    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]

    RFC 2658       RTP Payload Format for PureVoice(tm) Audio    August 1999

       Additionally, senders have the following restrictions:

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

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

    3.5 Finding Interleave Group Boundaries

       Given an RTP packet with sequence number S, interleave value (field
       LLL) L, and interleave index value (field NNN) N, the interleave
       group consists of RTP packets with sequence numbers from S-N to S-N+L
       inclusive.  In other words, the Interleave group always consists of
       L+1 RTP packets with sequential sequence numbers.  The bundling value
       for all RTP packets in an interleave group MUST be the same.

       The receiver determines the expected bundling value for all RTP
       packets in an interleave group by the number of CODEC data frames
       bundled in the first RTP packet of the interleave group received.
       Note that this may not be the first RTP packet of the interleave
       group sent if packets are delivered out of order by the underlying
       transport.

       On receipt of an RTP packet in an interleave group with other than
       the expected bundling value, the receiver MAY discard CODEC data
       frames off the end of the RTP packet or add erasure CODEC data frames
       to the end of the packet in order to manufacture a substitute packet
       with the expected bundling value.  The receiver MAY instead choose to
       discard the whole interleave group and play silence.

    3.6 Reconstructing Interleaved Audio

       Given an RTP sequence number ordered set of RTP packets in an
       interleave group numbered 0..L, where L is the interleave value and B
       is the bundling value, and CODEC data frames within each RTP packet
       that are numbered in order from first to last with the numbers 1..B,
       the original, time-ordered sequence of output frames from the CODEC
       may be reconstructed as follows:

       First L+1 frames:
          Frame 0 from packet 0 of interleave group
          Frame 0 from packet 1 of interleave group
          And so on up to...
          Frame 0 from packet L of interleave group

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    RFC 2658       RTP Payload Format for PureVoice(tm) Audio    August 1999

       Second L+1 frames:
          Frame 1 from packet 0 of interleave group
          Frame 1 from packet 1 of interleave group
          And so on up to...
          Frame 1 from packet L of interleave group

       And so on up to...

       Bth L+1 frames:
          Frame B from packet 0 of interleave group
          Frame B from packet 1 of interleave group
          And so on up to...
          Frame B from packet L of interleave group

    3.6.1 Additional Receiver Responsibility

       Assume that the receiver has begun playing frames from an interleave
       group.  The time has come to play frame x from packet n of the
       interleave group.  Further assume that packet n of the interleave
       group has not been received.  As described in section 4, an erasure
       frame will be sent to the Qcelp CODEC.

       Now, assume that packet n of the interleave group arrives before
       frame x+1 of that packet is needed.  Receivers SHOULD use frame x+1
       of the newly received packet n rather than substituting an erasure
       frame.  In other words, just because packet n wasn't available the
       first time it was needed to reconstruct the interleaved audio, the
       receiver SHOULD NOT assume it's not available when it's subsequently
       needed for interleaved audio reconstruction.

    4 Handling lost RTP packets

       The Qcelp CODEC supports the notion of erasure frames.  These are
       frames that for whatever reason are not available.  When
       reconstructing interleaved audio or playing back non-interleaved
       audio, erasure frames MUST be fed to the Qcelp CODEC for all of the
       missing packets.

       Receivers MUST use the timestamp clock to determine how many CODEC
       data frames are missing.  Each CODEC data frame advances the
       timestamp clock EXACTLY 160 counts.

       Since the bundling value may vary (it can only decrease), the
       timestamp clock is the only reliable way to calculate exactly how
       many CODEC data frames are missing when a packet is dropped.

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    RFC 2658       RTP Payload Format for PureVoice(tm) Audio    August 1999

       Specifically when reconstructing interleaved audio, a missing RTP
       packet in the interleave group should be treated as containing B
       erasure CODEC data frames where B is the bundling value for that
       interleave group.

    5 Discussion

       The Qcelp CODEC interpolates the missing audio content when given an
       erasure frame.  However, the best quality is perceived by the
       listener when erasure frames are not consecutive.  This makes
       interleaving desirable as it increases audio quality when dropped
       packets are more likely.

       On the other hand, interleaving can greatly increase the end-to-end
       delay.  Where an interactive session is desired, an interleave (field
       LLL) value of 0 or 1 and a bundling factor of 4 or less is
       recommended.

       When end-to-end delay is not a concern, a bundling value of at least
       4 and an interleave (field LLL) value of 4 or 5 is recommended
       subject to MTU limitations.

       The restrictions on senders set forth in sections 3.3 and 3.4
       guarantee that after receipt of the first payload packet from the
       sender, the receiver can allocate a well-known amount of buffer space
       that will be sufficient for all future reception from the same SSRC
       value.  Less buffer space may be required at some point in the future
       if the sender decreases the bundling value or interleave, but never
       more buffer space.  This prevents the possibility of the receiver
       needing to allocate more buffer space (with the possible result that
       none is available) should the bundling value or interleave value be
       increased by the sender.  Also, were the interleave or bundling value
       to increase, the receiver could be forced to pause playback while it
       receives the additional packets necessary for playback at an
       increased bundling value or increased interleave.

    6 Security Considerations

       RTP packets using the payload format defined in this specification
       are subject to the security considerations discussed in the RTP
       specification [2], and any appropriate profile (for example [4]).
       This implies that confidentiality of the media streams is achieved by
       encryption.  Because the data compression used with this payload
       format is applied end-to-end, encryption may be performed after
       compression so there is no conflict between the two operations.

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    RFC 2658       RTP Payload Format for PureVoice(tm) Audio    August 1999

       A potential denial-of-service threat exists for data encodings using
       compression techniques that have non-uniform receiver-end
       computational load.  The attacker can inject pathological datagrams
       into the stream which are complex to decode and cause the receiver to
       be overloaded.  However, this encoding does not exhibit any
       significant non-uniformity.

       As with any IP-based protocol, in some circumstances, a receiver may
       be overloaded simply by the receipt of too many packets, either
       desired or undesired.  Network-layer authentication may be used to
       discard packets from undesired sources, but the processing cost of
       the authentication itself may be too high.  In a multicast
       environment, pruning of specific sources may be implemented in future
       versions of IGMP [5] and in multicast routing protocols to allow a
       receiver to select which sources are allowed to reach it.

    7 References

       [1]  TIA/EIA/IS-733.  TR45: High Rate Speech Service Option for
            Wideband Spread Spectrum Communications Systems.  Available from
            Global Engineering +1 800 854 7179 or +1 303 792 2181.  May also
            be ordered online at http://www.eia.org/eng/.

       [2]  Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
            "RTP:  A Transport Protocol for Real-Time Applications", RFC
            1889, January 1996.

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

       [4]  Schulzrinne, H., "RTP Profile for Audio and Video Conferences
            with Minimal Control", RFC 1890, January 1996.

       [5]  Deering, S., "Host Extensions for IP Multicasting", STD 5, RFC
            1112, August 1989.

    8 Author's Address

       Kyle J. McKay
       QUALCOMM Incorporated
       5775 Morehouse Drive
       San Diego, CA 92121-1714
       USA

       Phone: +1 858 587 1121
       EMail: kylem@qualcomm.com

    K. McKay                    Standards Track                     [Page 9]

    RFC 2658       RTP Payload Format for PureVoice(tm) Audio    August 1999

    9 Full Copyright Statement

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

       This document and translations of it may be copied and furnished to
       others, and derivative works that comment on or otherwise explain it
       or assist in its implementation may be prepared, copied, published
       and distributed, in whole or in part, without restriction of any
       kind, provided that the above copyright notice and this paragraph are
       included on all such copies and derivative works.  However, this
       document itself may not be modified in any way, such as by removing
       the copyright notice or references to the Internet Society or other
       Internet organizations, except as needed for the purpose of
       developing Internet standards in which case the procedures for
       copyrights defined in the Internet Standards process must be
       followed, or as required to translate it into languages other than
       English.

       The limited permissions granted above are perpetual and will not be
       revoked by the Internet Society or its successors or assigns.

       This document and the information contained herein is provided on an
       "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
       TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
       BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
       HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
       MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

    Acknowledgement

       Funding for the RFC Editor function is currently provided by the
       Internet Society.

    K. McKay                    Standards Track                    [Page 10]