Internet DRAFT - draft-herlein-avt-rtp-speex


Internet Engineering Task Force                          Greg Herlein
Internet Draft                                        Jean-Marc Valin
draft-herlein-avt-rtp-speex-00.txt                       Simon Morlat
March 3, 2004                                          Roger Hardiman
Expires: September 3, 2004                                  Phil Kerr

                 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 RFC 2026.

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Copyright Notice

   Copyright (C) The Internet Society (2003).  All Rights Reserved.


   Speex is an open-source voice codec suitable for use in Voice over
   IP (VoIP) type applications.  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
   the Session Description Protocol (SDP) and a preliminary method of
   using Speex within H.323 applications.

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 [5].

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2. Overview of the Speex Codec

   Speex is based on the CELP [12] encoding technique with support for 
   either narrowband (nominal 8kHz), wideband (nominal 16kHz) or 
   ultra-wideband (nominal 32kHz), and (non-optimal) rates up to 48 kHz 
   sampling also available.  The main characteristics can be summarized 
   as follows:

   o  Free software/open-source
   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

3. RTP payload format for Speex

   For RTP based transportation of Speex encoded audio the standard 
   RTP header [2] is followed by one or more payload data blocks. 
   An optional padding terminator may also be used. 

      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                            |
     |                 one or more frames of Speex ....              |
     |        one or more frames of Speex ....       |    padding    |

3.1 RTP Header

      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
     |V=2|P|X|  CC   |M|     PT      |       sequence number         |
     |                           timestamp                           |
     |           synchronization source (SSRC) identifier            |
     |            contributing source (CSRC) identifiers             |
     |                              ...                              |

   The RTP header begins with an octet of fields (V, P, X, and CC) to   
   support specialized RTP uses (see [8] and [9] for details). For 
   Speex the following values are used.

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   Version (V): 2 bits
      This field identifies the version of RTP. The version
      used by this specification is two (2).

   Padding (P): 1 bit
      If the padding bit is set, the packet contains one or more
      additional padding octets at the end which are not part of
      the payload.  P is set if the total packet size is less than 
      the MTU.  

   Extension (X): 1 bit
      If the extension, X, bit is set, the fixed header MUST be 
      followed by exactly one header extension, with a format defined 
      in Section 5.3.1. of [8], 

   CSRC count (CC): 4 bits
      The CSRC count contains the number of CSRC identifiers.

   Marker (M): 1 bit
      The M bit indicates if the packet contains comfort noise.  This 
      field is used in conjunction with the cng SDP attribute and is 
      detailed further in section 5 below.  In normal usage this bit 
      is set if the packet contains comfort noise.

   Payload Type (PT): 7 bits
      An RTP profile for a class of applications is expected to assign 
      a payload type for this format, or a dynamically allocated 
      payload type SHOULD be chosen which designates the payload as 

   Sequence number: 16 bits
      The sequence number increments by one for each RTP data packet
      sent, and may be used by the receiver to detect packet loss and
      to restore packet sequence.  This field is detailed further in

   Timestamp: 32 bits
      A timestamp representing the sampling time of the first sample of
      the first Speex packet in the RTP packet.  The clock frequency 
      MUST be set to the sample rate of the encoded audio data.

      Speex uses 20 msec 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, a nominal 16kHz sampling rate for wideband use, 
      and a nominal 32kHz sampling rate for ultra-wideband use.

   SSRC/CSRC identifiers: 
      These two fields, 32 bits each with one SSRC field and a maximum 
      of 16 CSRC fields, are as defined in [2].  

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3.2 Speex payload

   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 [11], and present the same sequence to the decoder.  The 
   payload format described here maintains this sequence.  

   A typical Speex frame, encoded at the maximum bitrate, is approx.
   110 octets and the total number of Speex frames SHOULD be kept 
   less than the path MTU to prevent fragmentation.  Speex frames MUST
   NOT be fragmented across multiple RTP packets,

   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.

   The encoding and decoding algorithm can change the bit rate at any
   20 msec frame boundary, with the bit rate change notification provided
   in-band with the bit stream.  Each frame contains both "mode" 
   (narrowband, wideband or ultra-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.

   It is RECOMMENDED that values of 8000, 16000 and 32000 be used 
   for normal internet telephony applications, though the sample 
   rate is supported at rates as low as 6000 Hz and as high as 
   48 kHz.

   The RTP payload MUST be padded to provide an integer number of
   octets as the payload length.  These padding bits are LSB aligned
   in network byte order and consist of a 0 followed by all ones
   (until the end of the octet).  This padding is only required for
   the last frame in the packet, and only to ensure the packet
   contents ends on an octet boundary.

3.2.1 Example Speex packet

   In the example below we have a single Speex frame with 5 bits
   of padding to ensure the packet size falls on an octet boundary.

    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
   |V=2|P|X|  CC   |M|     PT      |       sequence number         |
   |                           timestamp                           |
   |         synchronization source (SSRC) identifier              |

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    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
   |         contributing source (CSRC) identifiers                |
   |                              ...                              |
   |                        ..speex data..                         |
   |                        ..speex data..               |0 1 1 1 1|

3.4 Multiple Speex frames in a RTP packet

   Below is an example of two Speex frames contained within one RTP 
   packet.  The Speex frame length in this example fall on an octet
   boundary so there is no padding.

   Speex codecs [11] are able to detect the the bitrate from the
   payload and are responsible for detecting the 20 msec boundaries
   between each frame.

    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
   |V=2|P|X|  CC   |M|     PT      |       sequence number         |
   |                           timestamp                           |
   |         synchronization source (SSRC) identifier              |
   |         contributing source (CSRC) identifiers                |
   |                              ...                              |
   |                        ..speex data..                         |
   |        ..speex data..         |        ..speex data..         |
   |                        ..speex data..                         |

4. MIME registration of Speex

   Full definition of the MIME type for Speex will be part of the Ogg
   Vorbis MIME type definition application [10].  

   MIME media type name: audio

   MIME subtype: speex

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   Optional parameters:

   Required parameters: to be included in the Ogg MIME specification.

   Encoding considerations:

   Security Considerations:
         See Section 6 of RFC 3047.

   Interoperability considerations: none

   Published specification:  

   Applications which use this media type:

   Additional information: none

   Person & email address to contact for further information:
	 Greg Herlein <>
	 Jean-Marc Valin <>

   Intended usage: COMMON

   Author/Change controller:
         Author:  Greg Herlein <>
         Change controller: Greg Herlein <>

   This transport type signifies that the content is to be interpreted
   according to this document if the contents are transmitted over RTP. 
   Should this transport type appear over a lossless streaming protocol
   such as TCP, the content encapsulation should be interpreted as an 
   Ogg Stream in accordance with RFC 3534, with the exception that the
   content of the Ogg Stream may be assumed to be Speex audio and 
   Speex audio only.

5. SDP usage of Speex

   When conveying information by SDP [4], the encoding name MUST be
   set to "speex".  An example of the media representation in SDP for
   offering a single channel of Speex at 8000 samples per second might

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

   Note that the RTP payload type code of 97 is defined in this media
   definition to be 'mapped' to the speex codec at an 8kHz sampling
   frequency using the 'a=rtpmap' line.  Any number from 96 to 127
   could have been chosen (the allowed range for dynamic types).  

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   The value of the sampling frequency is typically 8000 for narrow band
   operation, 16000 for wide band operation, and 32000 for ultra-wide
   band operation.

   If for some reason the offerer has bandwidth limitations, the client 
   may use the "b=" header, as explained in SDP [4]. 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 speex/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 via the a=fmtp: directive.  The
   following parameters are defined for use in this way:

         ptime: duration of each packet in milliseconds.

	 sr:    actual sample rate in Hz.

	 ebw:   encoding bandwidth - either 'narrow' or 'wide' or 
                'ultra' (corresponds to nominal 8000, 16000, and
		32000 Hz sampling rates).

	 vbr:   variable bit rate  - either 'on' 'off' or 'vad'
		(defaults to off).  If on, variable bit rate is
		enabled.  If off, disabled.  If set to 'vad' then
		constant bit rate is used but silence will be encoded
		with special short frames to indicate a lack of voice
		for that period.

	 cng:   comfort noise generation - either 'on' or 'off'. If
		off then silence frames will be silent; if 'on' then
		those frames will be filled with comfort noise.

	 mode:  Speex encoding mode. Can be {1,2,3,4,5,6,any}
                defaults to 3 in narrowband, 6 in wide and ultra-wide.

	 penh:	use of perceptual enhancement. 1 indicates 
	 	to the decoder that perceptual enhancement is recommended,
		0 indicates that it is not. Defaults to on (1).

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   	m=audio 8008 RTP/AVP 97
	a=rtpmap:97 speex/8000
	a=fmtp:97 mode=4

   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 speex/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

   The offerer may indicate that it wishes to send variable bit rate
   frames with comfort noise:

	m=audio 8088 RTP/AVP 97
	a=rtmap:97 speex/8000
	a=fmtp:97 vbr=on;cng=on

   The "ptime" attribute is used to denote the packetization 
   interval (ie, how many milliseconds of audio is encoded in a 
   single RTP packet).  Since Speex uses 20 msec frames, ptime values 
   of multiples of 20 denote multiple Speex frames per packet.  
   Values of ptime which are not multiples of 20 MUST be ignored 
   and clients MUST use the default value of 20 instead.
   In the example below the ptime value is set to 40, indicating that 
   there are 2 frames in each packet.	
   	m=audio 8008 RTP/AVP 97
	a=rtpmap:97 speex/8000
   Note that the ptime parameter applies to all payloads listed
   in the media line and is not used as part of an a=fmtp directive.

   Values of ptime not multiple of 20 msec are meaningless, so the 
   receiver of such ptime values MUST ignore them.  If during the 
   life of an RTP session the ptime value changes, when there are 
   multiple Speex frames for example, the SDP value must also reflect 
   the new value. 

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   Care must be taken when setting the value of ptime so that the 
   RTP packet size does not exceed the path MTU. 

6. ITU H.323/H.245 Use of Speex

   Application is underway to make Speex a standard ITU codec.
   However, until that is finalized, Speex MAY be used in H.323 [6] by
   using a non-standard codec block definition in the H.245 [7] codec
   capability negotiations.  

6.1 NonStandardMessage format

   For Speex use in H.245 [7] based systems, the fields in the
   NonStandardMessage should be:

   t35CountryCode   = Hex: B5
   t35Extension     = Hex: 00
   manufacturerCode = Hex: 0026
   [Length of the Binary Sequence (8 bit number)]
   [Binary Sequence consisting of an ASCII string, no NULL terminator]

   The binary sequence is an ascii string merely for ease of use.
   The string is not null terminated.  The format of this string is

       speex [optional variables]
   The optional variables are identical to those used for the SDP
   a=fmtp strings discussed in section 5 above.  The string is built
   to be all on one line, each key-value pair separated by a
   semi-colon.  The optional variables MAY be omitted, which causes
   the default values to be assumed.  They are:


   The fifth byte of the block is the length of the binary sequence.

   NOTE:  this method can result in the advertising of a large number
   of Speex 'codecs' based on the number of variables possible.  For
   most VoIP applications, use of the default binary sequence of
   'speex' is RECOMMENDED to be used in addition to all other options.
   This maximizes the chances that two H.323 based applications that
   support Speex can find a mutual codec.   

6.2 RTP Payload Types

   Dynamic payload type codes MUST be negotiated 'out-of-band'
   for the assignment of a dynamic payload type from the
   range of 96-127.  H.323 applications MUST use the H.245
   H2250LogicalChannelParameters encoding to accomplish this.  

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7. Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the security considerations discussed in the RTP
   specification [2], 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.  

8. Normative References

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

   2.  Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP:
       A Transport Protocol for real-time applications", RFC 1889,
       January 1996.

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

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

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

   6.  ITU-T Recommendation H.323.  "Packet-based Multimedia 
       Communications Systems," 1998.

   7.  ITU-T Recommendation H.245 (1998), "Control of communications
       between Visual Telephone Systems and Terminal Equipment".

   8.  RTP: A transport protocol for real-time applications. Work   
       in progress, draft-ietf-avt-rtp-new-12.txt.

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   9.  RTP Profile for Audio and Video Conferences with Minimal  
       Control.  Work in progress, draft-ietf-avt-profile-new-13.txt.

   10. L. Walleij, "The application/ogg Media Type", RFC 3534, May 

8.1 Informative References

   11. Speexenc/speexdec, reference command-line encoder/decoder, 
       Speex website,
   12. CELP, U.S. Federal Standard 1016.  National Technical 
       Information Service (NTIS) website, 

9. Acknowledgments

   The authors would like to thank Equivalence Pty Ltd of Australia
   for their assistance in attempting to standardize the use of Speex
   in H.323 applications, and for implementing Speex in their open
   source OpenH323 stack.  The authors would also like to thank Brian
   C. Wiles <> of StreamComm for his assistance in
   developing the proposed standard for Speex use in H.323

   The authors would also like to thank the following members of the 
   Speex and AVT communities for their input:  Ross Finlayson, 
   Federico Montesino Pouzols, Henning Schulzrinne, Magnus Westerlund.

10. Author's Address

   Greg Herlein <>
   2034 Filbert Street
   San Francisco, CA 
   United States 94123

   Jean-Marc Valin <>
   Department of Electrical and Computer Engineering
   University of Sherbrooke
   2500 blvd UniversitŁ√Ł≠Ł√ť
   Sherbrooke, Quebec, Canada, J1K 2R1

   Simon MORLAT <>
   35, av de Vizille App 42
   38000 GRENOBLE

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   Roger Hardiman <>
   49 Nettleton Road
   GL51 6NR

   Phil Kerr <>
   Centre for Music Technology
   University of Glasgow
   G12 8LT

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