] (Ludwig, S., Beda, J., Saint-Andre, P., McQueen, R., Egan, S., and J. Hildebrand, “Jingle,” June 2007.), the syntax and semantics of the <description/> element are defined in [JINGLE‑RTP] (Ludwig, S., Saint-Andre, P., Egan, S., and R. McQueen, “Jingle RTP Sessions,” February 2009.), and the syntax and semantics of the <transport/> element are defined in [JINGLE‑UDP] (Beda, J., Saint-Andre, P., Ludwig, S., Hildebrand, J., and S. Egan, “Jingle Raw UDP Transport,” February 2009.). Other <description/> elements are defined in specifications for the appropriate application types (see for example [JINGLE‑RTP] (Ludwig, S., Saint-Andre, P., Egan, S., and R. McQueen, “Jingle RTP Sessions,” February 2009.)) and other <transport/> elements are defined in the specifications for appropriate transport methods (see for example [JINGLE‑ICE] (Beda, J., Ludwig, S., Saint-Andre, P., Hildebrand, J., and S. Egan, “Jingle ICE-UDP Transport Method,” February 2009.), which defines an XMPP profile of [ICE] (Rosenberg, J., “Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols,” October 2007.)).
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 '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) | +-------------------+-----------------+ | 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 | [varies] | +-------------------+-----------------+
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A Jingle application format for audio exchange via RTP is specified in [JINGLE‑RTP] (Ludwig, S., Saint-Andre, P., Egan, S., and R. McQueen, “Jingle RTP Sessions,” February 2009.). This application format effectively maps to the "RTP/AVP" profile specified in [RTP‑AVP] (Schulzrinne, H. and S. Casner, “RTP Profile for Audio and Video Conferences with Minimal Control,” July 2003.), where the media type is "audio" and the specific mappings to SDP syntax are provided in [JINGLE‑RTP] (Ludwig, S., Saint-Andre, P., Egan, S., and R. McQueen, “Jingle RTP Sessions,” February 2009.).
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A Jingle application format for video exchange via RTP is specified in [JINGLE‑RTP] (Ludwig, S., Saint-Andre, P., Egan, S., and R. McQueen, “Jingle RTP Sessions,” February 2009.). This application format effectively maps to the "RTP/AVP" profile specified in [RTP‑AVP] (Schulzrinne, H. and S. Casner, “RTP Profile for Audio and Video Conferences with Minimal Control,” July 2003.), where the media type is "audio" and the specific mappings to SDP syntax are provided in [JINGLE‑RTP] (Ludwig, S., Saint-Andre, P., Egan, S., and R. McQueen, “Jingle RTP Sessions,” February 2009.).
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A basic Jingle transport method for exchanging media over UDP is specified in [JINGLE‑UDP] (Beda, J., Saint-Andre, P., Ludwig, S., Hildebrand, J., and S. Egan, “Jingle Raw UDP Transport,” February 2009.). 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.
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A more advanced Jingle transport method for exchanging media over UDP is specified in [JINGLE‑ICE] (Beda, J., Ludwig, S., Saint-Andre, P., Hildebrand, J., and S. Egan, “Jingle ICE-UDP Transport Method,” February 2009.). Under ideal conditions this transport method provides NAT traversal by following the Interactive Connectivity Exchange methodology specified in [ICE] (Rosenberg, J., “Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols,” October 2007.). The relevant SDP mappings are provided in [JINGLE‑ICE] (Beda, J., Ludwig, S., Saint-Andre, P., Hildebrand, J., and S. Egan, “Jingle ICE-UDP Transport Method,” February 2009.).
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The following sections provide sample scenarios (or "call flows") that illustrate the principles of interworking from Jingle to SIP. These scenarios are not exhaustive.
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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 [JINGLE‑UDP] (Beda, J., Saint-Andre, P., Ludwig, S., Hildebrand, J., and S. Egan, “Jingle Raw UDP Transport,” February 2009.).
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 0 a=rtpmap:96 SPEEX/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:app:rtp:1-info'/>
</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 0 a=rtpmap:97 SPEEX/8000 a=rtpmap:18 G729/8000
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'/>
<payload-type id='18' name='G729'/>
<payload-type id='0' name='PCMU' 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'/>
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To follow.
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Detailed security considerations for session management are given for SIP in [SIP] (Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, “SIP: Session Initiation Protocol,” June 2002.) and for XMPP in [JINGLE] (Ludwig, S., Beda, J., Saint-Andre, P., McQueen, R., Egan, S., and J. Hildebrand, “Jingle,” June 2007.) (see also [XMPP] (Saint-Andre, P., “Extensible Messaging and Presence Protocol (XMPP): Core,” October 2004.)).
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| [HTTP] | 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 (TXT, PS, PDF, HTML, XML). |
| [RTP] | Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications,” STD 64, RFC 3550, July 2003 (TXT, PS, PDF). |
| [TCP] | Postel, J., “Transmission Control Protocol,” STD 7, RFC 793, September 1981 (TXT). |
| [UDP] | Postel, J., “User Datagram Protocol,” STD 6, RFC 768, August 1980 (TXT). |
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| Peter Saint-Andre | |
| Cisco | |
| Email: | psaintan@cisco.com |