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Versions: (draft-legrand-rtp-isac) 00 01 02 03 04

Network Working Group                                        T. le Grand
Internet-Draft                                                    Google
Intended status: Standards Track                                P. Jones
Expires: August 12, 2013                                        P. Huart
                                                           Cisco Systems
                                                           T. Shabestary
                                                      H. Alvestrand, Ed.
                                                        February 8, 2013

                 RTP Payload Format for the iSAC Codec


   iSAC is a proprietary wideband speech and audio codec developed by
   Global IP Solutions (now part of Google), suitable for use in Voice
   over IP applications.  This document describes the payload format for
   iSAC generated bit streams within a Real-Time Protocol (RTP) packet.
   Also included here are the necessary details for the use of iSAC with
   the Session Description Protocol (SDP).

Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [RFC2119].

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   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."

   This Internet-Draft will expire on August 12, 2013.

Copyright Notice

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   Copyright (c) 2013 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  iSAC Codec Description . . . . . . . . . . . . . . . . . . . .  3
   3.  RTP Payload Format . . . . . . . . . . . . . . . . . . . . . .  4
     3.1.  Payload Header . . . . . . . . . . . . . . . . . . . . . .  5
     3.2.  iSAC Wideband Payload Format . . . . . . . . . . . . . . .  6
       3.2.1.  Encoded Speech Data  . . . . . . . . . . . . . . . . .  6
     3.3.  iSAC Superwideband Payload Format  . . . . . . . . . . . .  7
       3.3.1.  Encoded Upper-band Speech Data . . . . . . . . . . . .  8
     3.4.  Padding  . . . . . . . . . . . . . . . . . . . . . . . . .  8
     3.5.  Multiple iSAC frames in an RTP packet  . . . . . . . . . .  9
   4.  Congestion Control . . . . . . . . . . . . . . . . . . . . . .  9
   5.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 10
   6.  Mapping to SDP Parameters  . . . . . . . . . . . . . . . . . . 12
     6.1.  Example Initial Target Bit Rate  . . . . . . . . . . . . . 12
     6.2.  Example Max Bit Rate . . . . . . . . . . . . . . . . . . . 13
     6.3.  Example with both WB and SWB offered . . . . . . . . . . . 13
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 13
   8.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 14
   9.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 14
     9.1.  Normative References . . . . . . . . . . . . . . . . . . . 14
     9.2.  Informative References . . . . . . . . . . . . . . . . . . 14
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 15

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1.  Introduction

   This document gives a general description of the iSAC wideband speech
   codec and specifies the iSAC payload format for usage in RTP packets.
   Also included here are the necessary details for the use of iSAC with
   the Session Description Protocol (SDP).

2.  iSAC Codec Description

   The iSAC codec is an adaptive wideband/superwideband speech and audio
   codec that operates with short delay, making it suitable for high
   quality real time communication.  It is specially designed to deliver
   wideband speech quality in both low and medium bit rate applications.
   It also handles non-speech audio well, such as music and background
   noise.  A freely available reference implementation exists [iSAC].

   The iSAC codec compresses speech frames of 16 kHz, 16-bit sampled
   input speech, each frame containing 30 or 60 ms of speech.  It also
   has a superwideband mode which allows a 32 kHz sampling rate.  In
   super-wideband mode the input signal is split into wideband (0-8 kHz)
   and upper (8-16 kHz) signal.  Each sub-band is encoded independently,
   and their associated payloads concatenated, c.f.  Figure 2, to
   construct the overall iSAC super-wideband RTP payload.  Note that the
   same encoder/decoder is used for the wideband part for both wideband
   and super-wideband modes.

   The codec runs in one of two different modes called channel-adaptive
   mode and channel-independent mode.  In both modes iSAC is aiming at a
   target bit rate, which is neither the average nor the maximum bit
   rate that will be reach by iSAC, but corresponds to the average bit
   rate during peaks in speech activity.  The bit rate will sometimes
   exceed the target bit rate, but most of the time will be below.  The
   average bit rate obtained is on average about a factor of 1.2 times
   lower than the target bit rate on continuous speech, and will be
   lower on speech with pauses.

   In channel-adaptive mode the target bit rate is adapted to give a bit
   rate corresponding to the available bandwidth on the channel.  Even
   at dial-up modem data rates (including IP, UDP, and RTP overhead)
   iSAC delivers high quality by automatically adjusting transmission
   rates to give the best possible listening experience over the
   available bandwidth.

   In channel-independent mode a target bit rate has to be provided to
   iSAC prior to encoding; the target bit rate can be changed over the
   time of the call.

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   After encoding the speech signal the iSAC coder uses lossless coding
   to further reduce the size of each packet, and hence the total bit
   rate used.

   The adaptation and the lossless coding described above both result in
   a variation of packet size, depending both of the nature of speech
   and the available bandwidth.  Therefore, the iSAC codec, in wideband
   mode, operates at transmission rates from about 10 kbps to about 32
   kbps.  In super-wideband mode, the transmission rate is in the range
   of 10 kbps to 56 kbps.  If operating in super-wideband mode, the iSAC
   codec automatically adjusts the effective encoded audio bandwidth for
   the best experience.

   Bit Rate  | 10 - 32  |  32 - 38   | 38 - 45 | 45 - 50    | 50 - 56
    [kbps]   |          |            |         |            |
   Effective |          |  0 - 8     | 0 - 12  | 0 - 12     | 0 - 16
   Bandwidth | 0 - 8 kHz| operating  |         | operating  |
    [kHz]    |          | at 32 kbps |         | at 45 kbps |

   The main characteristics can be summarized as follows:

   o  Wideband or superwideband, 16 kHz or 32 kHz respectively, speech
      and audio codec

   o  Variable bit rate, which depends on the input signal

   o  Adaptive rate with two modes: channel-adaptive or channel-
      independent mode

   o  Bit rate range from around 10 kbps to 32 kbps when operating on
      wideband input.  For input audio sampled at 32 kHz, the bit rate
      range 10 kbps to 56 kbps.

   o  Operates on 30 or 60 ms of speech for wideband inputs, and only 30
      ms for super-wideband inputs.

   o  In super-wideband mode, depending on the target bit rate, the
      effective bandwidth is adjusted for the optimal experience.

3.  RTP Payload Format

   The iSAC codec in wideband mode uses a sampling rate clock of 16 kHz,
   so the RTP timestamp MUST be in units of 1/16000 of a second.  In
   super-wideband mode, the iSAC codec uses a sampling rate clock of 32
   kHz, so the RTP timestamp MUST be in units of 1/32000 of a second.

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   The RTP payload for iSAC has the format shown in Figure 1.  No
   additional header fields specific to this payload format are
   required.  For RTP based transportation of iSAC encoded audio, the
   standard RTP header [RFC3550] is followed by one payload data block.

   The assignment of an RTP payload type for the format defined in this
   memo is outside the scope of this document.  The RTP profiles in use
   currently mandate binding the payload type dynamically for this
   payload format.

   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                               |
   |                                                               |
   +                    iSAC Payload Block                         +
   |                                                               |
   Figure 1: RTP packet format for iSAC

3.1.  Payload Header

   The payload header holds information for the receiver about the
   available bandwidth, in the form of a Bandwidth Estimation Index
   (BEI), and the length of the speech data in the current payload
   (frame length, FL).  The header has the format defined in Figure 3.
   Note that the size of the header can vary due to the lossless
   encoding described in section 2 and in section 3.1.  Also note that
   the BEI is always estimated and transmitted, even if iSAC runs in
   channel-independent mode.

   | BEI |  FL |
   Figure 3: Payload Header

   o  BEI: Bandwidth Estimation Index.  The bandwidth estimate that the
      sender estimates for a stream originated at the receiver.  It is
      quantized into one out of 24 values.  Valid values are 0 to 23;
      consult source code for details.

   o  FL: The length of the speech data (Frame Length) present in the
      payload, given in number of speech samples.  Valid frame lengths
      are 480 (30 ms) and 960 (60 ms) samples.

   The BEI and FL are encoded together with the data using a lossless
   compressed encoding, which results in a variable number of bits used

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   to represent the fields.

3.2.  iSAC Wideband Payload Format

   The iSAC payload block consists of a payload header and one or two
   encoded 30 ms speech frames.  The iSAC payload is generated in the
   following manner:

   o  Parameters representing one or two 30 ms frames of speech data are
      determined by the encoder.  The parameters are quantized to
      generate encoded data corresponding to the one or two speech
      frames.  The length of the encoded data is variable and depends on
      the signal characteristics and the target bit rate.

   o  The payload header is generated (described in Section 3.1) and
      added before the encoded parameter data for the speech frame(s).

   o  Lossless coding is applied to the complete iSAC payload block,
      including payload header, to generate a compressed payload.  The
      length depends on the length of the data generated to represent
      the speech and the effectiveness of the lossless coding.

   No part of the payload header or the encoded speech data can be
   retrieved without partly or fully decoding the packet.

   The following figure shows an iSAC payload block containing 60 ms of
   encoded speech data.

   |Payload |       30 ms Encoded      |     30 ms Encoded      |
   |Header  |         Speech Data      |       Speech Data      |
   Figure 2: Payload format for iSAC

3.2.1.  Encoded Speech Data

   The iSAC encoded speech data consist of parameters representing one
   or two frames of 30 ms speech.  The length of the speech data is
   signaled in the header (in number of samples), and the length may
   change at any time during a session.  In channel-adaptive mode the
   length is changed to best utilize the available bandwidth, and extra
   padding is added to some packets as a bandwidth probe.

   The iSAC payload is padded to whole octets, and has a variable length
   depending on the input source signal, number of 30 ms speech frames,
   and target bit rate.

   The number of octets used to describe one frame of 30 ms speech

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   typically varies from around 50 to around 120 octets.  For the case
   of 60 ms speech (two 30 ms speech frames), the number of octets
   varies from around 100 to around 240 octets.  The absolute maximum
   allowed payload length is 400 octets.  The sender can choose to limit
   the packet size further when transmitting.  The minimum useful limit
   for the payload length is 100 octets.

   The sensitivity to bit errors is equal for all bits in the payload.

3.3.  iSAC Superwideband Payload Format

   In super-wideband mode, payloads associated with each sub-band
   (wideband 0-8 kHz and upper-band 8-16 kHz) are constructed
   independently and concatenated as depicted in Figure 2.  Note that in
   super-wideband mode only one 30 ms frame is encoded in each payload.

   The receiver will know from negotiation whether wideband or super-
   wideband is sent; it can also verify this for each packet by
   verifying the CRC checksum.

   | Payload +30 ms Encoded wideband|LEN|30 ms Encoded upper-band| CRC |
   | Header   speech data           |   |speech data             |check|
                                        |<--- CRC checked data ->|

   Figure 4: Super-Wideband payload format

   Payloads of wideband and upper-band are encoded independently,
   allowing the encoder to simply concatenate two payloads to construct
   one iSAC super-wideband payload.  The RTP payload of the iSAC super-
   wideband codec starts with the payload of the wideband part, which is
   padded to whole octets, followed by one byte (LEN in Figure 4)
   representing the length of the remaining sequence, payload of the
   upper-band plus 4 bytes for CRC sequence.

   If LEN_UB denotes the length of the upper-band payload, then LEN = 1
   + LEN_UB + 4.  If this value would exceed 255 at encoding, the upper-
   band payload is omitted.

   The CRC check is added to distinguish between upper-band payload and
   random bit-stream padding that can be added for probing available
   network bandwidth.

   At the receive side, a super-wideband payload is first given to the
   wideband decoder.  The wideband decoder decodes as many parameters as
   required to uniquely reproduce the encoded wideband audio.  The next
   byte in the payload should hold the value of LEN.  This provides a

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   sanity check that the decoding process has not failed.  Thereafter,
   the receiver runs a CRC check over the upper-band payload and
   compares the results with the last 4 bytes in the packet.

   If the computed CRC and the last four bytes of the payload don't
   match, the remaining bits are assumed to be added for probing the
   network.  Hence, the upper-band signal is replaced by zeros and
   combined with the wideband signal to generate the super-wideband

   If the two CRCs match, then the upper-band payload is given to the
   upper-band decoder.  Thereby, the output of the upper-band decoder is
   combined with the wide-band decoded audio to generate the super-
   wideband signal.

   It might be that for a given packet, the wideband decoder uses all
   the given payload.  This can be the case when a super-wideband
   encoder is operating at low rates and has adjusted the effective
   bandwidth to wideband.  In this case, the decoder inserts zeros as
   the reconstructed upper-band and combines both bands to reproduce the
   super-wideband signal.

3.3.1.  Encoded Upper-band Speech Data

   The iSAC encoded upper-band speech data consists of parameters
   representing one frame of 30 ms speech.  Depending on the target rate
   the upper-band encoder might choose to only encode the sub-band of 8
   kHz to 12 kHz.

3.4.  Padding

   Padding, which consists of randomly generated bits, may be added at
   the end of the payload in both wideband and superwideband modes.  It
   can be used by the sender for bandwidth probing, and is always
   ignored by the receiver.

   In wideband mode, padding simply follows the payload, preceded by a
   length field.

   | Wideband |LEN|Padding |
   | payload  |   |        |

   Figure 5: Wideband payload format with padding.

   LEN is the length of the padding in bytes + 1: LEN = LEN_PAD + 1

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   In superwideband mode, the format of a packet with padding looks like
   the following.

   | Wideband |LEN|Upper-band   |L2|Padding |CRC  |
   | payload  |   |speech data  |  |        |check|
                  |<-- CRC checked data --->|

   Figure 6: Super-Wideband payload format

   LEN is 1 + LEN_UB + 1 + LEN_PAD + 4, where LEN_UB is the length of
   the upper-band speech data in bytes, and LEN_PAD is the length of the
   padding in bytes.

   L2 is LEN_PAD + 1.

   The CRC check runs over the upper-band speech data, L2 and the

3.5.  Multiple iSAC frames in an RTP packet

   More than one iSAC payload block MUST NOT be included in an RTP
   packet by a sender.

   Further, iSAC payload blocks MUST NOT be split between RTP packets.

4.  Congestion Control

   When ISAC is used in an environment where congestion control is
   useful, there are two properties of importance:

   o  The ISAC format has the ability to pad packets.  This allows a
      sender to probe a channel with more bits per second than is
      strictly needed for the transmission of current data, so that it
      can check for the possibility of sending bigger packets without
      incurring increased packet loss.

   o  The iSAC encoder (in channel-adaptive mode) can continuously tune
      its encoding parameters so as to adapt the encoding to the
      available bandwidth, without introducing switching artifacts into
      the audio stream.

   o  In the case where two parties have one audio channel in each
      direction, they can use the BEI field of the A->B audio flow as a
      feedback channel for the B->A audio flow.

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   Coupled with a feedback channel (which may be of any type), the
   sender can send some packets of larger size than necessary; the
   recipient can then figure out if this increased size led to increased
   packet loss or delay, and can send back information about this to the

   The sender can then change its encoding parameters to produce smaller
   or larger packets; when in wideband mode, it can also switch between
   30-ms and 60-ms mode.

   In the particular case of one audio channel in each direction, both
   using iSAC, iSAC defines the BEI field as a feedback channel.  The
   available bandwidth is continuously estimated at the receiving iSAC;
   the receiver will signal the sender in-band in the iSAC bit stream,
   using the BEI field, what its estimate is.  If the sending iSAC is
   running in channel-adaptive mode, it will adjust its bitrate

   This specification does not specify any particular feedback mechanism
   for any other use case.

   Note: This mechanism is only capable of reducing iSAC traffic to the
   lowest available setting for iSAC.  If there is congestion that makes
   even less bandwidth available, other mechanisms, such as dropping the
   call, will have to be used to escape from the congestion situation.

5.  IANA Considerations

   This RTP payload format is identified using the media type audio/
   isac, which is registered in accordance with [RFC4855] and uses the
   template of [RFC6838].

   Type name:  audio

   Subtype name:  isac

   Required parameters:  None

   Optional parameters:

      *  ibitrate: The parameter indicates the upper bound in bits per
         second of the initial target bit rate (counting only payload
         bits) the device would like to receive. A sender SHOULD set its
         initial target bitrate to a value less than or equal to this
         parameter.  An acceptable value for ibitrate is in the range of

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         20000 to 32000 (bits per second).  In the absence of the
         parameter, the sender can choose any value up to the maximum
         bitrate possible.

      *  maxbitrate: The parameter indicates the maximum bit rate the
         endpoint expects to receive.  The recipient of this parameter
         SHOULD NOT transmit at a higher bit rate.  The default maximum
         value is 53400 bits per second, which is the maximum bitrate
         possible for iSAC.

   Encoding considerations:
      This media format is framed and binary.

   Security considerations:  See Section 7

   Interoperability considerations:  None

   Published specification:  RFC XXXX

   Applications which use this media type:
      This media type is suitable for use in numerous applications
      needing to transport encoded voice or other audio.  Some examples
      include Voice over IP, Streaming Media, Voice Messaging, and

   Fragment identifier considerations  The meaning of fragment
      identifiers is not defined by this specification.

   Additional information:  None

   Person to contact for further information:
      Tina le Grand [tlegrand@google.com]

   Intended usage:  COMMON

   Other Information/General Comment:
      iSAC is a speech and audio codec owned by Google.  The codec
      operates on 30 or 60 ms speech frames at a sampling rate clock of
      16 kHz or 32 kHz.

   Restrictions on usage:
      This media type depends on RTP framing, and hence is only defined
      for transfer via RTP [RFC3550].  Transport within other framing
      protocols is not defined at this time.

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   Author  Tina Le Grand and the listed authors of RFC XXXX

   Change controller:  The IETF Payload working group delegated from the

   Provisional registration?  No

   Note to the RFC Editor / IANA: Please replace "RFC XXXX" above with
   the number of this RFC when published, and remove this note.

6.  Mapping to SDP Parameters

   The information carried in the media type specification has a
   specific mapping to fields in the Session Description Protocol (SDP)
   [RFC4566], which is commonly used to describe RTP sessions.  When SDP
   is used to specify sessions employing the iSAC codec, the mapping is
   as follows:

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

   o  The media subtype (payload format name) goes in SDP "a=rtpmap" as
      the encoding name.

   o  The clock rate is 16000 for wideband, and 32000 for superwideband.

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

   The optional parameter ibitrate MUST NOT be higher than the parameter

   The iSAC parameters in an SDP offer are completely independent from
   those in the SDP answer.  For both ibitrate and maxbitrate it is
   legal for the answer to contain a value that is different than what
   is provided in an offer.  The parameter may be present in the answer,
   even if absent in the offer.

   When conveying information by SDP, the encoding name SHALL be "isac"
   (the same as the media subtype).

6.1.  Example Initial Target Bit Rate

   The offer indicates that it wishes to receive a wideband bitstream
   with an initial target rate of 20000 bits per second.  The remote
   party should change its initial target rate to the requested value or

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   m=audio 10000 RTP/AVP 98
   a=rtpmap: 98 isac/16000
   a=fmtp:98 ibitrate=20000

6.2.  Example Max Bit Rate

   The offer indicates that it wishes to receive a superwideband
   bitstream with an initial target rate of 20000 bits per second, and a
   maximum bit rate of 45000 bits per second.  The remote party should
   change its initial target rate to 20000 bits per second or less, and
   should not transmit at a higher rate than 45000.

   m=audio 10000 RTP/AVP 98
   a=rtpmap: 98 isac/32000
   a=fmtp:98 ibitrate=20000;maxbitrate=45000

6.3.  Example with both WB and SWB offered

   This offer indicates willingness to receive both wideband and
   superwideband iSAC encodings, with default values for ibitrate and
   bitrate.  Superwideband is preferred.

   m=audio 10000 RTP/AVP 98 99
   a=rtpmap: 98 isac/32000
   a=rtpmap: 99 isac/16000

7.  Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the general security considerations discussed in RFC
   3550 section 8.1.

   As this format transports encoded speech, the main security issues
   include confidentiality and authentication of the speech itself.  The
   payload format itself does not have any built-in security mechanisms.
   External mechanisms, such as SRTP [RFC3711], MAY be used.

   Since iSAC is a variable rate codec, the attack using the length of
   encoded packets described in [RFC6562] is of interest.  When using
   RTP for transport, the padding approach described in that document is
   usable; when such padding is not available or not feasible, the iSAC
   padding mechanism can be used to the same effect.

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8.  Acknowledgments

   Special thanks to Roni Even for his thorough review of the document,
   and to Colin Perkins for additional review.

   This document was originally prepared using 2-Word-v2.0.template.dot.

   The present version is prepared using xml2rfc and xxe-xml2rfc.

9.  References

9.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", RFC 3550, July 2003.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC4855]  Casner, S., "Media Type Registration of RTP Payload
              Formats", RFC 4855, February 2007.

   [RFC6838]  Freed, N., Klensin, J., and T. Hansen, "Media Type
              Specifications and Registration Procedures", BCP 13,
              RFC 6838, January 2013.

9.2.  Informative References

   [RFC6562]  Perkins, C. and JM. Valin, "Guidelines for the Use of
              Variable Bit Rate Audio with Secure RTP", RFC 6562,
              March 2012.

   [iSAC]     GIPS / Google, "iSAC reference implementation".

              Available at http://code.google.com/p/webrtc/source -
              directory src/modules/audio_coding/codecs/isac

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Authors' Addresses

   Tina le Grand
   Kungsbron 2
   Stockholm,   11122

   Paul E. Jones
   Cisco Systems
   7025 Kit Creek Rd.
   Research Triangle Park, NC  27709

   Phone: +1 919 476 2048
   Email: paulej@packetizer.com

   Pascal Huart
   Cisco Systems
   400, Avenue Roumanille, Batiment T3
   Biot - Sophia Antipolis,   06410

   Phone: +33 4 9723 2643
   Email: phuart@cisco.com

   Turaj Zakizadeh Shabestary
   1950 Charleston Road
   Mountain View, CA  94043

   Email: turajs@google.com

le Grand, et al.         Expires August 12, 2013               [Page 15]

Internet-Draft                    iSAC                     February 2013

   Harald Alvestrand (editor)
   Kungsbron 2
   Stockholm,   11122

   Email: hta@google.com

le Grand, et al.         Expires August 12, 2013               [Page 16]

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