rfc2658.txt

<VC++网络游戏建摸与实现>源代码

Network Working Group                                           K. McKay
Request for Comments: 2658                         QUALCOMM Incorporated
Category: Standards Track                                    August 1999


               RTP Payload Format for PureVoice(tm) Audio

Status of this Memo

   This document specifies an Internet standards track protocol for the
   Internet community, and requests discussion and suggestions for
   improvements.  Please refer to the current edition of the "Internet
   Official Protocol Standards" (STD 1) for the standardization state
   and status of this protocol.  Distribution of this memo is unlimited.

Copyright Notice

   Copyright (C) The Internet Society (1999).  All Rights Reserved.

ABSTRACT

   This document describes the RTP payload format for PureVoice(tm)
   Audio.  The packet format supports variable interleaving to reduce
   the effect of packet loss on audio quality.

1 Introduction

   This document describes how compressed PureVoice audio as produced by
   the Qualcomm PureVoice CODEC [1] may be formatted for use as an RTP
   payload type.  A method is provided to interleave the output of the
   compressor to reduce quality degradation due to lost packets.
   Furthermore, the sender may choose various interleave settings based
   on the importance of low end-to-end delay versus greater tolerance
   for lost packets.

   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 [3].

2 Background

   The Electronic Industries Association (EIA) & Telecommunications
   Industry Association (TIA) standard IS-733 [1] defines an audio
   compression algorithm for use in CDMA applications.  In addition to
   being the standard CODEC for all wireless CDMA terminals, the
   Qualcomm PureVoice CODEC (a.k.a. Qcelp) is used in several Internet
   applications most notably JFax(tm), Apple(r) QuickTime(tm), and
   Eudora(r).



K. McKay                    Standards Track                     [Page 1]

RFC 2658       RTP Payload Format for PureVoice(tm) Audio    August 1999


   The Qcelp CODEC [1] compresses each 20 milliseconds of 8000 Hz, 16-
   bit sampled input speech into one of four different size output
   frames:  Rate 1 (266 bits), Rate 1/2 (124 bits), Rate 1/4 (54 bits)
   or Rate 1/8 (20 bits).  The CODEC chooses the output frame rate based
   on analysis of the input speech and the current operating mode
   (either normal or reduced rate).  For typical speech patterns, this
   results in an average output of 6.8 k bits/sec for normal mode and
   4.7 k bits/sec for reduced rate mode.

3 RTP/Qcelp Packet Format

   The RTP timestamp is in 1/8000 of a second units.  The RTP payload
   data for the Qcelp CODEC has the following format:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                      RTP Header [2]                           |
   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   |RR | LLL | NNN |                                               |
   +-+-+-+-+-+-+-+-+       one or more codec data frames           |
   |                             ....                              |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The RTP header has the expected values as described in [2].  The
   extension bit is not set and this payload type never sets the marker
   bit.  The codec data frames are aligned on octet boundaries.  When
   interleaving is in use and/or multiple codec data frames are present
   in a single RTP packet, the timestamp is, as always, that of the
   oldest data represented in the RTP packet.  The other fields have the
   following meaning:

   Reserved (RR): 2 bits
      MUST be set to zero by sender, SHOULD be ignored by receiver.

   Interleave (LLL): 3 bits
      MUST have a value between 0 and 5 inclusive.  The remaining two
      values (6 and 7) MUST not be used by senders.  If this field is
      non-zero, interleaving is enabled.  All receivers MUST support
      interleaving.  Senders MAY support interleaving.  Senders that do
      not support interleaving MUST set field LLL and NNN to zero.

   Interleave Index (NNN): 3 bits
      MUST have a value less than or equal to the value of LLL.  Values
      of NNN greater than the value of LLL are invalid.






K. McKay                    Standards Track                     [Page 2]

RFC 2658       RTP Payload Format for PureVoice(tm) Audio    August 1999


3.1 Receiving Invalid Values

   On receipt of an RTP packet with an invalid value of the LLL or NNN
   field, the RTP packet MUST be treated as lost by the receiver for the
   purpose of generating erasure frames as described in section 4.

3.2 CODEC data frame format

   The output of the Qcelp CODEC must be converted into CODEC data
   frames for inclusion in the RTP payload as follows:

   a. Octet 0 of the CODEC data frame indicates the rate and total size
      of the CODEC data frame as indicated in this table:

      OCTET 0   RATE      TOTAL CODEC data frame size (in octets)
      -----------------------------------------------------------
        0       Blank     1
        1       1/8       4
        2       1/4       8
        3       1/2       17
        4       1         35
        5       reserved  8 (SHOULD be treated as a reserved value)
       14       Erasure   1 (SHOULD NOT be transmitted by sender)
       other    n/a       reserved

      Receipt of a CODEC data frame with a reserved value in octet 0
      MUST be considered invalid data as described in 3.1.

   b. The bits as numbered in the standard [1] from highest to lowest
      are packed into octets.  The highest numbered bit (265 for Rate 1,
      123 for Rate 1/2, 53 for Rate 1/4 and 19 for Rate 1/8) is placed
      in the most significant bit (Internet bit 0) of octet 1 of the
      CODEC data frame.  The second highest numbered bit (264 for Rate
      1, etc.) is placed in the second most significant bit (Internet
      bit 1) of octet 1 of the data frame.  This continues so that bit
      258 from the standard Rate 1 frame is placed in the least
      significant bit of octet 1.  Bit 257 from the standard is placed
      in the most significant bit of octet 2 and so on until bit 0 from
      the standard Rate 1 frame is placed in Internet bit 1 of octet 34
      of the CODEC data frame.  The remaining unused bits of the last
      octet of the CODEC data frame MUST be set to zero.










K. McKay                    Standards Track                     [Page 3]

RFC 2658       RTP Payload Format for PureVoice(tm) Audio    August 1999


      Here is a detail of how a Rate 1/8 frame is converted into a CODEC
      data frame:
                              CODEC data frame

       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |               |1|1|1|1|1|1|1|1|1|1| | | | | | | | | | | | | | |
      | 1 (Rate 1/8)  |9|8|7|6|5|4|3|2|1|0|9|8|7|6|5|4|3|2|1|0|Z|Z|Z|Z|
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

      Octet 0 of the data frame has value 1 (see table above) indicating
      the total data frame length (including octet 0) is 4 octets.  Bits
      19 through 0 from the standard Rate 1/8 frame are placed as
      indicated with bits marked with "Z" being set to zero.  The Rate
      1, 1/4 and 1/2 standard frames are converted similarly.

3.3 Bundling CODEC data frames

   As indicated in section 3, more than one CODEC data frame MAY be
   included in a single RTP packet by a sender.  Receivers MUST handle
   bundles of up to 10 CODEC data frames in a single RTP packet.

   Furthermore, senders have the following additional restrictions:

   o  MUST not bundle more CODEC data frames in a single RTP packet than
      will fit in the MTU of the RTP transport protocol.  For the
      purpose of computing the maximum bundling value, all CODEC data
      frames should be assumed to have the Rate 1 size.

   o  MUST never bundle more than 10 CODEC data frames in a single RTP
      packet.

   o  Once beginning transmission with a given SSRC and given bundling
      value, MUST NOT increase the bundling value.  If the bundling
      value needs to be increased, a new SSRC number MUST be used.

   o  MAY decrease the bundling value only between interleave groups
      (see section 3.4).  If the bundling value is decreased, it MUST
      NOT be increased (even to the original value), although it may be
      decreased again at a later time.










K. McKay                    Standards Track                     [Page 4]

RFC 2658       RTP Payload Format for PureVoice(tm) Audio    August 1999


3.3.1 Determining the number of bundled CODEC data frames

   Since no count is transmitted as part of the RTP payload and the
   CODEC data frames have differing lengths, the only way to determine
   how many CODEC data frames are present in the RTP packet is to
   examine octet 0 of each CODEC data frame in sequence until the end of
   the RTP packet is reached.

3.4 Interleaving CODEC data frames

   Interleaving is meaningful only when more than one CODEC data frame
   is bundled into a single RTP packet.

   All receivers MUST support interleaving.  Senders MAY support
   interleaving.

   Given a time-ordered sequence of output frames from the Qcelp CODEC
   numbered 0..n, a bundling value B, and an interleave value L where n
   = B * (L+1) - 1, the output frames are placed into RTP packets as
   follows (the values of the fields LLL and NNN are indicated for each
   RTP packet):

   First RTP Packet in Interleave group:
      LLL=L, NNN=0
      Frame 0, Frame L+1, Frame 2(L+1), Frame 3(L+1), ... for a total of
      B frames

   Second RTP Packet in Interleave group:
      LLL=L, NNN=1
      Frame 1, Frame 1+L+1, Frame 1+2(L+1), Frame 1+3(L+1), ... for a
      total of B frames

   This continues to the last RTP packet in the interleave group:

   L+1 RTP Packet in Interleave group:
      LLL=L, NNN=L
      Frame L, Frame L+L+1, Frame L+2(L+1), Frame L+3(L+1), ... for a
      total of B frames

   Senders MUST transmit in timestamp-increasing order.  Furthermore,
   within each interleave group, the RTP packets making up the
   interleave group MUST be transmitted in value-increasing order of the
   NNN field.  While this does not guarantee reduced end-to-end delay on
   the receiving end, when packets are delivered in order by the
   underlying transport, delay will be reduced to the minimum possible.






K. McKay                    Standards Track                     [Page 5]