Abstract: | This document describes implementation considerations related to audio codecs for use in Jingle RTP sessions. |
Author: | Peter Saint-Andre |
Copyright: | © 1999 - 2011 XMPP Standards Foundation. SEE LEGAL NOTICES. |
Status: | Experimental |
Type: | Standards Track |
Version: | 0.6 |
Last Updated: | 2011-06-27 |
WARNING: This Standards-Track document is Experimental. Publication as an XMPP Extension Protocol does not imply approval of this proposal by the XMPP Standards Foundation. Implementation of the protocol described herein is encouraged in exploratory implementations, but production systems are advised to carefully consider whether it is appropriate to deploy implementations of this protocol before it advances to a status of Draft.
1. Introduction
2. Basic Considerations
3. Codecs
3.1. G.711
3.2. Opus
3.3. Speex
4. Guidance for Implementers
5. Mandatory-to-Implement Codecs
6. Security Considerations
7. IANA Considerations
8. XMPP Registrar Considerations
9. Acknowledgements
Appendices
A: Document Information
B: Author Information
C: Legal Notices
D: Relation to XMPP
E: Discussion Venue
F: Requirements Conformance
G: Notes
H: Revision History
Jingle RTP Sessions [1] defines the Jingle [2] signalling exchanges needed to establish voice chat and other audio sessions using the Real-time Transport Protocol RFC 3550 [3]; however, it does not say which audio codecs are mandatory-to-implement, since the state of codec technologies is more fluid than the signalling interactions. This document fills that gap by providing guidance to Jingle developers regarding audio codecs.
Because codec technologies are typically subject to patents, the topics discussed here are controversial. This document attempts to steer a middle path between (1) specifying mandatory-to-implement technologies that realistically will not be implemented and deployed and (2) providing guidelines that, while realistic, do not encourage the implementation and deployment of patent-clear technologies.
The ideal audio codec would meet the following criteria:
Unfortunately, not all codecs meet those criteria. In the remainder of this document we discuss the audio codecs that are most appropriate for implementation in Jingle RTP applications.
This section is non-normative. Future versions of this specification might provide information about additional codecs not listed here.
G.711 refers to the Pulse Code Modulation (PCM) codec defined in International Telecommunication Union (ITU) [5] recommendation G.711, which is widely used on the public switched telephone network (PSTN) and by many voice over Internet Protocol (VoIP) providers. There are two versions: the μ-law ("U-law") version is widely deployed in North America and in Japan, whereas the A-law version is widely deployed in the rest of the world. The following table summarizes the available information about G.711.
The Opus codec is under development within the IETF's Codec Working Group. In essence it combines the best features of CELT (developed by Jean-Marc Valin, the creator of Speex) and SILK (created by and widely used in the Skype service). The following table summarizes the available information about Opus.
Quality | Packetization | Availability | Patents |
---|---|---|---|
Extremely high quality; can be used for wide-band audio. | See RTP Payload Format and File Storage Format for Opus Speech and Audio Codec [7]. | Covered under IETF IPR rules, the intent is for the codec to be covered under a simplified BSD license. See http://tools.ietf.org/html/draft-ietf-codec-opus for details. Not commonly deployed yet, but the SILK codec on which it is partly based is very widely deployed. | Designed to be patent-clear, but IPR claims have been filed. |
According to the speex.org website, the Speex codec is "an Open Source/Free Software patent-free audio compression format designed for speech". Speex was developed by Jean-Marc Valin and is maintained by the Xiph.org Foundation. The following table summarizes the available information about Speex.
Quality | Packetization | Availability | Patents |
---|---|---|---|
Good quality; optimized for voice; can be used for wide-band audio. | See RFC 5574 [8]. | Freely downloadable under a revised BSD license at <http://speex.org/> and commonly deployed on Internet (VoIP) systems; not commonly deployed on non-Internet systems. | Designed to be patent-clear. |
This section is non-normative.
Given that both Speex and G.711 are patent-clear, freely implementable, and commonly deployed, implementers are encouraged to consider including support for both codecs in audio applications of Jingle RTP sessions. Discussion on the jingle@xmpp.org mailing list indicates a slight preference for G.711 because it is easily available and so widely deployed (e.g., in SIP networks and the PSTN). The Opus codec is not yet widely deployed (or even fully developed), but it might become the "codec of the future" for audio applications over the Internet.
As of June 2011, this document makes the following recommendations:
Naturally, clients and gateways can implement additional codecs, such as those listed in this document.
For security considerations related to Jingle RTP sessions, refer to XEP-0167. This document introduces no new security considerations. See also the security considerations described in the relevant codec specifications.
This document requires no interaction with the Internet Assigned Numbers Authority (IANA) [9].
This document requires no interaction with the XMPP Registrar [10].
Thanks to Olivier Crête, Dave Cridland, Florian Jensen, Justin Karneges, Evgeniy Khramtsov, Marcus Lundblad, Tobias Markmann, Pedro Melo, Jack Moffitt, Jeff Muller, Jehan Pagès, Arc Riley, Kevin Smith, Remko Tronçon, Justin Uberti, and Paul Witty for their feedback.
Series: XEP
Number: 0266
Publisher: XMPP Standards Foundation
Status:
Experimental
Type:
Standards Track
Version: 0.6
Last Updated: 2011-06-27
Approving Body: XMPP Council
Dependencies: XMPP Core, XEP-0167
Supersedes: None
Superseded By: None
Short Name: N/A
Source Control:
HTML
This document in other formats:
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PDF
Email:
stpeter@jabber.org
JabberID:
stpeter@jabber.org
URI:
https://stpeter.im/
The Extensible Messaging and Presence Protocol (XMPP) is defined in the XMPP Core (RFC 3920) and XMPP IM (RFC 3921) specifications contributed by the XMPP Standards Foundation to the Internet Standards Process, which is managed by the Internet Engineering Task Force in accordance with RFC 2026. Any protocol defined in this document has been developed outside the Internet Standards Process and is to be understood as an extension to XMPP rather than as an evolution, development, or modification of XMPP itself.
There exists a special venue for discussion related to the technology described in this document: the <jingle@xmpp.org> mailing list.
The primary venue for discussion of XMPP Extension Protocols is the <standards@xmpp.org> discussion list.
Discussion on other xmpp.org discussion lists might also be appropriate; see <http://xmpp.org/about/discuss.shtml> for a complete list.
Errata can be sent to <editor@xmpp.org>.
The following requirements keywords as used in this document are to be interpreted as described in RFC 2119: "MUST", "SHALL", "REQUIRED"; "MUST NOT", "SHALL NOT"; "SHOULD", "RECOMMENDED"; "SHOULD NOT", "NOT RECOMMENDED"; "MAY", "OPTIONAL".
1. XEP-0167: Jingle RTP Sessions <http://xmpp.org/extensions/xep-0167.html>.
2. XEP-0166: Jingle <http://xmpp.org/extensions/xep-0166.html>.
3. RFC 3550: RTP: A Transport Protocol for Real-Time Applications <http://tools.ietf.org/html/rfc3550>.
4. The term patent-clear does not necessarily mean that no patents have ever been applied for or granted regarding a technology, or that the technology is completely free from patents (since such a judgment is nearly impossible to make, and is outside the purview of the XMPP developer community and the XMPP Standards Foundation); the term means only that those who implement the technology are generally understood to be relatively safe from the threat of patent litigation, either because any relevant patents have expired, were filed in a defensive manner, or are made available under suitable royalty-free licenses.
5. The International Telecommunication Union develops technical and operating standards (such as H.323) for international telecommunication services. For further information, see <http://www.itu.int/>.
6. RFC 5391: RTP Payload Format for ITU-T Recommendation G.711.1 <http://tools.ietf.org/html/rfc5391>.
7. RTP Payload Format and File Storage Format for Opus Speech and Audio Codec <http://tools.ietf.org/html/draft-spittka-payload-rtp-opus>. Work in progress.
8. RFC 5574: RTP Payload Format for the Speex Codec <http://tools.ietf.org/html/rfc5574>.
9. The Internet Assigned Numbers Authority (IANA) is the central coordinator for the assignment of unique parameter values for Internet protocols, such as port numbers and URI schemes. For further information, see <http://www.iana.org/>.
10. The XMPP Registrar maintains a list of reserved protocol namespaces as well as registries of parameters used in the context of XMPP extension protocols approved by the XMPP Standards Foundation. For further information, see <http://xmpp.org/registrar/>.
Note: Older versions of this specification might be available at http://xmpp.org/extensions/attic/
Clarified that the codec descriptions are non-normative.
(psa)Moved video codecs to XEP-0299.
(psa)Recommended G.711 as mandatory-to-implement for audio.
(psa)Added information about the Opus audio codec.
(psa)Added information about the Dirac video codec.
(psa)Initial published version.
(psa)Clarified status of H.264.
(psa)Rewrote document based on developer feedback and Council discussion.
(psa)Added more information about video codecs.
(psa)First draft, copied from XEP-0167 with slight revisions and addition of requirements section.
(psa)END