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 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 January 16, 2014. Copyright Notice Copyright (c) 2013 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect Saint-Andre, et al. Expires January 16, 2014 [Page 1] Internet-Draft SIP-XMPP Interworking: Media Sessions July 2013 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. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Compatibility with Offer-Answer model . . . . . . . . . . . . 3 4. Jingle to SIP . . . . . . . . . . . . . . . . . . . . . . . . 3 5. SIP to Jingle . . . . . . . . . . . . . . . . . . . . . . . . 13 6. Security Considerations . . . . . . . . . . . . . . . . . . . 13 7. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 13 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13 9. References . . . . . . . . . . . . . . . . . . . . . . . . . 13 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 15 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 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. Saint-Andre, et al. Expires January 16, 2014 [Page 2] Internet-Draft SIP-XMPP Interworking: Media Sessions July 2013 2. Terminology 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". 3. Compatibility with Offer-Answer model 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. 4. Jingle to SIP 4.1. 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. Saint-Andre, et al. Expires January 16, 2014 [Page 3] Internet-Draft SIP-XMPP Interworking: Media Sessions July 2013 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 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 has traditionally been 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. 4.2. Syntax Mappings 4.2.1. 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 element and its 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. Saint-Andre, et al. Expires January 16, 2014 [Page 4] Internet-Draft SIP-XMPP Interworking: Media Sessions July 2013 In the foregoing example, the syntax and semantics of the and elements are defined in [XEP-0166], the syntax and semantics of the element are defined in [XEP-0167], and the syntax and semantics of the element are defined in [XEP-0177]. Other elements are defined in specifications for the appropriate application types (see for example [XEP-0167]) and other 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 | +--------------------------------+--------------------------------+ | 'action' | [ see next table ] | +--------------------------------+--------------------------------+ | 'initiator' | [ no mapping ] | +--------------------------------+--------------------------------+ | 'responder' | [ no mapping ] | +--------------------------------+--------------------------------+ | 'sid' | local-part of Call-ID | +--------------------------------+--------------------------------+ | local-part of 'initiator' | in SDP o= line | +--------------------------------+--------------------------------+ | 'creator' | [ no mapping ] | +--------------------------------+--------------------------------+ | 'name' | [ no mapping ] | +--------------------------------+--------------------------------+ | 'profile' | in SDP m= line | +--------------------------------+--------------------------------+ | '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'. Saint-Andre, et al. Expires January 16, 2014 [Page 5] Internet-Draft SIP-XMPP Interworking: Media Sessions July 2013 The 'action' attribute of the 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 | +-------------------+-----------------+ 4.2.2. 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] 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. 4.2.3. 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] 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 Saint-Andre, et al. Expires January 16, 2014 [Page 6] Internet-Draft SIP-XMPP Interworking: Media Sessions July 2013 define how to use other RTP profiles such as "RTP/AVPF" and "RTP/ SAVPF" as fedined in RFC4585 and RFC5124 respectively. 4.2.4. 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. 4.2.5. 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], however there are a few syntax incompatibilities which need to be addressed by gateways conforming to this specification: o The 'foundation' attribute is defined as a number in Jingle (unsigned byte) whereas in SIP it's defined as a string which can contain letters, digits and the '+' and '/' symbols. Applications SHOULD convert the foundation element to an integer number. The mechanism for such conversion is undefined. o Jingle defines a 'generation' attribute which is used to determine if an ICE restart is required. Such attribute has no counterpart in SIP as ICE restarts are detected by detecting a change in the ICE ufrag and password. o The 'id' attribute defined by Jingle has no SIP counterpart thus applications are free to choose means to generate unique identifiers across different candidates. o The 'network' attribute defined by Jingle has no counterpart in SIP and SHOULD be ignored. 4.3. 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. Saint-Andre, et al. Expires January 16, 2014 [Page 7] Internet-Draft SIP-XMPP Interworking: Media Sessions July 2013 4.3.1. 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 | | | | 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. Saint-Andre, et al. Expires January 16, 2014 [Page 8] Internet-Draft SIP-XMPP Interworking: Media Sessions July 2013 The gateway returns an XMPP IQ-result to the initiator on behalf of the responder. 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 ;tag=t3hr0zny To: Romeo Montague Call-ID: 3848276298220188511@example.com CSeq: 1 INVITE Contact: Content-Type: application/sdp Content-Length: 184 v=0 Saint-Andre, et al. Expires January 16, 2014 [Page 9] Internet-Draft SIP-XMPP Interworking: Media Sessions July 2013 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 ;tag=t3hr0zny To: Romeo Montague ;tag=v3rsch1kk3l1jk Call-ID: 3848276298220188511@example.com CSeq: 1 INVITE Contact: Content-Length: 0 The gateway transforms the ringing message into XMPP syntax. 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. The responder sends a SIP 200 OK to the initiator. Saint-Andre, et al. Expires January 16, 2014 [Page 10] Internet-Draft SIP-XMPP Interworking: Media Sessions July 2013 SIP/2.0 200 OK Via: SIP/2.0/TCP client.example.com:5060;branch=z9hG4bK74bf9;received=192.0.2.101 From: Juliet Capulet ;tag=t3hr0zny To: Romeo Montague ;tag=v3rsch1kk3l1jk Call-ID: 3848276298220188511@example.com CSeq: 1 INVITE Contact: 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. If the payload types and transport candidate can be successfully used by both parties, then the initiator acknowledges the session-accept action. Saint-Andre, et al. Expires January 16, 2014 [Page 11] Internet-Draft SIP-XMPP Interworking: Media Sessions July 2013 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 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 ;tag=8321234356 To: Juliet Capulet ;tag=9fxced76sl Call-ID: 3848276298220188511@example.com CSeq: 1 BYE Content-Length: 0 The gateway transforms the SIP BYE into XMPP syntax. 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. Saint-Andre, et al. Expires January 16, 2014 [Page 12] Internet-Draft SIP-XMPP Interworking: Media Sessions July 2013 5. SIP to Jingle To follow. 6. Security Considerations Detailed security considerations for session management are given for SIP in [RFC3261] and for XMPP in [XEP-0166] (see also [RFC6120]). 7. Open Issues o Better text for OA compatibility section o Define how to handle session-info stanzas with 'active', 'hold' and 'mute' elements. Map that to SIP hold. o Translation of a=fmtp: SDP does not mandate to use a semicolon- separated list of values. 8. IANA Considerations This document has no actions for the IANA. 9. References 9.1. Normative References [I-D.ietf-stox-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-ietf-stox-core-00 (work in progress), July 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. Saint-Andre, et al. Expires January 16, 2014 [Page 13] Internet-Draft SIP-XMPP Interworking: Media Sessions July 2013 [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] 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. 9.2. Informative References [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. [RFC768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, August 1980. [RFC793] Postel, J., "Transmission Control Protocol", STD 7, RFC 793, September 1981. Saint-Andre, et al. Expires January 16, 2014 [Page 14] Internet-Draft SIP-XMPP Interworking: Media Sessions July 2013 Authors' Addresses Peter Saint-Andre Cisco Systems, Inc. 1899 Wynkoop Street, Suite 600 Denver, CO 80202 USA Phone: +1-303-308-3282 Email: psaintan@cisco.com Saul Ibarra Corretge AG Projects Dr. Leijdsstraat 92 Haarlem 2021RK The Netherlands Email: saul@ag-projects.com Emil Ivov Jitsi Strasbourg 67000 France Phone: +33-177-624-330 Email: emcho@jitsi.org Saint-Andre, et al. Expires January 16, 2014 [Page 15]