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RFC 4348

Proposed STD
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Real-Time Transport Protocol (RTP) Payload Format for the Variable-Rate Multimode Wideband (VMR-WB) Audio Codec

Updated by:    4424

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Network Working Group                                          S. Ahmadi
Request for Comments: 4348                                  January 2006
Category: Standards Track

       Real-Time Transport Protocol (RTP) Payload Format for the
         Variable-Rate Multimode Wideband (VMR-WB) Audio Codec

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 (2006).


   This document specifies a real-time transport protocol (RTP) payload
   format to be used for the Variable-Rate Multimode Wideband (VMR-WB)
   speech codec.  The payload format is designed to be able to
   interoperate with existing VMR-WB transport formats on non-IP
   networks.  A media type registration is included for VMR-WB RTP
   payload format.

   VMR-WB is a variable-rate multimode wideband speech codec that has a
   number of operating modes, one of which is interoperable with AMR-WB
   (i.e., RFC 3267) audio codec at certain rates.  Therefore, provisions
   have been made in this document to facilitate and simplify data
   packet exchange between VMR-WB and AMR-WB in the interoperable mode
   with no transcoding function involved.

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Table of Contents

   1. Introduction ....................................................3
   2. Conventions and Acronyms ........................................3
   3. The Variable-Rate Multimode Wideband (VMR-WB) Speech Codec ......4
      3.1. Narrowband Speech Processing ...............................5
      3.2. Continuous vs. Discontinuous Transmission ..................6
      3.3. Support for Multi-Channel Session ..........................6
   4. Robustness against Packet Loss ..................................7
      4.1. Forward Error Correction (FEC) .............................7
      4.2. Frame Interleaving and Multi-Frame Encapsulation ...........8
   5. VMR-WB Voice over IP Scenarios ..................................9
      5.1. IP Terminal to IP Terminal .................................9
      5.2. GW to IP Terminal .........................................10
      5.3. GW to GW (between VMR-WB- and AMR-WB-Enabled Terminals) ...10
      5.4. GW to GW (between Two VMR-WB-Enabled Terminals) ...........11
   6. VMR-WB RTP Payload Formats .....................................12
      6.1. RTP Header Usage ..........................................13
      6.2. Header-Free Payload Format ................................14
      6.3. Octet-Aligned Payload Format ..............................15
           6.3.1. Payload Structure ..................................15
           6.3.2. The Payload Header .................................15
           6.3.3. The Payload Table of Contents ......................18
           6.3.4. Speech Data ........................................20
           6.3.5. Payload Example: Basic Single Channel
                  Payload Carrying Multiple Frames ...................21
      6.4. Implementation Considerations .............................22
           6.4.1. Decoding Validation and Provision for Lost
                  or Late Packets ....................................22
   7. Congestion Control .............................................23
   8. Security Considerations ........................................23
      8.1. Confidentiality ...........................................24
      8.2. Authentication and Integrity ..............................24
   9. Payload Format Parameters ......................................24
      9.1. VMR-WB RTP Payload MIME Registration ......................25
      9.2. Mapping MIME Parameters into SDP ..........................27
      9.3. Offer-Answer Model Considerations .........................28
   10. IANA Considerations ...........................................29
   11. Acknowledgements ..............................................29
   12. References ....................................................30
      12.1. Normative References .....................................30
      12.2. Informative References ...................................30

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1.  Introduction

   This document specifies the payload format for packetization of VMR-
   WB-encoded speech signals into the Real-time Transport Protocol (RTP)
   [3].  The VMR-WB payload formats support transmission of single and
   multiple channels, frame interleaving, multiple frames per payload,
   header-free payload, the use of mode switching, and interoperation
   with existing VMR-WB transport formats on non-IP networks, as
   described in Section 3.

   The payload format is described in Section 6.  The VMR-WB file format
   (i.e., for transport of VMR-WB speech data in storage mode
   applications such as email) is specified in [7].  In Section 9, a
   media type registration for VMR-WB RTP payload format is provided.

   Since VMR-WB is interoperable with AMR-WB at certain rates, an
   attempt has been made throughout this document to maximize the
   similarities with RFC 3267 while optimizing the payload format for
   the non-interoperable modes of the VMR-WB codec.

2.  Conventions and Acronyms

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC2119 [2].

   The following acronyms are used in this document:

    3GPP   - The Third Generation Partnership Project
    3GPP2  - The Third Generation Partnership Project 2
    CDMA   - Code Division Multiple Access
    WCDMA  - Wideband Code Division Multiple Access
    GSM    - Global System for Mobile Communications
    AMR-WB - Adaptive Multi-Rate Wideband Codec
    VMR-WB - Variable-Rate Multimode Wideband Codec
    CMR    - Codec Mode Request
    GW     - Gateway
    DTX    - Discontinuous Transmission
    FEC    - Forward Error Correction
    SID    - Silence Descriptor
    TrFO   - Transcoder-Free Operation
    UDP    - User Datagram Protocol
    RTP    - Real-Time Transport Protocol
    RTCP   - RTP Control Protocol
    MIME   - Multipurpose Internet Mail Extension
    SDP    - Session Description Protocol
    VoIP   - Voice-over-IP

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   The term "interoperable mode" in this document refers to VMR-WB mode
   3, which is interoperable with AMR-WB codec modes 0, 1, and 2.

   The term "non-interoperable modes" in this document refers to VMR-WB
   modes 0, 1, and 2.

   The term "frame-block" is used in this document to describe the
   time-synchronized set of speech frames in a multi-channel VMR-WB
   session.  In particular, in an N-channel session, a frame-block will
   contain N speech frames, one from each of the channels, and all N
   speech frames represent exactly the same time period.

3.  The Variable-Rate Multimode Wideband (VMR-WB) Speech Codec

   VMR-WB is the wideband speech-coding standard developed by Third
   Generation Partnership Project 2 (3GPP2) for encoding/decoding
   wideband/narrowband speech content in multimedia services in 3G CDMA
   cellular systems [1].  VMR-WB is a source-controlled variable-rate
   multimode wideband speech codec.  It has a number of operating modes,
   where each mode is a tradeoff between voice quality and average data
   rate.  The operating mode in VMR-WB (as shown in Table 2) is chosen
   based on the traffic condition of the network and the desired quality
   of service.  The desired average data rate (ADR) in each mode is
   obtained by encoding speech frames at permissible rates (as shown in
   Tables 1 and 3) compliant with CDMA2000 system, depending on the
   instantaneous characteristics of input speech and the maximum and
   minimum rate constraints imposed by the network operator.

   While VMR-WB is a native CDMA codec complying with all CDMA system
   requirements, it is further interoperable with AMR-WB [4,12] at
   12.65, 8.85, and 6.60 kbps.  This is due to the fact that VMR-WB and
   AMR-WB share the same core technology.  This feature enables
   Transcoder-Free (TrFO) interconnections between VMR-WB and AMR-WB
   across different wireless/wireline systems (e.g., GSM/WCDMA and
   CDMA2000) without use of unnecessary complex media format conversion.

   Note that the concept of mode in VMR-WB is different from that of
   AMR-WB where each fixed-rate AMR-WB codec mode is adapted to
   prevailing channel conditions by a tradeoff between the total number
   of source-coding and channel-coding bits.

   VMR-WB is able to transition between various modes with no
   degradation in voice quality that is attributable to the mode
   switching itself.  The operating mode of the VMR-WB encoder may be
   switched seamlessly without prior knowledge of the decoder.  Any
   non-interoperable mode (i.e., VMR-WB modes 0, 1, or 2) can be chosen
   depending on the traffic conditions (e.g., network congestion) and
   the desired quality of service.

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   While in the interoperable mode (i.e., VMR-WB mode 3), mode switching
   between VMR-WB modes is not allowed because there is only one AMR-WB
   interoperable mode in VMR-WB.  Since the AMR-WB codec may request a
   mode change, depending on channel conditions, in-band data included
   in VMR-WB frame structure (see Section 8 of [1] for more details) is
   used during an interoperable interconnection to switch between VMR-WB
   frame types 0, 1, and 2 in VMR-WB mode 3 (corresponding to AMR-WB
   codec modes 0, 1, or 2).

   As mentioned earlier, VMR-WB is compliant with CDMA2000 system with
   the permissible encoding rates shown in Table 1.

   |        Frame Type         | Bits per Packet | Encoding Rate |
   |                           |   (Frame Size)  |     (kbps)    |
   | Full-Rate                 |      266        |     13.3      |
   | Half-Rate                 |      124        |      6.2      |
   | Quarter-Rate              |       54        |      2.7      |
   | Eighth-Rate               |       20        |      1.0      |
   | Blank                     |        0        |       0       |
   | Erasure                   |        0        |       0       |

     Table 1: CDMA2000 system permissible frame types and their
              associated encoding rates

   VMR-WB is robust to high percentage of frame loss and frames with
   corrupted rate information.  The reception of an Erasure
   (SPEECH_LOST) frame type at decoder invokes the built-in frame error
   concealment mechanism.  The built-in frame error concealment
   mechanism in VMR-WB conceals the effect of lost frames by exploiting
   in-band data and the information available in the previous frames.

3.1.  Narrowband Speech Processing

   VMR-WB has the capability to operate with either 16000-Hz or 8000-Hz
   sampled input/output speech signals in all modes of operation [1].
   The VMR-WB decoder does not require a priori knowledge about the
   sampling rate of the original media (i.e., speech/audio signals
   sampled at 8 or 16 kHz) at the input of the encoder.  The VMR-WB
   decoder, by default, generates 16000-Hz wideband output regardless of
   the encoder input sampling frequency.  Depending on the application,
   the decoder can be configured to generate 8000-Hz output, as well.

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   Therefore, while this specification defines a 16000-Hz RTP clock rate
   for VMR-WB codec, the injection and processing of 8000-Hz narrowband
   media during a session is also allowed; however, a 16000-Hz RTP clock
   rate MUST always be used.

   The choice of VMR-WB output sampling frequency depends on the
   implementation and the audio acoustic capabilities of the receiving

3.2.  Continuous vs. Discontinuous Transmission

   The circuit-switched operation of VMR-WB within a CDMA network
   requires continuous transmission of the speech data during a
   conversation.  The intrinsic source-controlled variable-rate feature
   of the CDMA speech codecs is required for optimal operation of the
   CDMA system and interference control.  However, VMR-WB has the
   capability to operate in a discontinuous transmission mode for some
   packet-switched applications over IP networks (e.g., VoIP), where the
   number of transmitted bits and packets during silence period are
   reduced to a minimum.  The VMR-WB DTX operation is similar to that of
   AMR-WB [4,12].

3.3.  Support for Multi-Channel Session

   The octet-aligned RTP payload format defined in this document
   supports multi-channel audio content (e.g., a stereophonic speech
   session).  Although VMR-WB codec itself does not support encoding of
   multi-channel audio content into a single bit stream, it can be used
   to encode and decode each of the individual channels separately.

   To transport the separately encoded multi-channel content, the speech
   frames for all channels that are framed and encoded for the same 20
   ms periods are logically collected in a frame-block.

   At the session setup, out-of-band signaling must be used to indicate
   the number of channels in the session and the order of the speech
   frames from different channels in each frame-block.  When using SDP
   for signaling (see Section 9.2 for more details), the number of
   channels is specified in the rtpmap attribute, and the order of
   channels carried in each frame-block is implied by the number of
   channels as specified in Section 4.1 in [6].

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4.  Robustness against Packet Loss

   The octet-aligned payload format described in this document (see
   Section 6 for more details) supports several features, including
   forward error correction (FEC) and frame interleaving, in order to
   increase robustness against lost packets.

4.1.  Forward Error Correction (FEC)

   The simple scheme of repetition of previously sent data is one way of
   achieving FEC.  Another possible scheme, which is more bandwidth
   efficient, is to use payload-external FEC; e.g., RFC2733 [8], which
   generates extra packets containing repair data.

   The repetition method involves the simple retransmission of
   previously transmitted frame-blocks together with the current frame-
   block(s).  This is done by using a sliding window to group the speech
   frame-blocks to send in each payload.  Figure 1 illustrates an

   In this example, each frame-block is retransmitted one time in the
   following RTP payload packet.  Here, f(n-2)..f(n+4) denotes a
   sequence of speech frame-blocks, and p(n-1)..p(n+4) a sequence of
   payload packets.

     | 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

   The use of this approach does not require signaling at the session
   setup.  In other words, the speech sender can choose to use this
   scheme without consulting the receiver.  This is because a packet
   containing redundant frames will not look different from a packet
   with only new frames.  The receiver may receive multiple copies or
   versions of a frame for a certain timestamp if no packet is lost.  If
   multiple versions of the same speech frame are received, it is
   RECOMMENDED that the highest rate be used by the speech decoder.

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   This redundancy scheme provides the same functionality as that
   described in RFC 2198, "RTP Payload for Redundant Audio Data" [10].
   In most cases, the mechanism in this payload format is more efficient
   and simpler than requiring both endpoints to support RFC 2198.  If
   the spread in time required between the primary and redundant
   encodings is larger than 5 frame times, the bandwidth overhead of RFC
   2198 will be lower.

   The sender is responsible for selecting an appropriate amount of
   redundancy based on feedback about the channel (e.g., in RTCP
   receiver reports) or network traffic.  A sender SHOULD NOT base
   selection of FEC on the CMR, as this parameter most probably was set
   based on non-IP information.  The sender is also responsible for
   avoiding congestion, which may be aggravated by redundant
   transmission (see Section 7).

4.2.  Frame Interleaving and Multi-Frame Encapsulation

   To decrease protocol overhead, the octet-aligned payload format,
   described in Section 6, allows several speech frame-blocks to be
   encapsulated into a single RTP packet.  One of the drawbacks of this
   approach is that in case of packet loss several consecutive speech
   frame-blocks are lost, which usually causes clearly audible
   distortion in the reconstructed speech.

   Interleaving of frame-blocks can improve the speech quality in such
   cases by distributing the consecutive losses into a series of single
   frame-block losses.  However, interleaving and bundling several
   frame-blocks per payload will also increase end-to-end delay and is
   therefore not appropriate for all types of applications.  Streaming
   applications will most likely be able to exploit interleaving to
   improve speech quality in lossy transmission conditions.

   The octet-aligned payload format supports the use of frame
   interleaving as an option.  For the encoder (speech sender) to use
   frame interleaving in its outbound RTP packets for a given session,
   the decoder (speech receiver) needs to indicate its support via out-
   of-band means (see Section 9).

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5.  VMR-WB Voice over IP Scenarios

5.1.  IP Terminal to IP Terminal

   The primary scenario for this payload format is IP end-to-end between
   two terminals incorporating VMR-WB codec, as shown in Figure 2.
   Nevertheless, this scenario can be generalized to an interoperable
   interconnection between VMR-WB-enabled and AMR-WB-enabled IP
   terminals using the offer-answer model described in Section 9.3.
   This payload format is expected to be useful for both conversational
   and streaming services.

       +----------+                         +----------+
       |          |                         |          |
       | TERMINAL |<----------------------->| TERMINAL |
       |          |    VMR-WB/RTP/UDP/IP    |          |
       +----------+                         +----------+
                     (or AMR-WB/RTP/UDP/IP)

          Figure 2: IP terminal to IP terminal

   A conversational service puts requirements on the payload format.
   Low delay is a very important factor, i.e., fewer speech frame-blocks
   per payload packet.  Low overhead is also required when the payload
   format traverses across low bandwidth links, especially if the
   frequency of packets will be high.

   Streaming service has less strict real-time requirements and
   therefore can use a larger number of frame-blocks per packet than
   conversational service.  This reduces the overhead from IP, UDP, and
   RTP headers.  However, including several frame-blocks per packet
   makes the transmission more vulnerable to packet loss, so
   interleaving may be used to reduce the effect of packet loss on
   speech quality.  A streaming server handling a large number of
   clients also needs a payload format that requires as few resources as
   possible when doing packetization.

   For VMR-WB-enabled IP terminals at both ends, depending on the
   implementation, all modes of the VMR-WB codec can be used in this
   scenario.  Also, both header-free and octet-aligned payload formats
   (see Section 6 for details) can be utilized.  For the interoperable
   interconnection between VMR-WB and AMR-WB, only VMR-WB mode 3 is
   used, and all restrictions described in Section 9.3 apply.

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5.2.  GW to IP Terminal

   Another scenario occurs when VMR-WB-encoded speech will be
   transmitted from a non-IP system (e.g., 3GPP2/CDMA2000 network) to an
   IP terminal, and/or vice versa, as depicted in Figure 3.

       VMR-WB over
   3GPP2/CDMA2000 network
                      +------+                        +----------+
                      |      |                        |          |
      <-------------->|  GW  |<---------------------->| TERMINAL |
                      |      |   VMR-WB/RTP/UDP/IP    |          |
                      +------+                        +----------+
                          |           IP network

                   Figure 3: GW to VoIP terminal scenario

   VMR-WB's capability to switch seamlessly between operational modes is
   exploited in CDMA (non-IP) networks to optimize speech quality for a
   given traffic condition.  To preserve this functionality in scenarios
   including a gateway to an IP network using the octet-aligned payload
   format, a codec mode request (CMR) field is considered.  The gateway
   will be responsible for forwarding the CMR between the non-IP and IP
   parts in both directions.  The IP terminal SHOULD follow the CMR
   forwarded by the gateway to optimize speech quality going to the
   non-IP decoder.  The mode control algorithm in the gateway SHOULD
   accommodate the delay imposed by the IP network on the response to
   CMR by the IP terminal.

   The IP terminal SHOULD NOT set the CMR (see Section 6.3.2), but the
   gateway can set the CMR value on frames going toward the encoder in
   the non-IP part to optimize speech quality from that encoder to the
   gateway and to perform congestion control on the IP network.

5.3.  GW to GW (between VMR-WB- and AMR-WB-Enabled Terminals)

   A third likely scenario is that RTP/UDP/IP is used as transport
   between two non-IP systems, i.e., IP is originated and terminated in
   gateways on both sides of the IP transport, as illustrated in Figure
   4.  This is the most likely scenario for an interoperable
   interconnection between 3GPP/(GSM-WCDMA)/AMR-WB and
   3GPP2/CDMA2000/VMR-WB-enabled mobile stations.  In this scenario, the
   VMR-WB-enabled terminal also declares itself capable of AMR-WB with
   restricted mode set as described in Section 9.3. The CMR value may be
   set in packets received by the gateways on the IP network side.  The
   gateway should forward to the non-IP side a CMR value that is the

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   minimum of three values: (1) the CMR value it receives on the IP
   side; (2) a CMR value it may choose for congestion control of
   transmission on the IP side; and (3) the CMR value based on its
   estimate of reception quality on the non-IP side.  The details of the
   traffic control algorithm are left to the implementation.

      VMR-WB over                                       AMR-WB over
   3GPP2/CDMA2000 network                      3GPP/(GSM-WCDMA) network

                     +------+                  +------+
    (AMR-WB Payload) |      | AMR-WB/RTP/UDP/IP|      |(AMR-WB Payload)
   <---------------->|  GW  |<---------------->|  GW  |<--------------->
                     |      |                  |      |
                     +------+                  +------+
                        |        IP network       |
                        |                         |

               Figure 4: GW to GW scenario (AMR-WB <-> VMR-WB
                      interoperable interconnection)

   During and upon initiation of an interoperable interconnection
   between VMR-WB and AMR-WB, only VMR-WB mode 3 can be used.  There are
   three Frame Types (i.e., FT=0, 1, or 2; see Table 3) within this mode
   that are compatible with AMR-WB codec modes 0, 1, and 2,
   respectively.  If the AMR-WB codec is engaged in an interoperable
   interconnection with VMR-WB, the active AMR-WB codec mode set needs
   to be limited to 0, 1, and 2.

5.4.  GW to GW (between Two VMR-WB-Enabled Terminals)

   The fourth example VoIP scenario is composed of a RTP/UDP/IP
   transport between two non-IP systems; i.e., IP is originated and
   terminated in gateways on both sides of the IP transport, as
   illustrated in Figure 5.  This is the most likely scenario for
   Mobile-Station-to-Mobile-Station (MS-to-MS) Transcoder-Free (TrFO)
   interconnection between two 3GPP2/CDMA2000 terminals that both use
   VMR-WB codec.

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        VMR-WB over                                     VMR-WB over
   3GPP2/CDMA2000 network                         3GPP2/CDMA2000 network

                      +------+                   +------+
                      |      |                   |      |
        <------------>|  GW  |<----------------->|  GW  |<------------>
                      |      | VMR-WB/RTP/UDP/IP |      |
                      +------+                   +------+
                          |         IP network       |
                          |                          |

        Figure 5: GW to GW scenario (a CDMA2000 MS-to-MS VoIP scenario)

6.  VMR-WB RTP Payload Formats

   For a given session, the payload format can be either header free or
   octet aligned, depending on the mode of operation that is established
   for the session via out-of-band means and the application.

   The header-free payload format is designed for maximum bandwidth
   efficiency, simplicity, and low latency.  Only one codec data frame
   can be sent in each header-free payload format packet.  None of the
   payload header fields or table of contents (ToC) entries is present
   (the same consideration is also made in [11]).

   In the octet-aligned payload format, all the fields in a payload,
   including payload header, table of contents entries, and speech
   frames themselves, are individually aligned to octet boundaries to
   make implementations efficient.

   Note that octet alignment of a field or payload means that the last
   octet is padded with zeroes in the least significant bits to fill the
   octet.  Also note that this padding is separate from padding
   indicated by the P bit in the RTP header.

   Between the two payload formats, only the octet-aligned format has
   the capability to use the interleaving to make the speech transport
   robust to packet loss.

   The VMR-WB octet-aligned payload format in the interoperable mode is
   identical to that of AMR-WB (i.e., RFC 3267).

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6.1.  RTP Header Usage

   The format of the RTP header is specified in [3].  This payload
   format uses the fields of the header in a manner consistent with that

   The RTP timestamp corresponds to the sampling instant of the first
   sample encoded for the first frame-block in the packet.  The
   timestamp clock frequency is the same as the default sampling
   frequency (i.e., 16 kHz), so the timestamp unit is in samples.

   The duration of one speech frame-block is 20 ms for VMR-WB.  For
   normal wideband operation of VMR-WB, the input/output media sampling
   frequency is 16 kHz, corresponding to 320 samples per frame from each
   channel.  Thus, the timestamp is increased by 320 for VMR-WB for each
   consecutive frame-block.

   The VMR-WB codec is capable of processing speech/audio signals
   sampled at 8 kHz.  By default, the VMR-WB decoder output sampling
   frequency is 16 kHz.  Depending on the application, the decoder can
   be configured to generate 8-kHz output sampling frequency, as well.
   Since the VMR-WB RTP payload formats for the 8- and 16-kHz sampled
   media are identical and the VMR-WB decoder does not need a priori
   knowledge about the encoder input sampling frequency, a fixed RTP
   clock rate of 16000 Hz is defined for VMR-WB codec.  This would allow
   injection or processing of 8-kHz sampled speech/audio media without
   having to change the RTP clock rate during a session.  Note that the
   timestamp is incremented by 320 per frame-block for 8-kHz sampled
   media, as well.

   A packet may contain multiple frame-blocks of encoded speech or
   comfort noise parameters.  If interleaving is employed, the frame-
   blocks encapsulated into a payload are picked according to the
   interleaving rules defined in Section 6.3.2. Otherwise, each packet
   covers a period of one or more contiguous 20-ms frame-block
   intervals.  In case the data from all the channels for a particular
   frame-block in the period is missing (for example, at a gateway from
   some other transport format), it is possible to indicate that no data
   is present for that frame-block instead of breaking a multi-frame-
   block packet into two, as explained in Section 6.3.2.

   No matter which payload format is used, the RTP payload is always
   made an integral number of octets long by padding with zero bits if
   necessary.  If additional padding is required to bring the payload
   length to a larger multiple of octets or for some other purpose, then
   the P bit in the RTP header MAY be set, and padding appended, as
   specified in [3].

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   The RTP header marker bit (M) SHALL be always set to 0 if the VMR-WB
   codec operates in continuous transmission.  When operating in
   discontinuous transmission (DTX), the RTP header marker bit SHALL be
   set to 1 if the first frame-block carried in the packet contains a
   speech frame, which is the first in a talkspurt.  For all other
   packets, the marker bit SHALL be set to zero (M=0).

   The assignment of an RTP payload type for this payload format is
   outside the scope of this document and will not be specified here.
   It is expected that the RTP profile under which this payload format
   is being used will assign a payload type for this encoding or specify
   that the payload type is to be bound dynamically (see Section 9).

6.2.  Header-Free Payload Format

   The header-free payload format is designed for maximum bandwidth
   efficiency, simplicity, and minimum delay.  Only one speech data
   frame presents in each header-free payload format packet.  None of
   the payload header fields or ToC entries is present.  The encoding
   rate for the speech frame can be determined from the length of the
   speech data frame, since there is only one speech data frame in each
   header-free payload format.

   The use of the RTP header fields for header-free payload format is
   the same as the corresponding one for the octet-aligned payload
   format.  The detailed bit mapping of speech data packets permissible
   for this payload format is described in Section 8 of [1].  Since the
   header-free payload format is not compatible with AMR-WB RTP payload,
   only non-interoperable modes of VMR-WB SHALL be used with this
   payload format.  That is, FT=0, 1, 2, and 9 SHALL NOT be used with
   header-free payload 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 [3]                           |
   |                                                               |
   +          ONLY one speech data frame           +-+-+-+-+-+-+-+-+
   |                                               |

   Note that the mode of operation, using this payload format, is
   decided by the transmitting (encoder) site.  The default mode of
   operation for VMR-WB encoder is mode 0 [1].  The mode change request
   MAY also be sent through non-RTP means, which is out of the scope of
   this specification.

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6.3.  Octet-Aligned Payload Format

6.3.1.  Payload Structure

   The complete payload consists of a payload header, a payload table of
   contents, and speech data representing one or more speech frame-
   blocks.  The following diagram shows the general payload format

   | Payload header | Table of contents | Speech data ...

6.3.2.  The Payload Header

   In octet-aligned payload format, the payload header consists of a
   4-bit CMR, 4 reserved bits, and, optionally, an 8-bit interleaving
   header, as shown below.

    0                   1
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
   +-+-+-+-+-+-+-+-+- - - - - - - -
   |  CMR  |R|R|R|R|  ILL  |  ILP  |
   +-+-+-+-+-+-+-+-+- - - - - - - -

   CMR (4 bits): This indicates a codec mode request sent to the speech
   encoder at the site of the receiver of this payload.  CMR value 15
   indicates that no mode request is present, and other unused values
   are reserved for future use.

   The value of the CMR field is set according to the following table:

   | CMR   |                 VMR-WB Operating Modes                   |
   |   0   | VMR-WB mode 3 (AMR-WB interoperable mode at 6.60 kbps)   |
   |   1   | VMR-WB mode 3 (AMR-WB interoperable mode at 8.85 kbps)   |
   |   2   | VMR-WB mode 3 (AMR-WB interoperable mode at 12.65 kbps)  |
   |   3   | VMR-WB mode 2                                            |
   |   4   | VMR-WB mode 1                                            |
   |   5   | VMR-WB mode 0                                            |
   |   6   | VMR-WB mode 2 with maximum half-rate encoding            |
   | 7-14  | (reserved)                                               |
   |  15   | No Preference (no mode request is present)               |

     Table 2: List of valid CMR values and their associated VMR-WB
              operating modes

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   R: This is a reserved bit that MUST be set to zero.  The receiver
   MUST ignore all R bits.

   ILL (4 bits, unsigned integer): This is an OPTIONAL field that is
   present only if interleaving is signaled out-of-band for the session.
   ILL=L indicates to the receiver that the interleaving length is L+1,
   in number of frame-blocks.

   ILP (4 bits, unsigned integer): This is an OPTIONAL field that is
   present only if interleaving is signaled.  ILP MUST take a value
   between 0 and ILL, inclusive, indicating the interleaving index for
   frame-blocks in this payload in the interleave group.  If the value
   of ILP is found greater than ILL, the payload SHOULD be discarded.

   ILL and ILP fields MUST be present in each packet in a session if
   interleaving is signaled for the session.

   The mode request received in the CMR field is valid until the next
   CMR is received, i.e., until a newly received CMR value overrides the
   previous one.  Therefore, if a terminal continuously wishes to
   receive frames in the same mode, x, it needs to set CMR=x for all its
   outbound payloads, and if a terminal has no preference in which mode
   to receive, it SHOULD set CMR=15 in all its outbound payloads.

   If a payload is received with a CMR value that is not valid, the CMR
   MUST be ignored by the receiver.

   In a multi-channel session, CMR SHOULD be interpreted by the receiver
   of the payload as the desired encoding mode for all the channels in
   the session, if the network allows.

   There are two factors that affect the VMR-WB mode selection: (i) the
   performance of any CDMA link connected via a gateway (e.g., in a GW
   to IP terminal scenario), and (ii) the congestion state of an IP
   network.  The CDMA link performance is signaled via the CMR field,
   which is not used by IP-only end-points.  The IP network state is
   monitored using, for example, RTCP.  A sender needs to select the
   operating mode to satisfy both these constraints (see Section 7).

   The encoder SHOULD follow a received mode request, but MAY change to
   a different mode if the network necessitates it, for example, to
   control congestion.

   The CMR field MUST be set to 15 for packets sent to a multicast
   group.  The encoder in the speech sender SHOULD ignore mode requests
   when sending speech to a multicast session but MAY use RTCP feedback
   information as a hint that a mode change is needed.

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   If interleaving option is utilized, interleaving MUST be performed on
   a frame-block basis, as opposed to a frame basis, in a multi-channel

   The following example illustrates the arrangement of speech frame-
   blocks in an interleave group during an interleave session.  Here we
   assume ILL=L for the interleave group that starts at speech frame-
   block n.  We also assume that the first payload packet of the
   interleave group is s and the number of speech frame-blocks carried
   in each payload is N.  Then we will have

    Payload s (the first packet of this interleave group):
      ILL=L, ILP=0,

    Carry frame-blocks: n, n+(L+1), n+2*(L+1),..., n+(N-1)*(L+1)

    Payload s+1 (the second packet of this interleave group):
      ILL=L, ILP=1,
      Carry frame-blocks: n+1, n+1+(L+1), n+1+2*(L+1),..., n+1+


    Payload s+L (the last packet of this interleave group):
      ILL=L, ILP=L,
      Carry frame-blocks: n+L, n+L+(L+1), n+L+2*(L+1), ..., n+L+

   The next interleave group will start at frame-block n+N*(L+1).  There
   will be no interleaving effect unless the number of frame-blocks per
   packet (N) is at least 2.  Moreover, the number of frame-blocks per
   payload (N) and the value of ILL MUST NOT be changed inside an
   interleave group.  In other words, all payloads in an interleave
   group MUST have the same ILL and MUST contain the same number of
   speech frame-blocks.

   The sender of the payload MUST only apply interleaving if the
   receiver has signaled its use through out-of-band means.  Since
   interleaving will increase buffering requirements at the receiver,
   the receiver uses MIME parameter "interleaving=I" to set the maximum
   number of frame-blocks allowed in an interleaving group to I.

   When performing interleaving, the sender MUST use a proper number of
   frame-blocks per payload (N) and ILL so that the resulting size of an
   interleave group is less than or equal to I, i.e., N*(L+1)<=I.

   The following example shows the ToC of three consecutive packets,
   each carrying 3 frame-blocks, in an interleaved two-channel session.

Top      ToC       Page 18 
   Here, the two channels are left (L) and right (R), with L coming
   before R, and the interleaving length is 3 (i.e., ILL=2).  This makes
   the interleave group 9 frame-blocks large.

   Packet #1

   ILL=2, ILP=0:
   | 1L | 1R | 4L | 4R | 7L | 7R |
      Frame     Frame     Frame
     Block 1   Block 4   Block 7

   Packet #2

   ILL=2, ILP=1:

   | 2L | 2R | 5L | 5R | 8L | 8R |
      Frame     Frame     Frame
     Block 2   Block 5   Block 8

   Packet #3

   ILL=2, ILP=2:
   | 3L | 3R | 6L | 6R | 9L | 9R |
         Frame     Frame     Frame
        Block 3   Block 6   Block 9

6.3.3.  The Payload Table of Contents

   The table of contents (ToC) in octet-aligned payload format consists
   of a list of ToC entries where each entry corresponds to a speech
   frame carried in the payload, i.e., when interleaving is used, the
   frame-blocks in the ToC will almost never be placed consecutive in
   time.  Instead, the presence and order of the frame-blocks in a
   packet will follow the pattern described in 6.3.2.

Top      ToC       Page 19 
   | list of ToC entries |

   A ToC entry for the octet-aligned payload format is as follows:

    0 1 2 3 4 5 6 7
   |F|  FT   |Q|P|P|

   The table of contents (ToC) consists of a list of ToC entries, each
   representing a speech frame.

   F (1 bit):   If set to 1, indicates that this frame is followed by
                another speech frame in this payload; if set to 0,
                indicates that this frame is the last frame in this

   FT (4 bits): Frame type index whose value is chosen according to
                Table 3.

                During the interoperable mode, FT=14 (SPEECH_LOST) and
                FT=15 (NO_DATA) are used to indicate frames that are
                either lost or not being transmitted in this payload,
                respectively.  FT=14 or 15 MAY be used in the non-
                interoperable modes to indicate frame erasure or blank
                frame, respectively (see Section 2.1 of [1]).

                If a payload with an invalid FT value is received, the
                payload MUST be discarded.  Note that for ToC entries
                with FT=14 or 15, there will be no corresponding speech
                frame in the payload.

                Depending on the application and the mode of operation
                of VMR-WB, any combination of the permissible frame
                types (FT) shown in Table 3 MAY be used.

   Q (1 bit):   Frame quality indicator.  If set to 0, indicates that
                the corresponding frame is corrupted.  During the
                interoperable mode, the receiver side (with AMR-WB
                codec) should set the RX_TYPE to either SPEECH_BAD or
                SID_BAD depending on the frame type (FT), if Q=0.  The
                VMR-WB encoder always sets Q bit to 1.  The VMR-WB
                decoder may ignore the Q bit.

   P bits:      Padding bits MUST be set to zero and MUST be ignored by
                a receiver.

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   | FT |                Encoding Rate               |Frame Size (Bits)|
   | 0  | Interoperable Full-Rate (AMR-WB 6.60 kbps) |       132       |
   | 1  | Interoperable Full-Rate (AMR-WB 8.85 kbps) |       177       |
   | 2  | Interoperable Full-Rate (AMR-WB 12.65 kbps)|       253       |
   | 3  | Full-Rate 13.3 kbps                        |       266       |
   | 4  | Half-Rate 6.2 kbps                         |       124       |
   | 5  | Quarter-Rate 2.7 kbps                      |        54       |
   | 6  | Eighth-Rate 1.0 kbps                       |        20       |
   | 7  | (reserved)                                 |         -       |
   | 8  | (reserved)                                 |         -       |
   | 9  | CNG (AMR-WB SID)                           |        40       |
   | 10 | (reserved)                                 |         -       |
   | 11 | (reserved)                                 |         -       |
   | 12 | (reserved)                                 |         -       |
   | 13 | (reserved)                                 |         -       |
   | 14 | Erasure (AMR-WB SPEECH_LOST)               |         0       |
   | 15 | Blank (AMR-WB NO_DATA)                     |         0       |

      Table 3: VMR-WB payload frame types for real-time transport

   For multi-channel sessions, the ToC entries of all frames from a
   frame-block are placed in the ToC in consecutive order.  Therefore,
   with N channels and K speech frame-blocks in a packet, there MUST be
   N*K entries in the ToC, and the first N entries will be from the
   first frame-block, the second N entries will be from the second
   frame-block, and so on.

6.3.4.  Speech Data

   Speech data of a payload contains one or more speech frames as
   described in the ToC of the payload.

   Each speech frame represents 20 ms of speech encoded in one of the
   available encoding rates depending on the operation mode.  The length
   of the speech frame is defined by the frame type in the FT field,
   with the following considerations:

   - The last octet of each speech frame MUST be padded with zeroes at
     the end if not all bits in the octet are used.  In other words,
     each speech frame MUST be octet-aligned.

   - When multiple speech frames are present in the speech data, the
     speech frames MUST be arranged one whole frame after another.

Top      ToC       Page 21 
   The order and numbering notation of the speech data bits are as
   specified in the VMR-WB standard specification [1].

   The payload begins with the payload header of one octet, or two if
   frame interleaving is selected.  The payload header is followed by
   the table of contents consisting of a list of one-octet ToC entries.

   The speech data follows the table of contents.  For the purpose of
   packetization, all the octets comprising a speech frame are appended
   to the payload as a unit.  The speech frames are packed in the same
   order as their corresponding ToC entries are arranged in the ToC
   list, with the exception that if a given frame has a ToC entry with
   FT=14 or 15, there will be no data octets present for that frame.

6.3.5.  Payload Example: Basic Single Channel Payload Carrying Multiple

   The following diagram shows an octet-aligned payload format from a
   single channel session that carries two VMR-WB Full-Rate frames
   (FT=3).  In the payload, a codec mode request is sent (e.g., CMR=4),
   requesting that the encoder at the receiver's side use VMR-WB mode 1.
   No interleaving is used.  Note that in the example below the last
   octet in both speech frames is padded with zeros to make them octet

    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
   | CMR=4 |R|R|R|R|1|FT#1=3 |Q|P|P|0|FT#2=3 |Q|P|P|   f1(0..7)    |
   |   f1(8..15)   |  f1(16..23)   |  ...                          |
   : ...                                                           :
   | r |P|P|P|P|P|P|  f2(0..7)     |   f2(8..15)   |  f2(16..23)   |
   : ...                                                           :
   |                        ...    | l |P|P|P|P|P|P|

      r= f1(264,265)
      l= f2(264,265)

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6.4.  Implementation Considerations

   An application implementing this payload format MUST understand all
   the payload parameters.  Any mapping of the parameters to a signaling
   protocol MUST support all parameters.  Therefore, 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.

   To enable efficient interoperable interconnection with AMR-WB and to
   ensure that a VMR-WB terminal appropriately declares itself as a
   AMR-WB-capable terminal (see Section 9.3), it is also RECOMMENDED
   that a VMR-WB RTP payload implementation understand relevant AMR-WB

   To further ensure interoperability between various implementations of
   VMR-WB, implementations SHALL support both header-free and octet-
   aligned payload formats.  Support of interleaving is optional.

6.4.1.  Decoding Validation and Provision for Lost or Late Packets

   When processing a received payload packet, if the receiver finds that
   the calculated payload length, based on the information of the
   session and the values found in the payload header fields, does not
   match the size of the received packet, the receiver SHOULD discard
   the packet to avoid potential degradation of speech quality and to
   invoke the VMR-WB built-in frame error concealment mechanism.
   Therefore, invalid packets SHALL be treated as lost packets.

   Late packets (i.e., the unavailability of a packet when it is needed
   for decoding at the receiver) should be treated as lost packets.
   Furthermore, if the late packet is part of an interleave group,
   depending upon the availability of the other packets in that
   interleave group, decoding must be resumed from the next available
   frame (sequential order).  In other words, the unavailability of a
   packet in an interleave group at a certain time should not invalidate
   the other packets within that interleave group that may arrive later.

Top      ToC       Page 23 
7.  Congestion Control

   The general congestion control considerations for transporting RTP
   data apply to VMR-WB speech over RTP as well.  However, the multimode
   capability of VMR-WB speech codec may provide an advantage over other
   payload formats for controlling congestion since the bandwidth demand
   can be adjusted by selecting a different operating mode.

   Another parameter that may impact the bandwidth demand for VMR-WB is
   the number of frame-blocks that are encapsulated in each RTP payload.
   Packing more frame-blocks in each RTP payload can reduce the number
   of packets sent and hence the overhead from RTP/UDP/IP headers, at
   the expense of increased delay.

   If forward error correction (FEC) is used to alleviate the packet
   loss, the amount of redundancy added by FEC will need to be regulated
   so that the use of FEC itself does not cause a congestion problem.

   Congestion control for RTP SHALL be used in accordance with RFC 3550
   [3] and any applicable RTP profile, for example, RFC 3551 [6].  This
   means that congestion control is required for any transmission over
   unmanaged best-effort networks.

   Congestion on the IP network is managed by the IP sender.  Feedback
   about congestion SHOULD be provided to that IP sender through RTCP or
   other means, and then the sender can choose to avoid congestion using
   the most appropriate mechanism.  That may include selecting an
   appropriate operating mode, but also includes adjusting the level of
   redundancy or number of frames per packet.

8.  Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the general security considerations discussed in RTP
   [3] and any applicable profile such as AVP [9] or SAVP [10].

   As this format transports encoded audio, the main security issues
   include confidentiality, integrity protection, and data origin
   authentication of the audio itself.  The payload format itself does
   not have any built-in security mechanisms.  Any suitable external
   mechanisms, such as SRTP [10], MAY be used.

   This payload format and the VMR-WB decoder do not exhibit any
   significant non-uniformity in the receiver-side computational
   complexity for packet processing; thus, they are unlikely to pose a
   denial-of-service threat due to the receipt of pathological data.

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8.1.  Confidentiality

   In order to ensure confidentiality of the encoded audio, all audio
   data bits MUST be encrypted.  There is less need to encrypt the
   payload header or the table of contents since they only carry
   information about the frame type.  This information could also be
   useful to a third party, for example, for quality monitoring.

   The use of interleaving in conjunction with encryption can have a
   negative impact on the confidentiality for a short period of time.
   Consider the following packets (in brackets) containing frame numbers
   as indicated: {10, 14, 18}, {13, 17, 21}, {16, 20, 24} (a typical
   continuous diagonal interleaving pattern).  The originator wishes to
   deny some participants the ability to hear material starting at time
   16.  Simply changing the key on the packet with the timestamp at or
   after 16, and denying the new key to those participants, does not
   achieve this; frames 17, 18, and 21 have been supplied in prior
   packets under the prior key, and error concealment may make the audio
   intelligible at least as far as frame 18 or 19, and possibly further.

8.2.  Authentication and Integrity

   To authenticate the sender of the speech, an external mechanism MUST
   be used.  It is RECOMMENDED that such a mechanism protects both the
   complete RTP header and the payload (speech and data bits).

   Data tampering by a man-in-the-middle attacker could replace audio
   content and also result in erroneous depacketization/decoding that
   could lower the audio quality.  For example, tampering with the CMR
   field may result in speech of a different quality than desired.

9.  Payload Format Parameters

   This section defines the parameters that may be used to select
   optional features in the VMR-WB RTP payload formats.

   The parameters are defined here as part of the MIME subtype
   registration for the VMR-WB speech codec.  A mapping of the
   parameters into the Session Description Protocol (SDP) [5] is also
   provided for those applications that use SDP.  In control protocols
   that do not use MIME or SDP, the media type parameters must be mapped
   to the appropriate format used with that control protocol.

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9.1.  VMR-WB RTP Payload MIME Registration

   The MIME subtype for the Variable-Rate Multimode Wideband (VMR-WB)
   audio codec is allocated from the IETF tree since VMR-WB is expected
   to be a widely used speech codec in multimedia streaming and
   messaging as well as in VoIP applications.  This MIME registration
   only covers real-time transfers via RTP.

   Note, the receiver MUST ignore any unspecified parameter and use the
   default values instead.  Also note that if no input parameters are
   defined, the default values will be used.

     Media Type name:      audio

     Media subtype name:   VMR-WB

     Required parameters:  none

   Furthermore, if the interleaving parameter is present, the parameter
   "octet-align=1" MUST also be present.

OPTIONAL parameters:

  mode-set:       Requested VMR-WB operating mode set.  Restricts
                  the active operating modes to a subset of all
                  modes.  Possible values are a comma-separated
                  list of integer values.  Currently, this list
                  includes modes 0, 1, 2, and 3 [1], but MAY be
                  extended in the future.  If such mode-set is
                  specified during session initiation, the encoder
                  MUST NOT use modes outside of the subset.  If not
                  present, all operating modes in the set 0 to 3 are
                  allowed for the session.

  channels:       The number of audio channels.  The possible
                  values and their respective channel order
                  is specified in Section 4.1 in [6].  If
                  omitted, it has the default value of 1.

  octet-align:    RTP payload format; permissible values are 0 and
                  1.  If 1, octet-aligned payload format SHALL be
                  used.  If 0 or if not present, header-free payload
                  format is employed (default).

  maxptime:       See RFC 3267 [4]

Top      ToC       Page 26 
  interleaving:   Indicates that frame-block level
                  interleaving SHALL be used for the session.
                  Its value defines the maximum number of
                  frame-blocks allowed in an interleaving
                  group (see Section 6.3.1).  If this
                  parameter is not present, interleaving
                  SHALL NOT be used.  The presence of this
                  parameter also implies automatically that
                  octet-aligned operation SHALL be used.

  ptime:          See RFC2327 [5].  It SHALL be at least one
                  frame size for VMR-WB.

  dtx:            Permissible values are 0 and 1.  The default
                  is 0 (i.e., No DTX) where VMR-WB normally
                  operates as a continuous variable-rate
                  codec.  If dtx=1, the VMR-WB codec will
                  operate in discontinuous transmission mode
                  where silence descriptor (SID) frames are
                  sent by the VMR-WB encoder during silence
                  intervals with an adjustable update
                  frequency.  The selection of the SID update-rate
                  depends on the implementation and
                  other network considerations that are
                  beyond the scope of this specification.

   Encoding considerations:

          This type is only defined for transfer of VMR-WB-encoded data
          via RTP (RFC 3550) using the payload formats specified in
          Section 6 of RFC 4348.

   Security considerations:

          See Section 8 of RFC 4348.

   Public specification:

          The VMR-WB speech codec is specified in
          3GPP2 specifications C.S0052-0 version 1.0.
          Transfer methods are specified in RFC 4348.

   Additional information:

   Person & email address to contact for further information:

          Sassan Ahmadi, Ph.D.

Top      ToC       Page 27 
   Intended usage: COMMON.

     It is expected that many VoIP, multimedia messaging and
     streaming applications (as well as mobile applications)
     will use this type.

   Author/Change controller:

     IETF Audio/Video Transport working group delegated from the IESG

9.2.  Mapping MIME Parameters into SDP

   The information carried in the MIME media type specification has a
   specific mapping to fields in the Session Description Protocol (SDP)
   [5], which is commonly used to describe RTP sessions.  When SDP is
   used to specify sessions employing the VMR-WB codec, the mapping is
   as follows:

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

      - The media subtype (payload format name) goes in SDP "a=rtpmap"
        as the encoding name.  The RTP clock rate in "a=rtpmap" MUST be
        16000 for VMR-WB.

      - The parameter "channels" (number of channels) MUST be either
        explicitly set to N or omitted, implying a default value of 1.
        The values of N that are allowed is specified in Section 4.1 in
        [6].  The parameter "channels", if present, is specified
        subsequent to the MIME subtype and RTP clock rate as an encoding
        parameter in the "a=rtpmap" attribute.

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

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

   Some examples of SDP session descriptions utilizing VMR-WB encodings

   Example of usage of VMR-WB in a possible VoIP scenario (wideband

      m=audio 49120 RTP/AVP 98
      a=rtpmap:98 VMR-WB/16000
      a=fmtp:98 octet-align=1

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   Example of usage of VMR-WB in a possible streaming scenario (two
   channel stereo):

      m=audio 49120 RTP/AVP 99
      a=rtpmap:99 VMR-WB/16000/2
      a=fmtp:99 octet-align=1; interleaving=30

9.3.  Offer-Answer Model Considerations

   To achieve good interoperability for the VMR-WB RTP payload in an
   Offer-Answer negotiation usage in SDP [13], the following
   considerations are made:

   - The rate, channel, and payload configuration parameters (octet-
     align and interleaving) SHALL be used symmetrically, i.e., offer
     and answer must use the same values.  The maximum size of the
     interleaving buffer is, however, declarative, and each agent
     specifies the value it supports to receive for recvonly and
     sendrecv streams.  For sendonly streams, the value indicates what
     the agent desires to use.

   - To maintain interoperability among all implementations of VMR-WB
     that may or may not support all the codec's modes of operation, the
     operational modes that are supported by an implementation MAY be
     identified at session initiation.  The mode-set parameter is
     declarative, and only operating modes that have been indicated to
     be supported by both ends SHALL be used.  If the answerer is not
     supporting any of the operating modes provided in the offer, the
     complete payload type declaration SHOULD be rejected by removing it
     from the answer.

   - The remaining parameters are all declarative; i.e., for sendonly
     streams they provide parameters that the agent desires to use,
     while for recvonly and sendrecv streams they declare the parameters
     that it accepts to receive.  The dtx parameter is used to indicate
     DTX support and capability, while the media sender is only
     RECOMMENDED to send using the DTX in these cases.  If DTX is not
     supported by the media sender, it will send media without DTX; this
     will not affect interoperability only the resource consumption.

   - Both header-free and octet-aligned payload format configurations
     MAY be offered by a VMR-WB enabled terminal.  However, for an
     interoperable interconnection with AMR-WB, only octet-aligned

   - The parameters "maxptime" and "ptime" should in most cases not
     affect the interoperability; however, the setting of the parameters
     can affect the performance of the application.

Top      ToC       Page 29 
   - To maintain interoperability with AMR-WB in cases where negotiation
     is possible using the VMR-WB interoperable mode, a VMR-WB-enabled
     terminal SHOULD also declare itself capable of AMR-WB with limited
     mode set (i.e., only AMR-WB codec modes 0, 1, and 2 are allowed)
     and of octet-align mode of operation.


                m=audio 49120 RTP/AVP 98 99
                a=rtpmap:98 VMR-WB/16000
                a=rtpmap:99 AMR-WB/16000
                a=fmtp:99 octet-align=1; mode-set=0,1,2

   An example of offer-answer exchange for the VoIP scenario described
   in Section 5.3 is as follows:

       CDMA2000 terminal -> WCDMA terminal Offer:
                m=audio 49120 RTP/AVP 98 97
                a=rtpmap:98 VMR-WB/16000
                a=fmtp:98 octet-align=1
                a=rtpmap:97 AMR-WB/16000
                a=fmtp:97 mode-set=0,1,2; octet-align=1

       WCDMA terminal -> CDMA2000 terminal Answer:
                m=audio 49120 RTP/AVP 97
                a=rtpmap:97 AMR-WB/16000
                a=fmtp:97 mode-set=0,1,2; octet-align=1;

   For declarative use of SDP such as in SAP [14] and RTSP [15], all
   parameters are declarative and provide the parameters that SHALL be
   used when receiving and/or sending the configured stream.

10.  IANA Considerations

   The IANA has registered one new MIME subtype (audio/VMR-WB); see
   Section 9.

11.  Acknowledgements

   The author would like to thank Redwan Salami of VoiceAge Corporation,
   Ari Lakaniemi of Nokia Inc., and IETF/AVT chairs Colin Perkins and
   Magnus Westerlund for their technical comments to improve this

   Also, the author would like to acknowledge that some parts of RFC
   3267 [4] and RFC 3558 [11] have been used in this document.

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12.  References

12.1.  Normative References

   [1]  3GPP2 C.S0052-0 v1.0 "Source-Controlled Variable-Rate Multimode
        Wideband Speech Codec (VMR-WB) Service Option 62 for Spread
        Spectrum Systems", 3GPP2 Technical Specification, July 2004.

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

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

   [4]  Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie, "Real-
        Time Transport Protocol (RTP) Payload Format and File Storage
        Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
        Wideband (AMR-WB) Audio Codecs", RFC 3267, June 2002.

   [5]  Handley, M. and V. Jacobson, "SDP: Session Description
        Protocol", RFC 2327, April 1998.

   [6]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
        Conferences with Minimal Control", STD 65, RFC 3551, July 2003.

12.2.  Informative References

   [7]  3GPP2 C.S0050-A v1.0 "3GPP2 File Formats for Multimedia
        Services", 3GPP2 Technical Specification, September 2005.

   [8]  Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for
        Generic Forward Error Correction", RFC 2733, December 1999.

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

   [10] 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.

   [11] Li, A., "RTP Payload Format for Enhanced Variable Rate Codecs
        (EVRC) and Selectable Mode Vocoders (SMV)", RFC 3558, July 2003.

   [12] 3GPP TS 26.193 "AMR Wideband Speech Codec; Source Controlled
        Rate operation", version 5.0.0 (2001-03), 3rd Generation
        Partnership Project (3GPP).

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

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

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

   Any 3GPP2 document can be downloaded from the 3GPP2 web server,
   "", see specifications.

Author's Address

   Dr. Sassan Ahmadi

Top      ToC       Page 32 
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