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Versions: 00 01 02 draft-ietf-stox-media

Network Working Group                                     P. Saint-Andre
Internet-Draft                                       Cisco Systems, Inc.
Intended status: Informational                         February 20, 2013
Expires: August 24, 2013


   Interworking between the Session Initiation Protocol (SIP) and the
   Extensible Messaging and Presence Protocol (XMPP): Media Sessions
                   draft-saintandre-sip-xmpp-media-02

Abstract

   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).

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on August 24, 2013.

Copyright Notice

   Copyright (c) 2013 IETF Trust and the persons identified as the
   document authors.  All rights reserved.












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   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  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Jingle to SIP . . . . . . . . . . . . . . . . . . . . . . . .   3
     2.1.  Overview  . . . . . . . . . . . . . . . . . . . . . . . .   3
     2.2.  Syntax Mappings . . . . . . . . . . . . . . . . . . . . .   3
     2.3.  Sample Scenarios  . . . . . . . . . . . . . . . . . . . .   6
   3.  SIP to Jingle . . . . . . . . . . . . . . . . . . . . . . . .  12
   4.  Security Considerations . . . . . . . . . . . . . . . . . . .  12
   5.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  12
   6.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  12
     6.1.  Normative References  . . . . . . . . . . . . . . . . . .  12
     6.2.  Informative References  . . . . . . . . . . . . . . . . .  13
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  13

1.  Introduction

   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 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 [I-D.saintandre-sip-xmpp-core] and other
   specifications in that "series"), this document describes a bi-



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   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 [RFC2119].

2.  Jingle to SIP

2.1.  Overview

   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:

   o  Base SIP messages and headers use a plaintext format similar in
      some ways to the Hypertext Transport Protocol [RFC2616], whereas
      Jingle messages are pure XML.  Mappings between SIP headers and
      Jingle message syntax are provided below.

   o  The SIP payloads defining session semantics use the Session
      Description Protocol [RFC4566], whereas the equivalent Jingle
      payloads are defined as XML child elements of the Jingle <content/
      > element.  However, the Jingle specifications defining such child
      elements specify mappings to SDP for all Jingle syntax, making the
      mapping relatively straightforward.

   o  The SIP signalling channel is transported over UDP, whereas the
      signalling channel for Jingle is XMPP over TCP.  Mapping between
      the transport layers typically happens within a gateway using
      techniques below the application level, and therefore is not
      addressed in this specification.

2.2.  Syntax Mappings

2.2.1.  Generic Jingle Syntax

   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.




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   <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 ]                 |



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   +--------------------------------+--------------------------------+
   | <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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2.2.2.  Audio Application Format

   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], where the media type is "audio"
   and the specific mappings to SDP syntax are provided in [XEP-0167].

2.2.3.  Video Application Format

   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], where the media type is "audio"
   and the specific mappings to SDP syntax are provided in [XEP-0167].

2.2.4.  Raw UDP Transport Method

   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.

2.2.5.  ICE-UDP Transport Method

   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].

2.3.  Sample Scenarios

   The following sections provide sample scenarios (or "call flows")
   that illustrate the principles of interworking from Jingle to SIP.
   These scenarios are not exhaustive.

2.3.1.  Basic Voice Chat

   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
     |                        |                       |



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     | 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'



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                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





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   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>



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   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'



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       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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3.  SIP to Jingle

   To follow.

4.  Security Considerations

   Detailed security considerations for session management are given for
   SIP in [RFC3261] and for XMPP in [XEP-0166] (see also [RFC6120]).

5.  IANA Considerations

   This document has no actions for the IANA.

6.  References

6.1.  Normative References

   [I-D.saintandre-sip-xmpp-core]
              Saint-Andre, P., Houri, A., and J. Hildebrand,
              "Interworking between the Session Initiation Protocol
              (SIP) and the Extensible Messaging and Presence Protocol
              (XMPP): Core", draft-saintandre-sip-xmpp-core-03 (work in
              progress), February 2013.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3261]  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.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245, April
              2010.

   [RFC6120]  Saint-Andre, P., "Extensible Messaging and Presence
              Protocol (XMPP): Core", RFC 6120, March 2011.

   [XEP-0166]



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              Ludwig, S., Beda, J., Saint-Andre, P., McQueen, R., Egan,
              S., and J. Hildebrand, "Jingle", XSF XEP 0166, June 2007.

   [XEP-0167]
              Ludwig, S., Saint-Andre, P., Egan, S., and R. McQueen,
              "Jingle RTP Sessions", XSF XEP 0167, February 2009.

   [XEP-0176]
              Beda, J., Ludwig, S., Saint-Andre, P., Hildebrand, J., and
              S. Egan, "Jingle ICE-UDP Transport Method", XSF XEP 0176,
              February 2009.

   [XEP-0177]
              Beda, J., Saint-Andre, P., Ludwig, S., Hildebrand, J., and
              S. Egan, "Jingle Raw UDP Transport", XSF XEP 0177,
              February 2009.

6.2.  Informative References

   [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.

Author's Address

   Peter Saint-Andre
   Cisco Systems, Inc.
   1899 Wynkoop Street, Suite 600
   Denver, CO  80202
   USA

   Phone: +1-303-308-3282
   Email: psaintan@cisco.com







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