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Versions: 00 01 02 03 04 05 06 07 08                                    
Internet Draft                                                Adam H. Li
draft-ietf-avt-evrc-08.txt                                          UCLA
October 10, 2001                                                  Editor
Expires: April 10, 2002


              An RTP Payload Format for EVRC Speech


STATUS OF THIS MEMO

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC 2026.

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

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time. It is inappropriate to use Internet- Drafts as reference
   material or to cite them other than as work in progress.

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html.


ABSTRACT

   This document describes the RTP payload format for Enhanced Variable
   Rate Codec (EVRC) Speech. The packet format supports various formats
   for different application scenarios. A bundled/interleaved format is
   included to reduce the effect of packet loss on Speech quality. A
   non-bundled format is also supported for conversational applications.


Table of Contents

   1. Introduction ................................................... 2
   2. Background ..................................................... 2
   3. RTP/EVRC Packet Format ......................................... 3
   3.1. Type 1 RTP/EVRC Packet Format ................................ 3
   3.2. Type 2 RTP/EVRC Packet Format ................................ 4
   3.3. Detection Between the Type 1 and Type 2 Packets .............. 4
   4. Packet Table of Content Entries and Codec Data Frame Format .... 5
   4.1. Packet Table of Content entries .............................. 5
   4.2. The Codec Data Frame ......................................... 6



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   5. Interleaving Codec Data Frames in Type 1 Packets ............... 7
   5.1. Finding Interleave Group Boundaries .......................... 8
   5.2. Reconstructing Interleaved Speech ............................ 8
   5.3. Receiving Invalid Interleaving Values ........................ 9
   5.4. Additional Receiver Responsibilities ......................... 9
   6. Bundling Codec Data Frames in Type 1 Packets ................... 9
   7. Handling Lost RTP Packets ..................................... 10
   8. Implementation Issues ......................................... 10
   8.1. Interleaving Length ......................................... 10
   8.2. Signaling of Reduce Rate .................................... 11
   9. IANA Considerations ........................................... 11
   9.1 Storage Mode ................................................. 11
   9.2 EVRC MIME Registration ....................................... 12
   10. Mapping to SDP Parameters .................................... 13
   11. Security Considerations ...................................... 13
   12. Acknowledgements ............................................. 14
   13. References ................................................... 14
   14. Authors√č Address ............................................. 14



1. Introduction

   This document describes how compressed EVRC speech as produced by the
   EVRC codec [1] may be formatted for use as an RTP payload type.
   Methods are provided to packetize the codec data frames into RTP
   packets, in interleaved/bundled and zero-header formats. The sender
   may choose among various formats the best solutions for different
   application scenarios based on the network condition, bandwidth
   restriction, delay requirements, and packet-loss tolerance.

   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 [2].


2. Background

   The Electronic Industries Association (EIA) & Telecommunications
   Industry Association (TIA) standard IS-127 [1] defines a speech
   compression algorithm for use in cdma2000 applications. IS-127, or
   EVRC, is the emerging speech codec standard for cdma2000.

   The EVRC codec [1] compresses each 20 milliseconds of 8000 Hz, 16-
   bit sampled input speech into one of three different size output
   frames: Rate 1 (171 bits), Rate 1/2 (80 bits), or Rate 1/8 (16 bits).
   The codec chooses the output frame rate based on analysis of the
   input speech and the current operating mode (either normal or one of
   several reduced rates). For typical speech patterns, this results in
   an average output of 4.2 kilobits/second for normal mode and lower
   for reduced rate modes.



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3. RTP/EVRC Packet Format

   The RTP timestamp is in 1/8000 of a second units. The RTP payload
   data for the EVRC codec MUST be transmitted in packets of one of the
   following two types.

3.1 Type 1 RTP/EVRC Packet Format

   This format is intended for the situation where the sender and the
   receiver use interleaving/bundling to send one or more codec data
   frames per packet. The RTP packet for this format is as follows:

    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]                           |
   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   | RR| LLL | NNN |                                               |
   +-+-+-+-+-+-+-+-+     one or more ToC entries     +-------------+
   |                                                 |             |
   +-------------------------------------------------+             |
   |                                                               |
   |                  one or more codec data frames                |
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The RTP header has the expected values as described in [3]. The M bit
   should be set as specified in the applicable RTP profile, for
   example, RFC 1890 [4]. Note that RFC 1890 [4] specifies that if the
   sender does not suppress silence (i.e., sends a frame on every 20
   millisecond interval), the M bit will always be zero. When 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 assignment of an RTP payload type for this new packet format is
   outside the scope of this document, and will not be specified here.
   It is expected that the RTP profile for a particular class of
   applications will assign a payload type for this encoding, or if that
   is not done, then a payload type in the dynamic range shall be chosen
   by the sender.

   The first octet of a Type 1 format packet is the Interleave Byte.
   The bits within the Interleave Byte are specified as follows:

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

   Interleave Length (LLL): 3 bits
      Indicates the length of interleave. MUST have a value between 0
      and 7 inclusive (where a value 0 indicates bundling, a special
      case of interleaving). See Section 5 and Section 6 for more
      detailed discussion.


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

   The Table of Content field (ToC) contains the index(es) for the codec
   data frame(s) in the packet. There is one entry for each codec data
   frame. The detailed formats of the ToC field and codec data frame are
   specified in Section 4.

   More than one codec data frames MAY be included in a single Type 1
   RTP/EVRC packet by a sender. Multiple data frames may be included
   within a Type 1 packet with interleaving/bundling format as described
   in Section 5 and Section 6.

   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 a Type 1 RTP/EVRC packet is
   to examine the ToC fields of the packet.

3.2 Type 2 RTP/EVRC Packet Format

   The Type 2 RTP/EVRC Packet Format is designed for maximum efficiency
   and low latency in transmission of the EVRC codec data. Exactly one
   codec data frame MUST be sent in each Type 2 RTP/EVRC packet. There
   MUST NOT be ToC field preceding the codec data. The EVRC codec rate
   for the data frame can be found out at the receiver from the length
   of the codec data frame, since there is only one codec data frame in
   each Type 2 packet. The Reduce Rate Signal (See Section 4.1) can not
   be send in-bound with the Type 2 packets because of the lacking of
   the ToC field in this type.

   Use of the RTP header fields for Type 2 RTP/EVRC Packet Format is the
   same as described in Section 3.1 for Type 1 RTP/EVRC Packet Format.
   The detailed formats of the codec data frame are specified in Section
   4.

    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 codec data frame            +-+-+-+-+-+-+-+-+
   |                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

3.3 Detection Between the Type 1 and Type 2 Packets

   All receivers MUST be able to process both types of packets. The
   sender MAY choose to use one or both types of packets.



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   The packets of the two types can be distinguished by using different
   payload type value for the two packet types at the sender and
   checking the payload type field in the RTP header at the receiver.
   The association of payload type number with the packet type is done
   out-of-band, for example by SDP during the setup of a session.


4. Packet Table of Content Entries and Codec Data Frame Format

4.1 Packet Table of Content entries

   For each of the codec data frames in Type 1 packets, there is a
   corresponding Table of Content (ToC) entry. The ToC entry includes
   flags that indicates if there are more entries following the current
   one, if rate reduction on the reverse direction is desired, and the
   rate of the corresponding codec frame. Type 2 packets MUST NOT have
   the ToC field, and there is always only one codec data frame in each
   Type 2 packet.

   Each ToC entry is one octet in size. The format of the octet is
   indicated below:

       0 1 2 3 4 5 6 7
      +-+-+-+-+-+-+-+-+
      |F|D|  frm type |
      +-+-+-+-+-+-+-+-+

   Further Entry Indication (F): 1 bit
      Indicates if there are more ToC entries following the current ToC
      entry. F = 1 indicates the next octet is another ToC entry. F = 0
      indicates that the current entry is the final ToC entry.

   Reduce Rate (D): 1 bit
      Setting the 'D' bit indicates that the sender is requesting a
      reduced codec rate for the reverse direction. When the 'D' bit is
      not set, the sender is requesting that the codec resume normal
      operation. In the case of packet loss, the codec SHOULD continue
      to operate in the mode indicated by the last codec frame
      received. Receivers are NOT REQUIRED to respond to the Reduce
      Rate signal. (See more discussion in Section 8.2).

   Frame Type: 6 bits
      The frame type values and size of the associated codec data frame
      are described in the table below:

      Value   Rate      Total codec data frame size (in octets)
      ---------------------------------------------------------
        0     Blank      0
        1     1/8        2
        3     1/2       10
        4     1         22
       14     Erasure    0    (SHOULD NOT be transmitted by sender)


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      All values not listed in the above table MUST be considered
      reserved. Receipt of a ToC entry with a reserved value in Frame
      Type MUST be considered invalid data.

4.2 The Codec Data Frame

   The output of the EVRC codec MUST be converted into codec data frames
   for inclusion in the RTP payload as specified below.

   The codec output data bits as numbered in the standard [1] are packed
   into octets. The lowest numbered bit (bit 1 for Rate 1, Rate 1/2 and
   Rate 1/8) is placed in the most significant bit (internet bit 0) of
   octet 1 of the codec data frame, the second lowest bit is placed in
   the second most significant bit of the first octet, the third lowest
   in the third most significant bit of the first octet, and so on.
   This continues until all of the bits have been placed in the codec
   data frame. The remaining unused bits of the last octet of the codec
   data frame MUST be set to zero. Note that this is only applicable to
   Rate 1 frames (171 bits) as the Rate 1/2 (80 bits) and Rate 1/8
   frames (16 bits) fit exactly into a whole number of octets.

   Following is a detailed listing showing a Rate 1 EVRC codec output
   frame converted into a codec data frame:

   The codec data frame for a EVRC codec Rate 1 frame is 22-byte long.
   Bits 1 through 171 from the EVRC codec Rate 1 frame are placed as
   indicated, with bits marked with "Z" set to zero. EVRC codec Rate 1/8
   and Rate 1/2 frames are converted similarly, but do not require zero
   padding because they align on octet boundaries.

                    Rate 1 codec data frame (bytes 0 - 3)

    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|0|
   |0|0|0|0|0|0|0|0|0|1|1|1|1|1|1|1|1|1|1|2|2|2|2|2|2|2|2|2|2|3|3|3|
   |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|2|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                    Rate 1 codec data frame (bytes 19 - 21)

    1           1                   1                   1
    4           5                   6                   7
    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 2 3 4 5
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1|1| | | | | |
   |4|4|4|4|4|5|5|5|5|5|5|5|5|5|5|6|6|6|6|6|6|6|6|6|6|7|7|Z|Z|Z|Z|Z|
   |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| | | | | |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+




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5. Interleaving Codec Data Frames in Type 1 Packets

   As indicated in Section 3.1, more than one codec data frame MAY be
   included in a single Type 1 packet by a sender. This is accomplished
   by interleaving/bundling. Interleaving/bundling of codec data frames
   is signaled by setting the LLL value in the Interleaving Byte to a
   value between 0 and 7 inclusive.

   The special case with the LLL value set to 0 is a reduced and
   simplified case of interleaving. This is sometimes called bundling,
   because multiple consecutive codec data frames are included in one
   RTP packet in this case. The discussions on general interleaving
   apply to the bundling case with reduced complexity. The bundling case
   is discussed in detail in Section 6.

   Senders MAY support interleaving/bundling. All receivers MUST support
   interleaving/bundling.

   Given a time-ordered sequence of output frames from the EVRC codec
   numbered 0..n, a bundling value B, and an interleave length 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.

   Receivers MAY signal the maximum number of codec data frames (i.e.,
   the maximum acceptable bundling value B) they can handle in a single
   RTP packet using the OPTIONAL maxptime RTP mode parameter identified
   in Section 9.2.


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   Receivers MAY signal the maximum interleave length (i.e., the maximum
   acceptable LLL value in the Interleaving Byte) they will accept using
   the OPTIONAL maxinterleave RTP mode parameter identified in Section
   9.2.

   Additionally, senders have the following restrictions:

   o  MUST NOT bundle more codec data frames in a single RTP packet than
      indicated by maxptime (see Section 9.2) if it is signaled.

   o  SHOULD NOT bundle more codec data frames in a single RTP packet
      than will fit in the MTU of the underlying network. For the
      purpose of computing the maximum bundling value, all codec data
      frames MUST be assumed to have the Rate 1 size.

   o  Once beginning a session with a given maximum interleaving value
      set by maxinterleave in Section 9.2, MUST NOT increase the
      interleaving value (LLL) exceeding the maximum interleaving value
      that is signaled.

   o  MAY change the interleaving value only between interleave groups.

5.1 Finding Interleave Group Boundaries

   Given an RTP packet with sequence number S, interleave length (field
   LLL) L, interleave index value (field NNN) N, and bundling value B,
   the interleave group consists of RTP packets with sequence numbers
   from S-N to S-N+L inclusive. (The sequence numbers used here are for
   illustrative purposes. When wrapping around happens, the sequence
   numbers need to be adjusted accordingly). 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.

5.2 Reconstructing Interleaved Speech

   Given an RTP sequence number ordered set of RTP packets in an
   interleave group numbered 0..L, where L is the interleave length and
   B is the bundling value, and codec data frames within each RTP packet

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   that are numbered in order from first to last with the numbers 1..B,
   the original, time-ordered sequence of output frames from the EVRC
   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


   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

5.3 Receiving Invalid Interleaving 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 7.

5.4 Additional Receiver Responsibilities

   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 7, an erasure
   frame will be sent to the receiving EVRC 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 was not available the
   first time it was needed to reconstruct the interleaved speech, the
   receiver SHOULD NOT assume it is not available when it is
   subsequently needed for interleaved speech reconstruction.


6. Bundling Codec Data Frames in Type 1 Packets

   As discussed in Section 5, the bundling of codec data frames is a
   special reduced case of interleaving with LLL value in the Interleave
   Byte set to 0.

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   Bundling codec data frames indicates multiple data frames are
   included consecutively in a packet, because the interleaving length
   (LLL) is 0. The interleaving group is thus reduced to a single RTP
   packet, and the reconstruction of the code data frames from RTP
   packets becomes a much simpler process.

   Furthermore, the additional restriction on the senders are reduced
   to:

   o  MUST NOT bundle more codec data frames in a single RTP packet than
      indicated by maxptime (see Section 9.2) if it is signaled.

   o  SHOULD NOT bundle more codec data frames in a single RTP packet
      than will fit in the MTU of the underlying network. For the
      purpose of computing the maximum bundling value, all codec data
      frames MUST be assumed to have the Rate 1 size.


7. Handling Lost RTP Packets

   The EVRC codec supports the notion of erasure frames. These are
   frames that for whatever reason are not available. When
   reconstructing or playing back speech, erasure frames MUST be fed to
   the receiving EVRC 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 interleaving length/bundling value may vary, the timestamp
   clock is the only reliable way to calculate exactly how many codec
   data frames are missing when a packet is dropped.

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


8. Implementation Issues

8.1 Interleaving Length

   The EVRC codec interpolates the missing speech 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 speech quality when packet loss may occur.

   On the other hand, interleaving can greatly increase the end-to-end
   delay. Where an interactive session is desired, either Type 1 with
   interleaving length 0 or Type 2 RTP payload types are RECOMMENDED.


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   When end-to-end delay is not a concern, an interleaving length (field
   LLL) of 4 or 5 is RECOMMENDED.

   The parameters maxptime and maxinterleave at the initial setup of the
   session guarantees that the receiver can allocate a well-known amount
   of buffer space at the beginning of the session that will be
   sufficient for all future reception in that session. Less buffer
   space may be required at some point in the future if the sender
   decreases the bundling value or interleaving length, 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).

8.2 Signaling of Reduce Rate

   The Reduce Rate signal requests a reduction of the codec rate on the
   reverse direction. It is NOT REQUIRED that all implementations react
   to the Reduce Rate signal. If an implementation does react to the
   Reduce Rate signal, it MUST be able to process/react to the D bit in
   Type 1 packets. The Reduce Rate signal SHOULD only be used in one-to-
   one sessions. In multiparty sessions, all the received Reduce Rate
   signals MUST be ignored.

   In addition, the Reduce Rate signal MAY also be sent through non-RTP
   means, which is out of the scope of this specification.


9. IANA Considerations

   One new MIME sub-type as described in this section is to be
   registered.

   The MIME-name for the EVRC codec is allocated from the IETF tree
   since EVRC is expected to be a widely used codec for Voice-over-IP
   applications.

   The RTP mode has been described in the previous sections.

9.1 Storage Mode

   The storage mode is used for storing speech frames, e.g. as a file or
   e-mail attachment.

   The file begins with a magic number to identify that it is an EVRC
   file. The magic number for EVRC corresponds to the ASCII character
   string "#!EVRC\n", i.e., "0x23 0x21 0x45 0x56 0x52 0x43 0x0A" in
   network byte order.

   The codec data frames are stored in consecutive order, with a single
   TOC entry field (1 octet) prefixing each codec data frame. The F bit
   and the D bit in the ToC entry field SHOULD be set to 0 and MUST be
   ignored when processing speech data from storage mode.


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   Speech frames lost in transmission and non-received frames MUST be
   stored as erasure frames (frame type 14, see definition in Section
   4.1) to maintain synchronization with the original media.

9.2 EVRC MIME Registration

   Media Type Name:     audio

   Media Subtype Name:  EVRC

   Required Parameters:

      ptype:    Indicates the Type of the RTP/EVRC packets. The valid
         values are 1 (Type 1) or 2 (Type 2).

   Optional parameters for RTP mode:

      ptime:    Defined as usual for RTP audio [5].

      maxptime: The maximum amount of media which can be encapsulated
         in each packet, expressed as time in milliseconds. The time
         SHALL be calculated as the sum of the time the media present
         in the packet represents. The time SHOULD be a multiple of the
         duration of a single codec data frame (20 msec). If not
         signaled, the default maxptime value SHALL be 200
         milliseconds.

      maxinterleave: Maximum number for interleaving length (field LLL
         in the Interleaving Byte). The interleaving lengths used in
         the entire session MUST NOT exceed this maximum value. If not
         signaled, the maxinterleave length SHALL be 5.

   Optional parameters for storage mode: none

   Encoding considerations for RTP mode: see Section 5 and Section 6 of
      RFC xxxx.

   Encoding considerations for storage mode: see Section 9.1 of RFC
      xxxx.

   Security considerations: see Section 11 "Security Considerations" of
      RFC xxxx.

   Public specification: RFC xxxx.

   Additional information for storage mode:
      Magic number: #!EVRC\n
      File extensions: evc, EVC
      Macintosh file type code: none
      Object identifier or OID: none

   Intended usage: COMMON. It is expected that many VoIP applications
      (as well as mobile applications) will use this type.

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   Person & email address to contact for further information:
      adamli@icsl.ucla.edu

   Author/Change controller:
      adamli@icsl.ucla.edu
      IETF Audio/Video transport working group


10. Mapping to SDP Parameters

   Please note that this section applies to the RTP mode only.

   Parameters are mapped to SDP [5] as usual.
   Example usage in SDP:
     m = audio 49120 RTP/AVP 97
     a = rtpmap:97 EVRC
     a = fmtp:97 ptype=1; maxinterleave=2
     a = maxptime:80


11. Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the security considerations discussed in the RTP
   specification [3], 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.

   A potential denial-of-service threat exists for data encoding 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
   become 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 [6] and in multicast routing protocols to
   allow a receiver to select which sources are allowed to reach it.

   Interleaving MAY affect encryption. Depending on the used encryption
   scheme there MAY be restrictions on for example the time when keys
   can be changed.




Adam H. Li                                                     [Page 13]


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12. Acknowledgements

   The editor thanks the following authors for contributions to this
   document:    J. D. Villasenor, D.S. Park, J.H. Park, K. Miller, S. C.
   Greer, D. Leon, N. Leung, K. J. McKay, M. Lioy, T. Hiller, P. J.
   McCann, M. D. Turner, A. Rajkumar, D. Gal, M. Westerlund, L.-E.
   Jonsson, G. Sherwood, and T. Zeng.


13. References

   [1]  TIA/EIA/IS-127, "Enhanced Variable Rate Codec, Speech Service
        Option 3 for Wideband Spread Spectrum Digital Systems", January
        1997.

   [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", RFC
        1889, January 1996.

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

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

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


14. Authors' Address

   Adam H. Li
   Image Communication Lab
   Electrical Engineering Department
   University of California
   Los Angeles, CA 90095
   USA
   Phone: +1 310 825 5178
   Email: adamli@icsl.ucla.edu

   John D. Villasenor
   Image Communication Lab
   Electrical Engineering Department
   University of California
   Los Angeles, CA 90095
   USA
   Phone: +1 310 825 0228
   Email: villa@icsl.ucla.edu



Adam H. Li                                                     [Page 14]


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   Dong-Seek Park
   Samsung Electronics
   Suwon, Kyungki  442-742
   Korea
   Phone: +82 31 200 3674
   Email: dspark@samsung.com

   Jeong-Hoon Park
   Samsung Electronics
   Suwon, Kyungki  442-742
   Korea
   Phone: +82 31 200 3747
   Email: dspark@samsung.com

   Keith Miller
   Nokia
   6000 Connection Drive
   Irving, Texas 75039
   USA
   Phone: +1 972 894 4296
   Email: keith.miller@nokia.com

   S. Craig Greer
   Nokia
   6000 Connection Drive
   Irving, Texas 75039
   USA
   Phone: +1 972 894 4867
   Email: craig.greer@nokia.com

   David Leon
   Nokia
   6000 Connection Drive
   Irving, Texas 75039
   USA
   Phone: +1 972 374 1860
   Email: david.leon@nokia.com

   Marcello Lioy
   QUALCOMM, Incorporated
   5775 Morehouse Drive
   San Diego, CA 92121
   USA
   Phone: +1 858 651 8220
   Email: mlioy@qualcomm.com

   Nikolai Leung
   QUALCOMM, Incorporated
   7710 Takoma Ave.
   Takoma Park, MD 20912
   USA
   Phone: +1 703 346 8351
   Email: nleung@qualcomm.com

Adam H. Li                                                     [Page 15]


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   Kyle J. McKay
   QUALCOMM, Incorporated
   5775 Morehouse Drive
   San Diego, CA 92121-1714
   USA
   Phone: +1 858 587 1121
   EMail: kylem@qualcomm.com

   Tom Hiller
   Lucent Technologies
   263 Shuman Drive, Room 2F-218
   Naperville, IL 60137
   USA
   Phone: +1 630 979 7673
   Email: tom.hiller@lucent.com

   Peter J. McCann
   Lucent Technologies
   263 Shuman Drive, Room 2Z-305
   Naperville, IL 60137
   USA
   Phone: +1 630 713 9359
   Email: mccap@lucent.com

   Michael D. Turner
   Lucent Technologies
   67 Whippany Rd, Room 2A-203
   Whippany, NJ 07981
   USA
   Phone: +1 973 386 3579
   Email: mdturner@lucent.com

   Ajay Rajkumar
   Lucent Technologies
   67 Whippany Rd, Room 1A-235
   Whippany, NJ 07981
   USA
   Phone: +1 973 386 5249
   Email: ajayrajkumar@lucent.com

   Dan Gal
   Lucent Technologies
   67 Whippany Rd
   Whippany, NJ 07981
   USA
   Phone: +1 973 428 7734
   Email: dgal@lucent.com






Adam H. Li                                                     [Page 16]


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   Magnus Westerlund
   Ericsson Radio Systems AB
   Torshamnsgatan 23
   SE-164 80 Stockholm
   Sweden
   Phone: +46 8 4048287
   Email: magnus.westerlund@ericsson.com

   Lars-Erik Jonsson
   Ericsson Erisoft AB
   Box 920
   SE-971 28 Luleà
   Sweden
   Phone: +46 920 20 21 07
   Email: lars-erik.jonsson@ericsson.com

   Greg Sherwood
   PacketVideo Corporation
   4820 Eastgate Mall
   San Diego, CA 92121
   USA
   Email: sherwood@packetvideo.com

   Thomas Zeng
   PacketVideo Corporation
   4820 Eastgate Mall
   San Diego, CA 92121
   USA
   Email: zeng@packetvideo.com

























Adam H. Li                                                     [Page 17]