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Versions: (draft-sollaud-avt-rtp-g711wb) 00 01 02 03 RFC 5391

Network Working Group                                         A. Sollaud
Internet-Draft                                            France Telecom
Intended status: Standards Track                          April 28, 2008
Expires: October 30, 2008


          RTP Payload Format for ITU-T Recommendation G.711.1
                      draft-ietf-avt-rtp-g711wb-03

Status of this Memo

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   This Internet-Draft will expire on October 30, 2008.

Abstract

   This document specifies a Real-time Transport Protocol (RTP) payload
   format to be used for the International Telecommunication Union
   (ITU-T) G.711.1 audio codec.  Two media type registrations are also
   included.











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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Background . . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  RTP Header Usage . . . . . . . . . . . . . . . . . . . . . . .  4
   4.  Payload Format . . . . . . . . . . . . . . . . . . . . . . . .  5
     4.1.  Payload Header . . . . . . . . . . . . . . . . . . . . . .  5
     4.2.  Audio Data . . . . . . . . . . . . . . . . . . . . . . . .  6
   5.  Payload Format Parameters  . . . . . . . . . . . . . . . . . .  7
     5.1.  PCMA-WB Media Type Registration  . . . . . . . . . . . . .  7
     5.2.  PCMU-WB Media Type Registration  . . . . . . . . . . . . .  8
     5.3.  Mapping to SDP Parameters  . . . . . . . . . . . . . . . .  9
       5.3.1.  Offer-Answer Model Considerations  . . . . . . . . . . 10
       5.3.2.  Declarative SDP Considerations . . . . . . . . . . . . 12
   6.  G.711 Interoperabilty  . . . . . . . . . . . . . . . . . . . . 12
   7.  Congestion Control . . . . . . . . . . . . . . . . . . . . . . 12
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 13
   9.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 13
   10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 13
     10.1. Normative References . . . . . . . . . . . . . . . . . . . 13
     10.2. Informative References . . . . . . . . . . . . . . . . . . 14
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 14
   Intellectual Property and Copyright Statements . . . . . . . . . . 15




























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1.  Introduction

   The International Telecommunication Union (ITU-T) Recommendation
   G.711.1 [ITU-G.711.1] is an embedded wideband extension of the
   Recommendation G.711 [ITU-G.711] audio codec.  This document
   specifies a payload format for packetization of G.711.1 encoded audio
   signals into the Real-time Transport Protocol (RTP).

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT","RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].


2.  Background

   G.711.1 is a G.711 embedded wideband speech and audio coding
   algorithm operating at 64, 80, and 96 kbps.  At 64 kbps, G.711.1 is
   fully interoperable with G.711.  Hence, an efficient deployment in
   existing G.711 based VoIP infrastructures is foreseen.

   The codec operates on 5 ms frames, and the default sampling rate is
   16 kHz.  Input and output at 8 kHz are also supported for narrowband
   modes.

   The encoder produces an embedded bitstream structured in three layers
   corresponding to three available bit rates: 64, 80, and 96 kbps.  The
   bitstream can be truncated at the decoder side or by any component of
   the communication system to adjust "on the fly" the bit rate to the
   desired value.

   The following table gives more details on these layers.

               +-------+------------------------+---------+
               | Layer | Description            | Bitrate |
               +-------+------------------------+---------+
               | L0    | G.711 compatible       | 64 kbps |
               | L1    | narrowband enhancement | 16 kbps |
               | L2    | wideband enhancement   | 16 kbps |
               +-------+------------------------+---------+

                        Table 1: Layers description

   The combinations of these 3 layers results in the definition of 4
   modes, as per the following table.







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              +------+----+----+----+------------+---------+
              | Mode | L0 | L1 | L2 | Audio band | Bitrate |
              +------+----+----+----+------------+---------+
              | R1   | x  |    |    | narrowband | 64 kbps |
              | R2a  | x  | x  |    | narrowband | 80 kbps |
              | R2b  | x  |    | x  | wideband   | 80 kbps |
              | R3   | x  | x  | x  | wideband   | 96 kbps |
              +------+----+----+----+------------+---------+

                        Table 2: Modes description


3.  RTP Header Usage

   The format of the RTP header is specified in [RFC3550].  The payload
   format defined in this document uses the fields of the header in a
   manner consistent with that specification.

   marker (M):
      G.711.1 does not define anything specific regarding Discontinuous
      Transmission (DTX), a.k.a. silence suppression.  Codec-independant
      mechanisms may be used, like the generic comfort noise payload
      format defined in [RFC3389].

      For applications which send either no packets or occasional
      comfort-noise packets during silence, the first packet of a
      talkspurt, that is, the first packet after a silence period during
      which packets have not been transmitted contiguously, SHOULD be
      distinguished by setting the marker bit in the RTP data header to
      one.  The marker bit in all other packets is zero.  The beginning
      of a talkspurt MAY be used to adjust the playout delay to reflect
      changing network delays.  Applications without silence suppression
      MUST set the marker bit to zero.

   payload type (PT):
      The assignment of an RTP payload type for this packet format is
      outside the scope of the document, and will not be specified here.
      It is expected that the RTP profile under which this payload
      format is being used will assign a payload type for this codec or
      specify that the payload type is to be bound dynamically (see
      Section 5.3).

   timestamp:
      The RTP timestamp clock frequency is the same as the default
      sampling frequency: 16 kHz.






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      G.711.1 has also the capability to operate with 8 kHz sampled
      input/output signals.  It does not affect the bitstream, and the
      decoder does not require a priori knowledge about the sampling
      rate of the original signal at the input of the encoder.
      Therefore, depending on the implementation and the audio acoustic
      capabilities of the devices, the input of the encoder and/or the
      output of the decoder can be configured at 8 kHz; however, a 16
      kHz RTP clock rate MUST always be used.

      The duration of one frame is 5 ms, corresponding to 80 samples at
      16 kHz.  Thus the timestamp is increased by 80 for each
      consecutive frame.


4.  Payload Format

   The complete payload consists of a payload header of 1 octet,
   followed by one or more consecutive G.711.1 audio frames of the same
   mode.

   The mode may change between packets, but not within a packet.

4.1.  Payload Header

   The payload header is illustrated below.

      0 1 2 3 4 5 6 7
     +-+-+-+-+-+-+-+-+
     |0 0 0 0 0|  MI |
     +-+-+-+-+-+-+-+-+

   The 5 most significant bits are reserved for further extension and
   MUST be set to zero.

   The MI field (3 bits) gives the mode of the following frame(s) as per
   the table:

                +------------+--------------+------------+
                | Mode Index | G.711.1 mode | Frame size |
                +------------+--------------+------------+
                |      1     |      R1      |  40 octets |
                |      2     |      R2a     |  50 octets |
                |      3     |      R2b     |  50 octets |
                |      4     |      R3      |  60 octets |
                +------------+--------------+------------+

                     Table 3: Modes in payload header




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   All other values of MI are reserved for future used and MUST NOT be
   used.

   Payloads received with an undefined MI value MUST be discarded.

   If a restricted mode set has been set up by the signaling (see
   Section 5), payloads received with a MI value not in this set MUST be
   discarded.

4.2.  Audio Data

   After this payload header, the audio frames are packed in order of
   time, that is, oldest first.  All frames MUST be of the same mode,
   indicated by the MI field of the payload header.

   Within a frame, layers are always packed in the same order: L0 then
   L1 for mode R2a, L0 then L2 for mode R2b, L0 then L1 then L2 for mode
   R3.  This is illustrated bellow.

         +-------------------------------+
     R1  |              L0               |
         +-------------------------------+

         +-------------------------------+--------+
     R2a |              L0               |   L1   |
         +-------------------------------+--------+

         +-------------------------------+--------+
     R2b |              L0               |   L2   |
         +-------------------------------+--------+

         +-------------------------------+--------+--------+
     R3  |              L0               |   L1   |   L2   |
         +-------------------------------+--------+--------+

   The size of one frame is given by the mode, as per Table 3, and the
   actual number of frames is easy to infer from the size of the audio
   data part:

      nb_frames = (size_of_audio_data) / (size_of_one_frame).

   Only full frames must be considered.  So if there is a remainder to
   the division above, the corresponding remaining bytes in the received
   payload MUST be ignored.







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5.  Payload Format Parameters

   This section defines the parameters that may be used to configure
   optional features in the G.711.1 RTP transmission.

   Both A-law and mu-law G.711 are supported in the core layer L0, but
   there is no interoperability between A-law and mu-law.  So two media
   types with the same parameters will be defined: audio/PCMA-WB for
   A-law core, and audio/PCMU-WB for mu-law core.  This is consistent
   with the audio/PCMA and audio/PCMU media types separation for G.711
   audio.

   The parameters are defined here as part of the media subtype
   registrations for the G.711.1 codec.  A mapping of the parameters
   into the Session Description Protocol (SDP) [RFC4566] is also
   provided for those applications that use SDP.  In control protocols
   that do not use MIME or SDP, the media type parameters must be mapped
   to the appropriate format used with that control protocol.

5.1.  PCMA-WB Media Type Registration

   [Note to RFC Editor: Please replace all occurrences of RFC XXXX by
   the RFC number assigned to this document]

   This registration is done using the template defined in [RFC4288] and
   following [RFC4855].

   Type name: audio

   Subtype name: PCMA-WB

   Required parameters: none

   Optional parameters:

   mode-set:  restricts the active codec mode set to a subset of all
      modes.  Possible values are a comma-separated list of modes from
      the set: 1,2,3,4 (see Mode Index in Table 3 of RFC XXXX).  The
      modes are listed in order of preference, first is prefered.  If
      mode-set is specified, frames encoded with modes outside of the
      subset MUST NOT be sent in any RTP payload.  If not present, all
      codec modes are allowed.

   ptime:  the recommended length of time (in milliseconds) represented
      by the media in a packet.  It should be an integer multiple of 5
      ms (the frame size).  See Section 6 of RFC 4566.





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   maxptime:  the maximum length of time (in milliseconds) that can be
      encapsulated in a packet.  It should be an integer multiple of 5
      ms (the frame size).  See Section 6 of RFC 4566.

   Encoding considerations: This media type is framed and contains
   binary data; see Section 4.8 of RFC 4288.

   Security considerations: See Section 8 of RFC XXXX

   Interoperability considerations: none

   Published specification: RFC XXXX

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

   Additional information: none

   Person & email address to contact for further information: Aurelien
   Sollaud, aurelien.sollaud@orange-ftgroup.com

   Intended usage: COMMON

   Restrictions on usage: This media type depends on RTP framing, and
   hence is only defined for transfer via RTP.

   Author: Aurelien Sollaud

   Change controller: IETF Audio/Video Transport working group delegated
   from the IESG

5.2.  PCMU-WB Media Type Registration

   This registration is done using the template defined in [RFC4288] and
   following [RFC4855].

   Type name: audio

   Subtype name: PCMU-WB

   Required parameters: none

   Optional parameters:







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   mode-set:  restricts the active codec mode set to a subset of all
      modes.  Possible values are a comma-separated list of modes from
      the set: 1,2,3,4 (see Mode Index in Table 3 of RFC XXXX).  The
      modes are listed in order of preference, first is prefered.  If
      mode-set is specified, frames encoded with modes outside of the
      subset MUST NOT be sent in any RTP payload.  If not present, all
      codec modes are allowed.

   ptime:  the recommended length of time (in milliseconds) represented
      by the media in a packet.  It should be an integer multiple of 5
      ms (the frame size).  See Section 6 of RFC 4566.

   maxptime:  the maximum length of time (in milliseconds) that can be
      encapsulated in a packet.  It should be an integer multiple of 5
      ms (the frame size).  See Section 6 of RFC 4566.

   Encoding considerations: This media type is framed and contains
   binary data; see Section 4.8 of RFC 4288.

   Security considerations: See Section 8 of RFC XXXX

   Interoperability considerations: none

   Published specification: RFC XXXX

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

   Additional information: none

   Person & email address to contact for further information: Aurelien
   Sollaud, aurelien.sollaud@orange-ftgroup.com

   Intended usage: COMMON

   Restrictions on usage: This media type depends on RTP framing, and
   hence is only defined for transfer via RTP.

   Author: Aurelien Sollaud

   Change controller: IETF Audio/Video Transport working group delegated
   from the IESG

5.3.  Mapping to SDP Parameters

   The information carried in the media type specification has a
   specific mapping to fields in the Session Description Protocol (SDP)
   [RFC4566], which is commonly used to describe RTP sessions.  When SDP



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   is used to specify sessions employing the G.711.1 codec, the mapping
   is as follows:

   o  The media type ("audio") goes in SDP "m=" as the media name.

   o  The media subtype ("PCMA-WB" or "PCMU-WB") goes in SDP "a=rtpmap"
      as the encoding name.  The RTP clock rate in "a=rtpmap" MUST be
      16000 for G.711.1.

   o  The parameter "mode-set" goes in the SDP "a=fmtp" attribute by
      copying it as a "mode-set=<value>" string.

   o  The parameters "ptime" and "maxptime" go in the SDP "a=ptime" and
      "a=maxptime" attributes, respectively.

5.3.1.  Offer-Answer Model Considerations

   The following considerations apply when using SDP offer-answer
   procedures [RFC3264] to negotiate the use of G.711.1 payload in RTP:

   o  Since G.711.1 is an extension of G.711, the offerer SHOULD
      announce G.711 support in its "m=audio" line, with G.711.1
      preferred.  This will allow interoperability with both G.711.1 and
      G.711-only capable parties.  This is done by offering the PCMA
      media subtype in addition to PCMA-WB, and/or PCMU in addition to
      PCMU-WB.

      Below is an example part of such an offer, for A-law:

         m=audio 54874 RTP/AVP 96 8
         a=rtpmap:96 PCMA-WB/16000
         a=rtpmap:8 PCMA/8000

      As a reminder, the payload format for G.711 is defined in Section
      4.5.14 of [RFC3551].

   o  The "mode-set" parameter is bi-directional, i.e., the restricted
      mode set applies to media both to be received and sent by the
      declaring entity.  If a mode-set was supplied in the offer, the
      answerer MUST return either the same mode-set or a subset of this
      mode-set.  The answerer MAY change the preference order.  If no
      mode-set was supplied in the offer, the anwerer MAY return a mode-
      set to restrict the possible modes.  In any cases, the mode-set in
      the answer then applies for both offerer and answerer.  The
      offerer MUST NOT send frames of a mode that has been removed by
      the answerer.

      For multicast sessions, if "mode-set" is supplied in the offer,



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      the answerer SHALL only participate in the session if it supports
      the offered mode set.

   o  The parameters "ptime" and "maxptime" will in most cases not
      affect interoperability.  The SDP offer-answer handling of the
      "ptime" parameter is described in [RFC3264].  The "maxptime"
      parameter MUST be handled in the same way.

   o  Any unknown parameter in an offer MUST be ignored by the receiver
      and MUST NOT be included in the answer.


   Below are some example parts of SDP offer-answer exchanges.

   o  Example 1

      Offer: G.711.1 all modes, with G.711 fallback, prefers mu-law
         m=audio 54874 RTP/AVP 96 97 0 8
         a=rtpmap:96 PCMU-WB/16000
         a=rtpmap:97 PCMA-WB/16000
         a=rtpmap:0 PCMU/8000
         a=rtpmap:8 PCMA/8000

      Answer: all modes accepted, both mu- and A-law.
         m=audio 59452 RTP/AVP 96 97
         a=rtpmap:96 PCMU-WB/16000
         a=rtpmap:97 PCMA-WB/16000

   o  Example 2

      Offer: G.711.1 all modes, with G.711 fallback, prefers A-law
         m=audio 54874 RTP/AVP 96 97 8 0
         a=rtpmap:96 PCMA-WB/16000
         a=rtpmap:97 PCMU-WB/16000

      Answer: wants only A-law mode R3
         m=audio 59452 RTP/AVP 96
         a=rtpmap:96 PCMA-WB/16000
         a=fmtp:96 mode-set=4

   o  Example 3

      Offer: G.711.1 A-law with two modes, R2b and R3, with R3 preferred
         m=audio 54874 RTP/AVP 96
         a=rtpmap:96 PCMA-WB/16000
         a=fmtp:96 mode-set=4,3





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      Answer: accepted
         m=audio 59452 RTP/AVP 96
         a=rtpmap:96 PCMA-WB/16000
         a=fmtp:96 mode-set=4,3

      If the answerer wanted to restrict to one mode, it would have
      answered with only one value in the mode-set.  For example mode-
      set=3 for mode R2b.

5.3.2.  Declarative SDP Considerations

   For declarative use of SDP, nothing specific is defined for this
   payload format.  The configuration given by the SDP MUST be used when
   sending and/or receiving media in the session.


6.  G.711 Interoperabilty

   The L0 layer of G.711.1 is fully interoperable with G.711, and is
   embedded in all modes of G.711.1.  This provides an easy G.711.1 -
   G.711 transcoding process.

   A gateway or any other network device receiving a G.711.1 packet can
   easily extract a G.711-compatible payload, without the need to decode
   and re-encode the audio signal.  It simply has to take the audio data
   of the payload, and strip the upper layers (L1 and/or L2), if any.

   If a G.711.1 packet contains several frames, the concatenation of the
   L0 layers of each frame will form a G.711-compatible payload.


7.  Congestion Control

   Congestion control for RTP SHALL be used in accordance with [RFC3550]
   and any appropriate profile (for example, [RFC3551]).

   The embedded nature of G.711.1 audio data can be helpful for
   congestion control, since a coding mode with a lower bit rate can be
   selected when needed.  This property is usable only when multiple
   modes have been negotiated (either no "mode-set" parameter in the
   SDP, or a "mode-set" with at least 2 modes).

   The number of frames encapsulated in each RTP payload influences the
   overall bandwidth of the RTP stream, due to the header overhead.
   Packing more frames in each RTP payload can reduce the number of
   packets sent and hence the header overhead, at the expense of
   increased delay and reduced error robustness.



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8.  Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the general security considerations discussed in the
   RTP specification [RFC3550] and any appropriate profile (for example,
   [RFC3551]).

   As this format transports encoded speech/audio, the main security
   issues include confidentiality, integrity protection, and
   authentication of the speech/audio itself.  The payload format itself
   does not have any built-in security mechanisms.  Any suitable
   external mechanisms, such as SRTP [RFC3711], MAY be used.

   This payload format and the G.711.1 encoding do not exhibit any
   significant non-uniformity in the receiver-end computational load and
   thus are unlikely to pose a denial-of-service threat due to the
   receipt of pathological datagrams.


9.  IANA Considerations

   It is requested that two new media subtype (audio/PCMA-WB and audio/
   PCMU-WB) are registered by IANA, see Section 5.1 and Section 5.2.


10.  References

10.1.  Normative References

   [ITU-G.711.1]
              International Telecommunications Union, "Wideband embedded
              extension for G.711 pulse code modulation", ITU-
              T Recommendation G.711.1, March 2008.

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

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [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.




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   [RFC4288]  Freed, N. and J. Klensin, "Media Type Specifications and
              Registration Procedures", BCP 13, RFC 4288, December 2005.

   [RFC4855]  Casner, S., "Media Type Registration of RTP Payload
              Formats", RFC 4855, February 2007.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              June 2002.

10.2.  Informative References

   [ITU-G.711]
              International Telecommunications Union, "Pulse code
              modulation (PCM) of voice frequencies", ITU-
              T Recommendation G.711, November 1988.

   [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
              Comfort Noise (CN)", RFC 3389, September 2002.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.


Author's Address

   Aurelien Sollaud
   France Telecom
   2 avenue Pierre Marzin
   Lannion Cedex  22307
   France

   Phone: +33 2 96 05 15 06
   Email: aurelien.sollaud@orange-ftgroup.com
















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   Copies of IPR disclosures made to the IETF Secretariat and any
   assurances of licenses to be made available, or the result of an
   attempt made to obtain a general license or permission for the use of
   such proprietary rights by implementers or users of this
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   http://www.ietf.org/ipr.

   The IETF invites any interested party to bring to its attention any
   copyrights, patents or patent applications, or other proprietary
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   this standard.  Please address the information to the IETF at
   ietf-ipr@ietf.org.











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