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Versions: (draft-westerlund-avt-rtp-gsm-hr) 00 01 02 03 RFC 5993

Network Working Group                                            X. Duan
Internet-Draft                                                   S. Wang
Intended status: Standards Track             China Mobile Communications
Expires: July 25, 2010                                       Corporation
                                                           M. Westerlund
                                                              K. Hellwig
                                                            I. Johansson
                                                             Ericsson AB
                                                        January 21, 2010


                     RTP Payload format for GSM-HR
                      draft-ietf-avt-rtp-gsm-hr-03

Abstract

   This document specifies the payload format for packetization of the
   GSM Half-Rate speech codec data into the Real-time Transport Protocol
   (RTP).  The payload format supports transmission of multiple frames
   per payload and packet loss robustness methods using redundancy.

Status of this Memo

   This Internet-Draft is submitted to IETF in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
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   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
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   http://www.ietf.org/shadow.html.

   This Internet-Draft will expire on July 25, 2010.

Copyright Notice

   Copyright (c) 2010 IETF Trust and the persons identified as the
   document authors.  All rights reserved.



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   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
   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 BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Conventions  . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  GSM Half Rate  . . . . . . . . . . . . . . . . . . . . . . . .  3
   4.  Payload format Capabilities  . . . . . . . . . . . . . . . . .  4
     4.1.  Use of Forward Error Correction (FEC)  . . . . . . . . . .  4
   5.  Payload format . . . . . . . . . . . . . . . . . . . . . . . .  5
     5.1.  RTP Header Usage . . . . . . . . . . . . . . . . . . . . .  6
     5.2.  Payload Structure  . . . . . . . . . . . . . . . . . . . .  6
       5.2.1.  Encoding of Speech Frames  . . . . . . . . . . . . . .  8
       5.2.2.  Encoding of Silence Description Frames . . . . . . . .  8
     5.3.  Implementation Considerations  . . . . . . . . . . . . . .  8
       5.3.1.  Transmission of SID frames . . . . . . . . . . . . . .  8
       5.3.2.  Receiving Redundant Frames . . . . . . . . . . . . . .  8
       5.3.3.  Decoding Validation  . . . . . . . . . . . . . . . . .  8
   6.  Examples . . . . . . . . . . . . . . . . . . . . . . . . . . .  9
     6.1.  3 frames . . . . . . . . . . . . . . . . . . . . . . . . .  9
     6.2.  3 Frames with lost frame in the middle . . . . . . . . . . 10
   7.  Payload Format Parameters  . . . . . . . . . . . . . . . . . . 10
     7.1.  Media Type Definition  . . . . . . . . . . . . . . . . . . 11
     7.2.  Mapping to SDP . . . . . . . . . . . . . . . . . . . . . . 12
       7.2.1.  Offer/Answer Considerations  . . . . . . . . . . . . . 13
       7.2.2.  Declarative SDP Considerations . . . . . . . . . . . . 13
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 14
   9.  Congestion Control . . . . . . . . . . . . . . . . . . . . . . 14
   10. Security Considerations  . . . . . . . . . . . . . . . . . . . 14
   11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 15
   12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
     12.1. Normative References . . . . . . . . . . . . . . . . . . . 15
     12.2. Informative References . . . . . . . . . . . . . . . . . . 16
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 16








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1.  Introduction

   This document specifies the payload format for packetization of GSM
   Half Rate (GSM-HR) codec [TS46.002] encoded speech signals into the
   Real-time Transport Protocol (RTP) [RFC3550].  The payload format
   supports transmission of multiple frames per payload and packet loss
   robustness methods using redundancy.

   This document starts with conventions, a brief description of the
   codec, and the payload formats capabilities.  The payload format is
   specified in Section 5.  Examples can be found in Section 6.  The
   media type and its mappings to SDP, usage in SDP offer/answer is then
   specified.  The document ends with considerations around congestion
   control and security.

   This document registers a media type (audio/gsm-hr-08) for the Real-
   time Transport protocol (RTP) payload format for the GSM-HR codec.
   Note: This format is not compatible with the one that was drafted
   back in 1999 to 2000 in the Internet drafts:
   draft-ietf-avt-profile-new-05 to draft-ietf-avt-profile-new-09.  A
   later version of the AVP profile draft was published as RFC 3551
   without any specification of the GSM-HR payload format.  To avoid a
   possible conflict with this older format, the media type of the
   payload format specified in this document has a media type name that
   is different from (audio/gsm-hr).


2.  Conventions

   This document uses the normal IETF bit-order representation.  Bit
   fields in figures are read left to right and then down.  The left
   most bit in each field is the most significant.  The numbering starts
   from 0 and ascends, where bit 0 will be the most significant.

   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 RFC 2119 [RFC2119].


3.  GSM Half Rate

   The Global System for Mobile Communication (GSM) network provides
   with mobile communication services for nearly 3 billion users (status
   2008).  The GSM Half Rate Codec (GSM-HR) is one of the speech codecs
   that are used in GSM networks.  GSM-HR denotes the Half-Rate speech
   codec as specified in [TS46.002].

   Note: for historical reasons these 46-series specifications are



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   internally referenced as 06-series.  A simple mapping applies, for
   example 46.020 is referenced as 06.20 and so on.

   The GSM-HR codec has a frame length of 20 ms, with narrowband speech
   sampled at 8 kHz, i.e. 160 samples per frame.  Each speech frame is
   compressed into 112 bits of speech parameters, which is equivalent to
   a bit rate of 5.6 kbit/s.  Speech pauses are detected by a
   standardized Voice Activity Detection (VAD).  During speech pauses
   the transmission of speech frames is inhibited.  Silence Descriptor
   (SID) frames are transmitted at the end of a talk spurt and about
   every 480ms during speech pauses to allow for a decent Comfort Noise
   (CN) quality at receiver side.

   The SID frame generation in the GSM radio network is determined by
   the GSM mobile station and the GSM radio subsystem.  SID frames come
   during speech pauses in uplink from the mobile station about every
   480ms.  In downlink to the mobile station, when they are generated by
   the encoder of the GSM radio subsystem, SID frames are sent every
   20ms to the GSM base station, which then picks only one every 480ms
   for downlink radio transmission.  For other applications, like
   transport over IP, it is more appropriate to send the SID frames less
   often than every 20ms, but 480 ms may be too sparse.  We recommend as
   a compromise that a GSM-HR encoder outside of the GSM radio network
   (i.e. not in the GSM mobile station and not in the GSM radio
   subsystem, but for example in the media gateway of the core network)
   should generate and send SID frames every 160ms.


4.  Payload format Capabilities

   This RTP payload format carries one or more GSM-HR encoded frames,
   either full voice or silence descriptor (SID), representing a mono
   speech signal.  To maintain synchronization or express not sent or
   lost frames it has the capability to indicate No_Data frames.

4.1.  Use of Forward Error Correction (FEC)

   Generic forward error correction within RTP is defined, for example,
   in RFC 5109 [RFC5109].  Audio redundancy coding is defined in RFC
   2198 [RFC2198].  Either scheme can be used to add redundant
   information to the RTP packet stream and make it more resilient to
   packet losses, at the expense of a higher bit rate.  Please see
   either RFCs for a discussion of the implications of the higher bit
   rate to network congestion.

   In addition to these media-unaware mechanisms, this memo specifies an
   optional to use GSM-HR specific form of audio redundancy coding,
   which may be beneficial in terms of packetization overhead.



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   Conceptually, previously transmitted transport frames are aggregated
   together with new ones.  A sliding window can be used to group the
   frames to be sent in each payload.  Figure 1 below shows an example.
   --+--------+--------+--------+--------+--------+--------+--------+--
     | f(n-2) | f(n-1) |  f(n)  | f(n+1) | f(n+2) | f(n+3) | f(n+4) |
   --+--------+--------+--------+--------+--------+--------+--------+--

      <---- p(n-1) ---->
               <----- p(n) ----->
                        <---- p(n+1) ---->
                                 <---- p(n+2) ---->
                                          <---- p(n+3) ---->
                                                   <---- p(n+4) ---->

              Figure 1: An example of redundant transmission

   Here, each frame is retransmitted once in the following RTP payload
   packet. f(n-2)...f(n+4) denote a sequence of audio frames, and p(n-
   1)...p(n+4) a sequence of payload packets.

   The mechanism described does not really require signaling at the
   session setup.  However, signalling has been defined to allow for the
   sender to voluntarily bounding the buffering and delay requirements.
   If nothing is signalled the use of this mechanism is allowed and
   unbounded.  For a certain timestamp, the receiver may receive
   multiple copies of a frame containing encoded audio data.  The cost
   of this scheme is bandwidth and the receiver delay necessary to allow
   the redundant copy to arrive.

   This redundancy scheme provides a functionality similar to the one
   described in RFC 2198, but it works only if both original frames and
   redundant representations are GSM-HR frames.  When the use of other
   media coding schemes is desirable, one has to resort to RFC 2198.

   The sender is responsible for selecting an appropriate amount of
   redundancy based on feedback about the channel conditions, e.g., in
   the RTP Control Protocol (RTCP) [RFC3550] receiver reports.  The
   sender is also responsible for avoiding congestion, which may be
   exacerbated by redundancy (see Section 9 for more details).


5.  Payload format

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

   The duration of one speech frame is 20 ms.  The sampling frequency is



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   8kHz, corresponding to 160 speech samples per frame.  An RTP packet
   may contain multiple frames of encoded speech or SID parameters.
   Each packet covers a period of one or more contiguous 20 ms frame
   intervals.  During silence periods no speech packets are sent,
   however SID packets are transmitted every now and then.

   To allow for error resiliency through redundant transmission, the
   periods covered by multiple packets MAY overlap in time.  A receiver
   MUST be prepared to receive any speech frame multiple times.  A given
   frame MUST NOT be encoded as speech frame in one packet and as SID
   frame or as No_Data frame in another packet.  Furthermore, a given
   frame MUST NOT be encoded with different voicing modes in different
   packets.

   The rules regarding maximum payload size given in Section 3.2 of
   [RFC5405] SHOULD be followed.

5.1.  RTP Header Usage

   The RTP timestamp corresponds to the sampling instant of the first
   sample encoded for the first frame in the packet.  The timestamp
   clock frequency SHALL be 8000 Hz.  The timestamp is also used to
   recover the correct decoding order of the frames.

   The RTP header marker bit (M) SHALL be set to 1 whenever the first
   frame carried in the packet is the first frame in a talkspurt (see
   definition of the talkspurt in section 4.1 of [RFC3551]).  For all
   other packets the marker bit SHALL be set to zero (M=0).

   The assignment of an RTP payload type for the format defined in this
   memo is outside the scope of this document.  The RTP profiles in use
   currently mandates binding the payload type dynamically for this
   payload format.

   The remaining RTP header fields are used as specified in RFC 3550
   [RFC3550].

5.2.  Payload Structure

   The complete payload consists of a payload table of contents (ToC)
   section, followed by speech data representing one or more speech
   frames, SID frames or No_Data frames.  The following diagram shows
   the general payload format layout:
      +-------------+-------------------------
      | ToC section | speech data section ...
      +-------------+-----------------------

                  Figure 2: General payload format layout



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   Each ToC element is one octet and corresponds to one speech frame,
   the number of ToC elements is thus equal to the number of speech
   frames (including SID frames and No_Data frames).  Each ToC entry
   represents a consecutive speech or SID or No_Data frame.  The
   timestamp value for ToC element (and corresponding speech frame data)
   N within the payload is (RTP timestamp field + (N-1)*160) mod 2^32 .
   The format of the ToC element is as follows.

       0 1 2 3 4 5 6 7
      +-+-+-+-+-+-+-+-+
      |F| FT  |R R R R|
      +-+-+-+-+-+-+-+-+

                         Figure 3: The TOC element

   F: Follow flag, 1 denotes that more ToC elements follow, 0 denotes
      the last ToC element.

   R: Reserved bits, MUST be set to zero and MUST be ignored by
      receiver.

   FT:  Frame type
      000 = Good Speech frame
      001 = Reserved
      010 = Good SID frame
      011 = Reserved
      100 = Reserved
      101 = Reserved
      110 = Reserved
      111 = No_Data frame

   The length of the payload data depends on the frame type:

   Good Speech frame:   The 112 speech data bits are put in 14 octets.

   Good SID frame:   The 33 SID data bits are put in 14 octets, as in
      case of Speech frames, with the unused 79 bits set all to "1".

   No data frame:   Length of payload data is zero octets.

   Frames marked in the GSM radio subsystem as "Bad Speech frame", "Bad
   SID frame" or "No_Data frame" are not sent in RTP packets in order to
   save bandwidth.  They are marked as "No_Data frame", if they occur
   within an RTP packet that carries more than one speech frame, SID
   frame or No_Data frame.






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5.2.1.  Encoding of Speech Frames

   The 112 bits of GSM-HR-coded speech (b1...b112) are defined in TS
   46.020, Annex B [TS46.020], in the order of occurrence.  The first
   bit (b1) of the first parameter is placed in bit 0 (the MSB) of the
   first octet (octet 1) of the payload field; the second bit is placed
   in bit 1 of the first octet and so on.  The last bit (b112) is placed
   in the LSB (bit 7) of octet 14.

5.2.2.  Encoding of Silence Description Frames

   The GSM-HR Codec applies a specific coding for silence periods in so
   called SID frames.  The coding of SID frames is based on the coding
   of speech frames by using only the first 33 bits for SID parameters
   and by setting the remaining 79 bits all to "1".

5.3.  Implementation Considerations

   An application implementing this payload format MUST understand all
   the payload parameters that is defined in this specification.  Any
   mapping of the parameters to a signaling protocol MUST support all
   parameters.  So an implementation of this payload format in an
   application using SDP is required to understand all the payload
   parameters in their SDP-mapped form.  This requirement ensures that
   an implementation always can decide whether it is capable of
   communicating when the communicating entities support this version of
   the specification.

5.3.1.  Transmission of SID frames

   When using this RTP payload format the sender SHOULD generate and
   send SID frames every 160ms, i.e. every 8th frame, during silent
   periods.  Other SID transmission intervals may occur due to gateways
   to other systems that uses other transmission intervals.

5.3.2.  Receiving Redundant Frames

   The reception of redundant audio frames, i.e. more than one audio
   frame from the same source for the same time slot, MUST be supported
   by the implementation.

5.3.3.  Decoding Validation

   If the receiver finds a mismatch between the size of a received
   payload and the size indicated by the ToC of the payload, the
   receiver SHOULD discard the packet.  This is recommended because
   decoding a frame parsed from a payload based on erroneous ToC data
   could severely degrade the audio quality.



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6.  Examples

   A few examples to highlight the payload format.

6.1.  3 frames

   A basic example of the aggregation of 3 consecutive speech frames
   into a single frame.

      The first 24 bits are ToC elements.
      Bit 0 is '1' as another ToC element follow.
      Bits 1..3 is 000 = Good speech frame
      Bits 4..7 is 0000 = Reserved
      Bits 8 is '1' as another ToC element follow.
      Bits 9..11 is 000 = Good speech frame
      Bits 12..15 is 0000 = Reserved
      Bit 16 is '0', no more ToC element follows
      Bits 17..19 is 000 = Good speech frame
      Bits 20..23 is 0000 = Reserved


       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
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |1|0 0 0|0 0 0 0|1|0 0 0|0 0 0 0|0|0 0 0|0 0 0 0|b1           b8|
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               +
      |b9   Frame 1                                                b40|
      +                                                               +
      |b41                                                         b72|
      +                                                               +
      |b73                                                        b104|
      +               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |b105       b112|b1                                          b24|
      +-+-+-+-+-+-+-+-+                                               +
      |b25  Frame 2                                                b56|
      +                                                               +
      |b57                                                         b88|
      +                                               +-+-+-+-+-+-+-+-+
      |b89                                        b112|b1           b8|
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               +
      |b9   Frame 3                                                b40|
      +                                                               +
      |b41                                                         b72|
      +                                                               +
      |b73                                                        b104|
      +               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |b105       b112|
      +-+-+-+-+-+-+-+-+



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6.2.  3 Frames with lost frame in the middle

   An example of payload carrying 3 frames where the middle one is
   No_Data, for example due to loss prior to transmission by the RTP
   source.

      The first 24 bits are ToC elements.
      Bit 0 is '1' as another ToC element follow.
      Bits 1..3 is 000 = Good speech frame
      Bits 4..7 is 0000 = Reserved
      Bits 8 is '1' as another ToC element follow.
      Bits 9..11 is 111 = No_Data frame
      Bits 12..15 is 0000 = Reserved
      Bit 16 is '0', no more ToC element follows
      Bits 17..19 is 000 = Good speech frame
      Bits 20..23 is 0000 = Reserved


       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
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |1|0 0 0|0 0 0 0|1|1 1 1|0 0 0 0|0|0 0 0|0 0 0 0|b1           b8|
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+               +
      |b9   Frame 1                                                b40|
      +                                                               +
      |b41                                                         b72|
      +                                                               +
      |b73                                                        b104|
      +               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |b105       b112|b1                                          b24|
      +-+-+-+-+-+-+-+-+                                               +
      |b25  Frame 3                                                b56|
      +                                                               +
      |b57                                                         b88|
      +                                               +-+-+-+-+-+-+-+-+
      |b89                                        b112|
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


7.  Payload Format Parameters

   This RTP payload format is identified using the media type "audio/
   gsm-hr-08", which is registered in accordance with [RFC4855] and
   using the template of [RFC4288].  Note: Media subtype names are case-
   insensitive.






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7.1.  Media Type Definition

   The media type for the GSM-HR codec is allocated from the IETF tree
   since GSM-HR is a well know speech codec.  This media type
   registration covers real-time transfer via RTP.  The media subtype
   name contains "-08" to avoid potential conflict with any earlier
   drafts of GSM-HR RTP payload types that aren't bit compatible.

   Note, reception of any unspecified parameter MUST be ignored by the
   receiver to ensure that additional parameters can be added in the
   future.

   Type name: audio

   Subtype name: GSM-HR-08

   Required parameters: none

   Optional parameters:

   max-red:  The maximum duration in milliseconds that elapses between
      the primary (first) transmission of a frame and any redundant
      transmission that the sender will use.  This parameter allows a
      receiver to have a bounded delay when redundancy is used.  Allowed
      values are integers between 0 (no redundancy will be used) and
      65535.  If the parameter is omitted, no limitation on the use of
      redundancy is present.

   ptime:  see [RFC4566].

   maxptime:  see [RFC4566].

   Encoding considerations:

      This media type is framed and binary, see section 4.8 in RFC4288
      [RFC4288].

   Security considerations:

      See Section 10 of RFCXXXX.

   Interoperability considerations:

   Published specification:

      RFC XXXX, 3GPP TS 46.002

   Applications that use this media type:



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      Real-time audio applications like voice over IP and
      teleconference.

   Additional information: none

   Person & email address to contact for further information:

      Ingemar Johansson <ingemar.s.johansson@ericsson.com>

   Intended usage: COMMON

   Restrictions on usage:

      This media type depends on RTP framing, and hence is only defined
      for transfer via RTP [RFC3550].  Transport within other framing
      protocols is not defined at this time.

   Author:

      Xiaodong Duan <duanxiaodong@chinamobile.com>

      Shuaiyu Wang <wangshuaiyu@chinamobile.com>

      Magnus Westerlund <magnus.westerlund@ericsson.com>

      Ingemar Johansson <ingemar.s.johansson@ericsson.com>

      Karl Hellwig <karl.hellwig@ericsson.com>

   Change controller:

      IETF Audio/Video Transport working group delegated from the IESG.

7.2.  Mapping to SDP

   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
   is used to specify sessions employing the GSM-HR codec, the mapping
   is as follows:

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

   o  The media subtype (payload format name) goes in SDP "a=rtpmap" as
      the encoding name.  The RTP clock rate in "a=rtpmap" MUST be 8000,
      and the encoding parameters (number of channels) MUST either be
      explicitly set to 1 or omitted, implying a default value of 1.




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

   o  Any remaining parameters go in the SDP "a=fmtp" attribute by
      copying them directly from the media type parameter string as a
      semicolon-separated list of parameter=value pairs.

7.2.1.  Offer/Answer Considerations

   The following considerations apply when using SDP Offer-Answer
   procedures to negotiate the use of GSM-HR payload in RTP:

   o  The SDP offerer and answerer MUST generate GSM-HR packets as
      described by the offered parameters.

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

   o  The parameter "max-red" is a stream property parameter.  For
      sendonly or sendrecv unicast media streams, the parameter declares
      the limitation on redundancy that the stream sender will use.  For
      recvonly streams, it indicates the desired value for the stream
      sent to the receiver.  The answerer MAY change the value, but is
      RECOMMENDED to use the same limitation as the offer declares.  In
      the case of multicast, the offerer MAY declare a limitation; this
      SHALL be answered using the same value.  A media sender using this
      payload format is RECOMMENDED to always include the "max-red"
      parameter.  This information is likely to simplify the media
      stream handling in the receiver.  This is especially true if no
      redundancy will be used, in which case "max-red" is set to 0.

   o  Any unknown media type parameter in an offer SHALL be removed in
      the answer.

7.2.2.  Declarative SDP Considerations

   In declarative usage, like SDP in RTSP [RFC2326] or SAP [RFC2974],
   the parameters SHALL be interpreted as follows:

   o  The stream property parameter ("max-red") is declarative, and a
      participant MUST follow what is declared for the session.  In this
      case it means that the receiver MUST be prepared to allocate
      buffer memory for the given redundancy.  Any transmissions MUST
      NOT use more redundancy then what has been declared.  More than



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      one configuration may be provided if necessary by declaring
      multiple RTP payload types; however, the number of types should be
      kept small.

   o  Any "maxptime" and "ptime" values should be selected with care to
      ensure that the session's participants can achieve reasonable
      performance.


8.  IANA Considerations

   One media type (audio/gsm-hr-08) has been defined and needs
   registration in the media types registry; see Section 7.1.


9.  Congestion Control

   The general congestion control considerations for transporting RTP
   data apply; see RTP [RFC3550] and any applicable RTP profile like AVP
   [RFC3551].

   The number of frames encapsulated in each RTP payload highly
   influences the overall bandwidth of the RTP stream due to header
   overhead constraints.  Packetizing 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.  If
   forward error correction (FEC) is used, the amount of FEC-induced
   redundancy needs to be regulated such that the use of FEC itself does
   not cause a congestion problem.


10.  Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the security considerations discussed in the RTP
   specification [RFC3550] , and in any applicable RTP profile.  The
   main security considerations for the RTP packet carrying the RTP
   payload format defined within this memo are confidentiality,
   integrity and source authenticity.  Confidentiality is achieved by
   encryption of the RTP payload.  Integrity of the RTP packets through
   suitable cryptographic integrity protection mechanism.  Cryptographic
   system may also allow the authentication of the source of the
   payload.  A suitable security mechanism for this RTP payload format
   should provide confidentiality, integrity protection and at least
   source authentication capable of determining if an RTP packet is from
   a member of the RTP session or not.

   Note that the appropriate mechanism to provide security to RTP and



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   payloads following this memo may vary.  It is dependent on the
   application, the transport, and the signalling protocol employed.
   Therefore a single mechanism is not sufficient, although if suitable
   the usage of SRTP [RFC3711] is recommended.  Other mechanism that may
   be used are IPsec [RFC4301] and TLS [RFC5246] (RTP over TCP), but
   also other alternatives may exist.

   This RTP payload format and its media decoder do not exhibit any
   significant non-uniformity in the receiver-side computational
   complexity for packet processing, and thus are unlikely to pose a
   denial-of-service threat due to the receipt of pathological data.
   Nor does the RTP payload format contain any active content.


11.  Acknowledgements

   The author would like to thank Xiaodong Duan, Shuaiyu Wang, Rocky
   Wang and Ying Zhang for their initial work in this area.  Many thanks
   also go to Tomas Frankkila for useful input and comments.


12.  References

12.1.  Normative References

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

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

   [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.

   [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
              for Application Designers", BCP 145, RFC 5405,
              November 2008.

   [TS46.002]



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              3GPP, "Specification : 3GPP TS 46.002 http://www.3gpp.org/
              ftp/Specs/archive/46_series/46.002/46002-700.zip",
              June 2007.

   [TS46.020]
              3GPP, "Specification : 3GPP TS 46.020 http://www.3gpp.org/
              ftp/Specs/archive/46_series/46.002/46020-700.zip",
              June 2007.

12.2.  Informative References

   [RFC2198]  Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
              Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
              Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
              September 1997.

   [RFC2326]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
              Streaming Protocol (RTSP)", RFC 2326, April 1998.

   [RFC2974]  Handley, M., Perkins, C., and E. Whelan, "Session
              Announcement Protocol", RFC 2974, October 2000.

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

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

   [RFC4301]  Kent, S. and K. Seo, "Security Architecture for the
              Internet Protocol", RFC 4301, December 2005.

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

   [RFC5109]  Li, A., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, December 2007.

   [RFC5246]  Dierks, T. and E. Rescorla, "The Transport Layer Security
              (TLS) Protocol Version 1.2", RFC 5246, August 2008.











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Authors' Addresses

   Xiaodong Duan
   China Mobile Communications Corporation
   53A, Xibianmennei Ave., Xuanwu District
   Beijing,   100053
   P.R. China

   Phone:
   Fax:
   Email: duanxiaodong@chinamobile.com
   URI:


   Shuaiyu Wang
   China Mobile Communications Corporation
   53A, Xibianmennei Ave., Xuanwu District
   Beijing,   100053
   P.R. China

   Phone:
   Fax:
   Email: wangshuaiyu@chinamobile.com
   URI:


   Magnus Westerlund
   Ericsson AB
   Farogatan 6
   Stockholm,   SE-164 80
   Sweden

   Phone: +46 8 719 0000
   Fax:
   Email: magnus.westerlund@ericsson.com
   URI:


   Karl Hellwig
   Ericsson AB
   Kackertstrasse 7-9
   52072 Aachen
   Germany

   Phone: +49 2407 575-2054
   Email: karl.hellwig@ericsson.com





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   Ingemar Johansson
   Ericsson AB
   Laboratoriegrand 11
   SE-971 28 Lulea
   SWEDEN

   Phone: +46 73 0783289
   Email: ingemar.s.johansson@ericsson.com











































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