[speex-dev] First draft for Speex RTP profile - Please send your comments

Jean-Marc Valin jean-marc.valin at hermes.usherb.ca
Tue Oct 22 18:32:22 PDT 2002


We'd like to announce the first draft for the Speex RTP profile. It was
written essentially by Greg Herlein, with some help from Simon Morlat
and I. We'd like to get some feedback on it before it is sent to the
IETF. Basically this will allow all SIP based VoIP applications using
Speex to inter-operate. For those interested, there's already Simon's
LinPhone (www.linphone.org) VoIP phone that supports Speex (not yet
compliant with the current draft though).

Anyway, please read the draft in attachment and send you comments so we
can improve it.


Jean-Marc Valin, M.Sc.A.
LABORIUS (http://www.gel.usherb.ca/laborius)
Université de Sherbrooke, Québec, Canada

Network Working Group                  Audio-Video Transport Working Group
Internet Draft			                             Herlein/Valin
draft-herlein-speex-rtp-profile-03	                  October 22, 2002
                                                   Expires: April 22. 2003

<p>          RTP Payload Format for the Speex Codec

Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
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Copyright Notice

   Copyright (C) The Internet Society (2001).  All Rights Reserved.


Speex is an open-source, patent-free voice codec suitable for use in
Voice over IP (VoIP) type applications.  The Speex codec supports two
modes of operation: narrowband at a nominal 8kHz sample rate and
wideband at a nominal 16kHz sample rate.  Speex supports Voice
Activity Detection (VAD) and Variable Bit Rate (VBR).  This document
describes the payload format for Speex generated bit streams within an
RTP packet.  Also included here are the necessary details for the use
of Speex with MIME and SDP.

<p>1. Conventions used in this document

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC-2119 [6].

2. Overview of the Speex Codec

Speex is based on the CELP encoding technique with support for either
wideband (nominal 16kHz) or narrowband (nominal 8kHz) sampling.  The
main characteristics can be summerized as follows:

   o  Free software/open-source, royalty-free
   o  Integration of wideband and narrowband in the same bit-stream
   o  Wide range of bit-rates available
   o  Dynamic bit-rate switching and variable bit-rate (VBR)
   o  Voice Activity Detection (VAD, integrated with VBR)
   o  Variable complexity

<p>3. RTP payload format for Speex

   Speex uses 20 ms frames and a variable sampling rate clock.  The
   RTP timestamp MUST be in units of 1/X of a second where X is the
   sample rate used.  Speex uses a nominal 8kHz sampling rate for
   narrowband use and a nominal 16kHz sampling rate for wideband use.

   The RTP payload for Speex has the format shown in Figure 1.  No
   additional header specific to this payload format is required.

       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      |                      RTP Header [3]                           |
      |                                                               |
      +                 one or more frames of Speex                   |
      |                             ....                            |p|

                     Figure 1: RTP payload for Speex

   The encoding and decoding algorithm can change the bit rate at any
   20ms frame boundary but the bit rate change notification is provided
   in-band with the bit stream.  Each frame contains both 
   "mode" (narrowband or wideband) and "sub-mode" (bit-rate)
   information in the bit stream.  No out-of-band notification is
   required for the decoder to process changes in the bit rate sent by
   the encoder.

   For the purposes of packetizing the bit stream in RTP, it is only
   necessary to consider the sequence of bits as output by the Speex
   encoder, and present the same sequence to the decoder.  The payload
   format described here maintains this sequence.

   An RTP packet MAY contain Speex frames of the same bit rate or of
   varying bit rates, since the bit-rate for a frame is conveyed in
   band with the signal.

   It is RECOMMENDED that values of 8000 or 16000 be used for normal
   internet telephony applications, though the sample rate is
   supported at rates as low as 6000 Hz and as high as 22.05 kHz.

   The RTP payload MUST be padded to provide an integer number of
   octets as the payload length.  These padding bits MUST be all zero.
   This padding is only required for the last frame in the packet, and
   only to ensure the packet contents ends on an octet boundary.

<p>3.1 Multiple Speex frames in a RTP packet

   By default only one Speex frame is permitted in a single RTP
   packet.  When operating with multiple frames per packet then the
   end points MUST use out-of-band negotiation to determine the number
   of frames per packet.  See section 5 below for an example of how to
   do this with SDP [5].

3.2 Computing the number of Speex frames

   If using SDP [5] (see section 5 below for an example) this can be
   done using the "ptime" variable to denote the packetization
   interval (ie, how many milliseconds of audio is encoded in a single
   RTP packet).  Since Speex uses 20ms frames, ptime values of
   multiples of 20 denote multiple Speex frames per packet.  Values of
   ptime in other than multiples of 20 SHOULD be ignored and SHOULD
   use the default value of one instead.

<p>4. MIME registration of Speex

   MIME media type name: audio

   MIME subtype: SPX

   Required parameters:

<p>   Optional parameters:

         ptime: RECOMMENDED duration of each packet in milliseconds.
         mode:  RECOMMENDED speex encoding mode. Can be {1,2,3,4,5,any}
         penh:	RECOMMENDED use of perceptual enhancement. 1 indicates 
                 to the decoder that perceptual enhancement is recommended,
                0 indicates that it is not.

   Encoding considerations:
         This type is only defined for transfer via RTP as specified in
         a Work in Progress.

   Security Considerations:
         See Section 6 of RFC 3047.

   Interoperability considerations: none

   Published specification:

   Applications which use this media type:
         Audio and video streaming and conferencing tools

   Additional information: none

   Person & email address to contact for further information:

   Intended usage: COMMON

   Author/Change controller:
         Change controller:

<p>5. SDP usage of Speex

   When conveying information by SDP [5], the encoding name SHALL be
   "SPX" (the same as the MIME subtype).  An example of the media
   representation in SDP for offering a single channel of Speex at
   8000 samples per second might be:

        m=audio 8088 RTP/AVP 97
        a=rtpmap:97 SPX/8000

   Note that the sampling frequency is given on the a=rtpmap line. Its 
   value is typically 8000 for narrow band operation and 16000 for
   wide band operation.

   If for some reason the offerer has bandwith limitations, he may use
   the "b=" header, as explained in SDP [5]. The following example
   illustrates the case where the offerer cannot receive more than
   10 kbit/s.

           m=audio 8088 RTP/AVP 97
        a=rtmap:97 SPX/8000

   In this case, if the remote part agrees, it should configure its
   speex encoder so that it does not use modes that produce more than
   10 kbit/s. Note that the "b=" constraint also applies on all
   payload types that may be proposed in the media line ("m=").

   An other way to make recommendations to the remote speex encoder
   is to use its specific parameters:

           m=audio 8008 RTP/AVP 97
        a=rtpmap:97 SPX/8000
        a=fmtp:97 mode=3

   This examples illustrate an offerer that wishes to receive
   a speex stream at 8000Hz, but only using speex mode 3.
   The offerer may suggest to the remote decoder to activate
   its perceptual enhancement filter like this:
        m=audio 8088 RTP/AVP 97
        a=rtmap:97 SPX/8000
        a=fmtp:97 penh=1
   Several speex specific parameters can be given in a single
   a=fmtp line provided that they are separated by a semi-colon:
           a=fmtp:97 mode=any;penh=1

   Finally the use of a particular packetization interval may be
   suggested to the remote encoder using the ptime parameter:
           m=audio 8008 RTP/AVP 97
        a=rtpmap:97 SPX/8000
   Note that the ptime parameter applies to all payloads listed
   in the media line. 

   Speex can encode frames of 20 ms. Values of ptime not multiple
   of 20 ms are meaningless, so the receiver of such ptime values
   SHOULD ignore them.

<p>6. Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the security considerations discussed in the RTP
   specification [3], and any appropriate RTP profile.  This implies
   that confidentiality of the media streams is achieved by encryption.
   Because the data compression used with this payload format is applied
   end-to-end, encryption may be performed after compression so there is
   no conflict between the two operations.

   A potential denial-of-service threat exists for data encodings using
   compression techniques that have non-uniform receiver-end
   computational load.  The attacker can inject pathological datagrams
   into the stream which are complex to decode and cause the receiver to
   be overloaded.  However, this encoding does not exhibit any
   significant non-uniformity.

   As with any IP-based protocol, in some circumstances a receiver may
   be overloaded simply by the receipt of too many packets, either
   desired or undesired.  Network-layer authentication may be used to
   discard packets from undesired sources, but the processing cost of
   the authentication itself may be too high.  In a multicast
   environment, pruning of specific sources may be implemented in future
   versions of IGMP [7] and in multicast routing protocols to allow a
   receiver to select which sources are allowed to reach it.

<p>7. References

   1. Bradner, S., "The Internet Standards Process -- Revision 3", BCP
      9, RFC 2026, October 1996.

   2. ITU-T Recommendation G.722.1, available online from the ITU
      bookstore at http://www.itu.int.

   3. Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP:
      A Transport Protocol for real-time applications", RFC 1889,
      January 1996.  (Updated by a Work in Progress.)

   4. Freed, N. and N. Borenstein, "Multipurpose Internet Mail
      Extensions (MIME) Part One: Format of Internet Message Bodies",
      RFC 2045, November 1996.

   5. Handley, M. and V. Jacobson, "SDP: Session Description Protocol",
      RFC 2327, April 1998.

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

   7. Deering, S., "Host Extensions for IP Multicasting", STD 5, RFC
      1112, August 1989.

8. Acknowledgments

9. Author's Address

<p>10. Full Copyright Statement

   Copyright (C) The Internet Society (2001).  All Rights Reserved.

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   The limited permissions granted above are perpetual and will not be
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   This document and the information contained herein is provided on an


   Funding for the RFC Editor function is currently provided by the
   Internet Society.

<p><p><p><p><p><p><p><p><p><p><p><p><p><p><p>Herlein                      Internet Draft                   [Page 6]

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