path: root/doc/rfc/rfc5993.txt

Internet Engineering Task Force (IETF)                           X. Duan
Request for Comments: 5993                                       S. Wang
Category: Standards Track        China Mobile Communications Corporation
ISSN: 2070-1721                                            M. Westerlund
                                                              K. Hellwig
                                                            I. Johansson
                                                             Ericsson AB
                                                            October 2010


                         RTP Payload Format for
       Global System for Mobile Communications Half Rate (GSM-HR)

Abstract

   This document specifies the payload format for packetization of
   Global System for Mobile Communications Half Rate (GSM-HR) 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 is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc5993.

















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Copyright Notice

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

   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 Simplified BSD License.

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Conventions Used in This Document  . . . . . . . . . . . . . .  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  . . . . . . . . . . . . . . . . .  9
   6.  Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
     6.1.  3 Frames . . . . . . . . . . . . . . . . . . . . . . . . . 10
     6.2.  3 Frames with Lost Frame in the Middle . . . . . . . . . . 11
   7.  Payload Format Parameters  . . . . . . . . . . . . . . . . . . 11
     7.1.  Media Type Definition  . . . . . . . . . . . . . . . . . . 12
     7.2.  Mapping to SDP . . . . . . . . . . . . . . . . . . . . . . 13
       7.2.1.  Offer/Answer Considerations  . . . . . . . . . . . . . 14
       7.2.2.  Declarative SDP Considerations . . . . . . . . . . . . 14
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 15
   9.  Congestion Control . . . . . . . . . . . . . . . . . . . . . . 15
   10. Security Considerations  . . . . . . . . . . . . . . . . . . . 15
   11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 16
   12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 16
     12.1. Normative References . . . . . . . . . . . . . . . . . . . 16
     12.2. Informative References . . . . . . . . . . . . . . . . . . 17





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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 payload format capabilities.  The payload format is
   specified in Section 5.  Examples can be found in Section 6.  The
   media type specification and its mappings to SDP, and considerations
   when using the Session Description Protocol (SDP) offer/answer
   procedures are then specified.  The document ends with considerations
   related to 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 provided back in
   1999 to 2000 in early draft versions of what was later published as
   RFC 3551.  RFC 3551 was based on a later version of the Audio-Visual
   Profile (AVP) draft, which did not provide 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 Used in This Document

   This document uses the normal IETF bit-order representation.  Bit
   fields in figures are read left to right and then down.  The leftmost
   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 Communications (GSM) network provides
   with mobile communication services for nearly 3 billion users
   (statistics as of 2008).  The GSM Half Rate (GSM-HR) codec is one of
   the speech codecs 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
   internally referenced as 06-series.  A simple mapping applies; for
   example, 46.020 is referenced as 06.20, and so on.



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   The GSM-HR codec has a frame length of 20 ms, with narrowband speech
   sampled at 8000 Hz, 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 talkspurt and about
   every 480 ms during speech pauses to allow for a decent comfort noise
   (CN) quality on the 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 the uplink from the mobile station about
   every 480 ms.  In the downlink to the mobile station, when they are
   generated by the encoder of the GSM radio subsystem, SID frames are
   sent every 20 ms to the GSM base station, which then picks only one
   every 480 ms for downlink radio transmission.  For other
   applications, like transport over IP, it is more appropriate to send
   the SID frames less often than every 20 ms, 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 160 ms.

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 to indicate unsent 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 RFC 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.
   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.




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   --+--------+--------+--------+--------+--------+--------+--------+--
     | 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, signaling has been defined to allow the
   sender to voluntarily bound the buffering and delay requirements.  If
   nothing is signaled, the use of this mechanism is allowed and
   unbounded.  For a certain timestamp, the receiver may acquire
   multiple copies of a frame containing encoded audio data.  The cost
   of this scheme is bandwidth, and the receiver delay is 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 regarding 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].  The payload
   format described in this document uses the header fields in a manner
   consistent with that specification.

   The duration of one speech frame is 20 ms.  The sampling frequency is
   8000 Hz, 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




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   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 a speech frame in one packet and as a
   SID frame or as a 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 mandate 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
      the case of Speech frames, with the unused 79 bits all set 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 their order of occurrence.  The first
   bit (b1) of the first parameter is placed in the most significant bit
   (MSB) (bit 0) 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 least significant bit (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 all of the remaining 79 bits to "1".

5.3.  Implementation Considerations

   An application implementing this payload format MUST understand all
   the payload parameters that are 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 160 ms, i.e., every 8th frame, during silent
   periods.  Other SID transmission intervals may occur due to gateways
   to other systems that use 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.










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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 below highlight the payload format.

6.1.  3 Frames

   Below is a basic example of the aggregation of 3 consecutive speech
   frames into a single packet.

      The first 24 bits are ToC elements.

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

   Below is an example of a 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 follows.
      Bits 1..3 are 000 = Good speech frame
      Bits 4..7 are 0000 = Reserved
      Bit 8 is '1', as another ToC element follows.
      Bits 9..11 are 111 = No_Data frame
      Bits 12..15 are 0000 = Reserved
      Bit 16 is '0'; no more ToC elements follow.
      Bits 17..19 are 000 = Good speech frame
      Bits 20..23 are 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 uses
   [RFC4288] as a template.  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-known speech codec.  This media type
   registration covers real-time transfer via RTP.

   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 of RFC 4288
      [RFC4288].

   Security considerations:

      See Section 10 of RFC 5993.

   Interoperability considerations:

      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.







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   Published specifications:

      RFC 5993, 3GPP TS 46.002

   Applications that use this media type:

      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.

   Authors:

      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.



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   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.

   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 the Real Time Streaming Protocol
   (RTSP) [RFC2326] or the Session Announcement Protocol (SAP)
   [RFC2974], the parameters SHALL be interpreted as follows:



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   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 than what has been declared.  More than
      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 it has been
   registered 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 profiles, e.g.,
   "RTP/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, and integrity of the RTP packets through a suitable
   cryptographic integrity protection mechanism.  A 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 whether or not an RTP packet is
   from a member of the RTP session.



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   Note that the appropriate mechanism to provide security to RTP and
   payloads following this may vary.  It is dependent on the
   application, the transport, and the signaling protocol employed.
   Therefore, a single mechanism is not sufficient, although if
   suitable, the usage of the Secure Real-time Transport Protocol (SRTP)
   [RFC3711] is recommended.  Other mechanisms that may be used are
   IPsec [RFC4301] and Transport Layer Security (TLS) [RFC5246] (e.g.,
   for RTP over TCP), but other alternatives may also 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 authors 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.





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   [TS46.002]  3GPP, "Half rate speech; Half rate speech processing
               functions", 3GPP TS 46.002, June 2007, <http://
               www.3gpp.org/ftp/Specs/archive/46_series/46.002/
               46002-700.zip>.

   [TS46.020]  3GPP, "Half rate speech; Half rate speech transcoding",
               3GPP TS 46.020, June 2007, <http://www.3gpp.org/ftp/
               Specs/archive/46_series/46.020/46020-700.zip>.

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
   EMail: duanxiaodong@chinamobile.com


   Shuaiyu Wang
   China Mobile Communications Corporation
   53A, Xibianmennei Ave., Xuanwu District
   Beijing,   100053
   P.R. China
   EMail: wangshuaiyu@chinamobile.com


   Magnus Westerlund
   Ericsson AB
   Farogatan 6
   Stockholm,   SE-164 80
   Sweden
   Phone: +46 8 719 0000
   EMail: magnus.westerlund@ericsson.com


   Karl Hellwig
   Ericsson AB
   Ericsson Allee 1
   52134 Herzogenrath
   Germany
   Phone: +49 2407 575-2054
   EMail: karl.hellwig@ericsson.com


   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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