[Docs] [txt|pdf|xml|html] [Tracker] [Email] [Nits]

Versions: 00 01 02 03 draft-ietf-payload-rtp-opus

Network Working Group                                         J. Spittka
Internet-Draft                                                    K. Vos
Intended status: Informational                   Skype Technologies S.A.
Expires: January 5, 2012                                       JM. Valin
                                                            Octasic Inc.
                                                            July 4, 2011


  RTP Payload Format and File Storage Format for Opus Speech and Audio
                                 Codec
                 draft-spittka-payload-rtp-opus-00

Abstract

   This document defines the Real-time Transport Protocol (RTP) payload
   format and file storage format for packetization of Opus encoded
   speech and audio data that is essential to integrate the codec in the
   most compatible way.  Further, media type registrations are described
   for the RTP payload format and the file storage format.

Status of this Memo

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

   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.

   This Internet-Draft will expire on January 5, 2012.

Copyright Notice

   Copyright (c) 2011 IETF Trust and the persons identified as the
   document authors.  All rights reserved.




Spittka, et al.          Expires January 5, 2012                [Page 1]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


   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 BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Conventions, Definitions and Acronyms used in this document  .  4
   3.  Opus Codec . . . . . . . . . . . . . . . . . . . . . . . . . .  5
     3.1.  Modes  . . . . . . . . . . . . . . . . . . . . . . . . . .  5
       3.1.1.  Audio Mode . . . . . . . . . . . . . . . . . . . . . .  5
       3.1.2.  Audio Mode . . . . . . . . . . . . . . . . . . . . . .  6
     3.2.  Network Bandwidth  . . . . . . . . . . . . . . . . . . . .  6
       3.2.1.  Variable versus Constant Bit Rate  . . . . . . . . . .  6
       3.2.2.  Discontinuous Transmission (DTX) . . . . . . . . . . .  7
     3.3.  Complexity . . . . . . . . . . . . . . . . . . . . . . . .  7
     3.4.  Forward Error Correction (FEC) . . . . . . . . . . . . . .  7
     3.5.  Stereo Operation . . . . . . . . . . . . . . . . . . . . .  8
   4.  Opus RTP Payload Format  . . . . . . . . . . . . . . . . . . .  9
     4.1.  RTP Header Usage . . . . . . . . . . . . . . . . . . . . .  9
     4.2.  Payload Structure  . . . . . . . . . . . . . . . . . . . . 10
   5.  Opus Storage Format  . . . . . . . . . . . . . . . . . . . . . 12
     5.1.  Storage Header Structure . . . . . . . . . . . . . . . . . 12
     5.2.  Storage Block Structure  . . . . . . . . . . . . . . . . . 12
   6.  Congestion Control . . . . . . . . . . . . . . . . . . . . . . 14
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 15
     7.1.  Opus Media Type Registration . . . . . . . . . . . . . . . 15
     7.2.  Mapping to SDP Parameters  . . . . . . . . . . . . . . . . 18
       7.2.1.  Offer-Answer Model Considerations for Opus . . . . . . 19
       7.2.2.  Declarative SDP Considerations for Opus  . . . . . . . 20
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 22
   9.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23
   10. Normative References . . . . . . . . . . . . . . . . . . . . . 24
   A.  Informational References . . . . . . . . . . . . . . . . . . . 25
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 26









Spittka, et al.          Expires January 5, 2012                [Page 2]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


1.  Introduction

   The Opus codec is a speech and audio codec developed within the IETF
   Internet Wideband Audio Codec working group [codec].  The codec has a
   very low algorithmic delay and is is highly scalable in terms of
   audio bandwidth, network bit rate, and complexity.  Further, it
   provides different modes to efficiently encode speech signals as well
   as music signals, thus, making it the codec of choice for various
   applications using the Internet or similar networks.

   This document defines the Real-time Transport Protocol (RTP)
   [RFC3550] payload format and file storage format for packetization of
   Opus encoded speech and audio data that is essential to integrate the
   Opus codec in the most compatible way.  Further, media type
   registrations are described for the RTP payload format and the file
   storage format.  More information on the Opus codec can be obtained
   from the following IETF draft [Opus].


































Spittka, et al.          Expires January 5, 2012                [Page 3]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


2.  Conventions, Definitions and Acronyms used in this document

   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 [RFC2119].

   CPU:  Central Processing Unit
   IP:  Internet Protocol
   PSTN:  Public Switched Telephone Network
   samples:  Speech or audio samples
   SDP:  Session Description Protocol








































Spittka, et al.          Expires January 5, 2012                [Page 4]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


3.  Opus Codec

   The Opus speech and audio codec has been developed to encode speech
   signals as well as audio signals.  Two different modes, a voice mode
   or an audio mode, may be chosen to allow the most efficient coding
   dependent on the type of input signal, the sampling frequency of the
   input signal, and the specific application.

   The voice mode allows to efficiently encode voice signals at lower
   bit rates while the audio mode is optimized for audio signals at
   medium and higher bit rates.

   The Opus speech and audio codec is highly scalable in terms of audio
   bandwidth, network bit rate, and complexity.  Further, Opus allows to
   transmit stereo signals.

   The Opus speech and audio codec is based on the SILK codec [SILK] and
   the CELT codec [CELT].  For more detailed information on how Opus
   operates, also refer to [Opus].

3.1.  Modes

   Opus supports five different audio bandwidths, 8000, 12000, 16000,
   24000, and 48000 Hz sampling frequency, for the voice mode and four
   different audio bandwidths, 8000, 16000, 24000, and 48000 Hz sampling
   frequency, for the audio mode.

3.1.1.  Audio Mode

   For low bit rate applications transmitting mostly speech signals the
   voice mode of Opus SHOULD be used.  The voice mode allows to encode
   voice signals at 8000, 12000, 16000, 24000, and 48000 Hz sampling
   frequency.

   A sampling rate of 8000 Hz SHOULD only be used to interface to PSTN
   networks or on low end devices that do not support greater than 8000
   Hz sampling frequency.  A sampling rate of 12000 Hz SHOULD be used
   for lower end devices that do not support greater than 12000 Hz
   sampling frequency or are under severe network bandwidth constrains
   (e.g. wireless devices).  A sampling rate of 16000 Hz SHOULD be used
   for all-IP platforms that do not support greater than 16000 Hz
   sampling frequency.  Higher sampling rates are recommended for all
   devices that support those high sampling rates and desire full-
   bandwidth speech at medium bit rates.







Spittka, et al.          Expires January 5, 2012                [Page 5]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


3.1.2.  Audio Mode

   For applications desiring very low delay speech transmission as well
   as music transmission in trade off to a higher bit rate, the audio
   mode SHOULD be used.  This mode supports audio sampling rates of
   8000, 16000, 24000, and 48000 Hz.

3.2.  Network Bandwidth

   The network bit rate is adaptive within the range specified in
   Table 1 for corresponding modes and audio sampling rates.  The
   average target network bit rate can be defined and modified in real-
   time while the actual bit rate will be dependent on the settings of
   Opus and the input signal and may change over time.

                      +-------+---------+-----------+
                      |  Mode | fs (Hz) | BR (kbps) |
                      +-------+---------+-----------+
                      | voice |   8000  |   6 - 20  |
                      |       |         |           |
                      | voice |  12000  |   7 - 25  |
                      |       |         |           |
                      | voice |  16000  |   8 - 30  |
                      |       |         |           |
                      | voice |  24000  |  18 - 28  |
                      |       |         |           |
                      | voice |  48000  |  24 - 32  |
                      |       |         |           |
                      | audio |   8000  |  20 - 28  |
                      |       |         |           |
                      | audio |  16000  |  24 - 32  |
                      |       |         |           |
                      | audio |  24000  |  28 - 40  |
                      |       |         |           |
                      | audio |  48000  |  32 - 128 |
                      +-------+---------+-----------+

     Mode specifies the Opus mode of operation; fs specifies the audio
    sampling frequency in Hertz (Hz); BR specifies the network bit rate
                   range in kilobits per second (kbps).

                                  Table 1

3.2.1.  Variable versus Constant Bit Rate

   The voice mode will always use a variable bit rate at audio sampling
   rates of 8000, 12000, and 16000 Hz.  The average target bit rate can
   be adjusted at any point in time.  To avoid congestion of the



Spittka, et al.          Expires January 5, 2012                [Page 6]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


   connection the average target bit rate SHOULD be adjusted to the
   available network bandwidth.  If no target bit rate is specified the
   average bit rate may go up to the highest bit rate specified in
   Table 1.

   In voice mode at audio sampling rates higher than 16000 Hz, i.e.
   24000, and 48000 Hz, and audio mode Opus can be operated in both
   variable and constant bit rate.  The target bit rate can be adjusted
   at any point in time.

3.2.2.  Discontinuous Transmission (DTX)

   The Opus codec may, as described in Section 3.2.1, be operated with
   an adaptive bit rate.  In that case, the bit rate will automatically
   be reduced for certain input signals like periods of silence.  During
   continuous transmission the bit rate will be reduced, when the input
   signal allows to do so, but the transmission to the receiver itself
   will never be interrupted.  Therefore, the received signal will
   maintain the same high level of quality over the full duration of a
   transmission while minimizing the average bit rate over time.

   In cases where the bit rate of Opus needs to be reduced even further
   or in cases where only constant bit rate is available, the Opus
   encoder may be set to use discontinuous transmission (DTX), where
   parts of the encoded signal that correspond to periods of silence in
   the input speech or audio signal are not transmitted to the receiver.

   On the receiving side, the non-transmitted parts will be handled by a
   frame loss concealment unit in the Opus decoder which generates a
   comfort noise signal to replace the non transmitted parts of the
   speech or audio signal.

   The DTX mode of Opus will have a slightly lower speech or audio
   quality than the continuous mode.  Therefore, it is RECOMMENDED to
   use Opus in the continuous mode unless restraints on network
   bandwidth are severe.  The DTX mode can be engaged for operation in
   both adaptive or constant bit rate.

3.3.  Complexity

   Complexity can be scaled to optimize for CPU resources in real-time,
   mostly in trade-off to network bit rate.  Also, different modes of
   Opus have different complexity.

3.4.  Forward Error Correction (FEC)

   The voice mode of Opus allows for "in-band" forward error correction
   (FEC) data to be embedded into the bit stream of Opus.  This FEC



Spittka, et al.          Expires January 5, 2012                [Page 7]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


   scheme adds redundant information about the previous packet (n-1) to
   the current output packet n.  For each frame, the encoder decides
   whether to use FEC based on (1) an externally-provided estimate of
   the channel's packet loss rate; (2) an externally-provided estimate
   of the channel's capacity; (3) the sensitivity of the audio or speech
   signal to packet loss; (4) whether the receiving decoder has
   indicated it can take advantage of "in-band" FEC information.  The
   decision to send "in-band" FEC information is entirely controlled by
   the encoder and therefore no special precautions for the payload or
   storage format have to be taken.

   On the receiving side, the decoder can take advantage of this
   additional information when, in case of a packet loss, the next
   packet is available.  In order to use the FEC data, the jitter buffer
   needs to provide access to payloads with the FEC data.  The decoder
   API function has a flag to indicate that a FEC frame rather than a
   regular frame should be decoded.  If no FEC data is available for the
   current frame, the decoder will consider the frame lost and invokes
   the frame loss concealment.

   If the FEC scheme is not implemented on the receiving side, FEC
   SHOULD NOT be used, as it leads to an inefficient usage of network
   bandwidth.  Decoder support for FEC SHOULD be indicated at the time a
   session is set up.

3.5.  Stereo Operation

   Opus allows for transmission of stereo audio signals.  This operation
   will be signaled in the Opus payload and no special arrangements have
   to be made in the payload format.  Any implementation of the Opus
   decoder MUST be capable to receive stereo signals.

   If a decoder can not take advantage of the benefits of a stereo
   signal this SHOULD be indicated at the time a session is set up.  In
   that case the sending side SHOULD NOT send stereo signals as it leads
   to an inefficient usage of network bandwidth.















Spittka, et al.          Expires January 5, 2012                [Page 8]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


4.  Opus RTP Payload Format

   The payload format for Opus consists of the RTP header and Opus
   payload data.

4.1.  RTP Header Usage

   The format of the RTP header is specified in [RFC3550].  The Opus
   payload format uses the fields of the RTP header consistent with this
   specification.

   The payload length of Opus is a multiple number of octets and
   therefore no padding is required.  The payload MAY be padded by an
   integer number of octets according to [RFC3550].

   The marker bit (M) of the RTP header has no function in combination
   with Opus and MAY be ignored.

   The RTP payload type for Opus has not been assigned statically and is
   expected to be assigned dynamically.

   The receiving side MUST be prepared to receive duplicates of RTP
   packets.  Only one of those payloads MUST be provided to the Opus
   decoder for decoding and others MUST be discarded.

   Opus supports 5 different sampling rates which may be adjusted during
   the duration of a call.  The RTP timestamp clock frequency is defined
   as the highest supported sampling frequency of Opus, i.e. 48000 Hz,
   for all modes and sampling rates of Opus.  The unit for the timestamp
   is samples.  The RTP timestamp corresponds to the sample time of the
   first encoded sample in the encoded frame.  For sampling rates lower
   than 48000 Hz the number of samples has to be multiplied with a
   multiplier according to Table 2 to determine the RTP timestamp.


















Spittka, et al.          Expires January 5, 2012                [Page 9]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


                         +---------+------------+
                         | fs (Hz) | Multiplier |
                         +---------+------------+
                         |   8000  |      6     |
                         |         |            |
                         |  12000  |      4     |
                         |         |            |
                         |  16000  |      3     |
                         |         |            |
                         |  24000  |      2     |
                         |         |            |
                         |  48000  |      1     |
                         +---------+------------+

    fs specifies the audio sampling frequency in Hertz (Hz); Multiplier
   is the value that the number of samples have to be multiplied with to
                       calculate the RTP timestamp.

                                  Table 2

4.2.  Payload Structure

   The Opus encoder can be set to output encoded frames representing
   2.5, 5, 10, 20, 40, or 60 ms of speech or audio data.  Further, an
   arbitrary number of frames can be combined into a packet.  The
   maximum packet length is limited to the amount of encoded data
   representing 120 ms of speech or audio data.  The packetization of
   encoded data is purely done by the Opus encoder and therefore only
   one packet output from the Opus encoder MUST be used as a payload.

   Figure 1 shows the structure combined with the RTP header.


   +----------+--------------+
   |RTP Header| Opus Payload |
   +----------+--------------+


                Figure 1: Payload Structure with RTP header

   Table 3 shows supported frame sizes for different modes and sampling
   rates of Opus and how the timestamp needs to be incremented for
   packetization.








Spittka, et al.          Expires January 5, 2012               [Page 10]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


   +------+------------------------+----+----+-----+-----+------+------+
   | Mode |           fs           | 2. |  5 |  10 |  20 |  40  |  60  |
   |      |                        |  5 |    |     |     |      |      |
   +------+------------------------+----+----+-----+-----+------+------+
   |  ts  |           all          | 12 | 24 | 480 | 960 | 1920 | 2880 |
   | incr |                        |  0 |  0 |     |     |      |      |
   |      |                        |    |    |     |     |      |      |
   | voic | 8000/12000/16000/24000 |    |    |  x  |  x  |   x  |   x  |
   |   e  |         /48000         |    |    |     |     |      |      |
   |      |                        |    |    |     |     |      |      |
   | audi | 8000/16000/24000/48000 |  x |  x |  x  |  x  |      |      |
   |   o  |                        |    |    |     |     |      |      |
   +------+------------------------+----+----+-----+-----+------+------+

     Mode specifies the Opus mode of operation; fs specifies the audio
       sampling frequency in Hertz (Hz); 2.5, 5, 10, 20, 40, and 60
    represent the duration of encoded speech or audio data in a packet;
   ts incr specifies the value the timestamp needs to be incremented for
   the representing packet size.  For multiple frames in a packet these
    values have to be multiplied with the respective number of frames.

                                  Table 3





























Spittka, et al.          Expires January 5, 2012               [Page 11]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


5.  Opus Storage Format

   The Opus storage format allows to store Opus encoded data into e.g. a
   file or an email attachment.  The storage format consists of a header
   and a series of blocks containing encoded speech or audio frames.
   The storage format closely mimics the real-time payload format and
   allows to easily convert packets, e.g. received by a voicemail
   system, into a storage format and vice versa and therefore allowing
   maximum flexibility and low overhead.  Please note that this storage
   format is not meant to be a robust storage format, nor the most
   efficient storage format.  For a robust storage format that allows
   advanced functionality like e.g. seeking, a more advanced container
   format should be used.

   Figure 2 shows an example of an Opus encoded file.  Note that due to
   the potentially adaptive bit rate the packet length may be variable
   and no fixed block size can be defined for blocks containing encoded
   data.

   +------------------+
   | Header           |
   +-----------+------+
   | block 1   |
   +-----------+--+
   | block 2      |
   +--------------+--+
   : ...             :
   +--------------+--+
   | block n         |
   +-----------------+

   Figure 2: Example of Opus file storage format showing different block
           lengths due to potentially adaptive bit rate of Opus

5.1.  Storage Header Structure

   An Opus storage header contains the following ASCII character string
   as a magic number:

   "#!opus\n" (hexadecimal: 0x23 0x21 0x6f 0x70 0x75 0x73 0x0A)

5.2.  Storage Block Structure

   Following the storage header, blocks of encoded data are stored in
   consecutive order in time according to Figure 2.  Each block contains
   a block header followed by a payload according to Figure 3.

   The block header contains information that, for an RTP-based session,



Spittka, et al.          Expires January 5, 2012               [Page 12]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


   can be derived from the IP and RTP headers: The number of octets
   contained in the subsequent payload and the RTP timestamp.

   The number of octets in the payload is represented by 16 bits and the
   timestamp is specified by 32 bits.  For the first block, the
   timestamp MAY be a random number.  For the following blocks, the
   timestamp MUST be incremented according to the way timestamps are
   incremented when Opus payloads are transmitted over RTP.

   0                   16                           48
   +-------------------+----------------------------+-----------------
   |    # of octets    |        Timestamp           |  Payload
   +-------------------+----------------------------+-----------------

                 Figure 3: Storage block header structure

   The payload of each block in Figure 2 represents one packet of Opus
   encoded data the way as originally encoded by the Opus encoder.
   Information about frame size representing the duration of encoded
   speech or audio data, number of encoded frames, stereo information,
   and DTX is embedded into the payload of Opus and not subject to the
   storage format.  It can be extracted from the payload during decoding
   of the encoded data.

   During the usage of DTX no blocks are stored when the channel is
   inactive.  Timestamps MUST be used to reassemble the decoded signal
   in a time-aligned way.
























Spittka, et al.          Expires January 5, 2012               [Page 13]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


6.  Congestion Control

   The adaptive nature of the Opus codec allows for an efficient
   congestion control.

   The voice mode of Opus at audio sampling rates of 8000, 12000, and
   16000 always runs with a variable bit rate.  The average bit rate in
   that mode is dependent on the input signal and will especially
   decrease during silent periods.  The voice mode at audio sampling
   rates of 24000 and 48000 Hz and the audio mode may run at a variable
   or constant bit rate.  In either way, the target bit rate of Opus can
   be adjusted at any point in time and thus allowing for an efficient
   congestion control.

   Furthermore, the amount of encoded speech or audio data encoded in a
   single packet can be used for congestion control since the
   transmission rate is inversely proportional to these frame sizes.  A
   lower packet transmission rate reduces the amount of header overhead
   but at the same time increases latency and error sensitivity and
   should be done with care.

   It is RECOMMENDED that congestion control is applied during the
   transmission of Opus encoded data.




























Spittka, et al.          Expires January 5, 2012               [Page 14]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


7.  IANA Considerations

   One media subtype (audio/opus) has been defined and registered as
   described in the following section.

7.1.  Opus Media Type Registration

   Media type registration is done according to [RFC4288] and [RFC4855].


   Type name: audio


   Subtype name: opus


   Required parameters:

   rate:  RTP timestamp clock rate is incremented with 48000 Hz clock
      rate for all modes of Opus and all sampling frequencies.  For
      audio sampling rates other than 48000 Hz the rate has to be
      adjusted to 48000 Hz according to Table 2.

   Optional parameters:

   maxcodedaudiobandwidth:  the decoder's maximum sampling frequency
      specified in Hertz (Hz) that the application can take advantage
      of.  The decoder MUST be capable to receive any allowed sampling
      frequency but due to hardware limitations only signals up to the
      specified sampling frequency can be processed.  Sending signals
      with higher sampling frequency may result in higher than necessary
      network bandwidth and encoding complexity.  Possible values are
      8000, 12000, 16000, 24000, 48000.

   maxptime:  the decoder's maximum length of time in milliseconds
      rounded up to the next full integer value represented by the media
      in a packet that can be encapsulated in a received packet
      according to Section 6 of [RFC4566].  Possible values are 3, 5,
      10, 20, 40, and 60 or an arbitrary multiple of Opus frame sizes
      rounded up to the next full integer value up to a maximum value of
      120 as defined in Section 4 and Section 5 of this document.  If no
      value is specified, 120 is assumed as default.  This value is a
      recommendation by the decoding side to ensure the best performance
      for the decoder.  The decoder MUST be capable to accept any
      allowed packet sizes to ensure maximum compatibility.






Spittka, et al.          Expires January 5, 2012               [Page 15]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


   ptime:  the decoder's recommended length of time in milliseconds
      rounded up to the next full integer value represented by the media
      in a packet according to Section 6 of [RFC4566].  Possible values
      are 3, 5, 10, 20, 40, or 60 or an arbitrary multiple of Opus frame
      sizes rounded up to the next full integer value up to a maximum
      value of 120 as defined in Section 4 and Section 5 of this
      document.  If no value is specified, 20 is assumed as default.  If
      ptime is greater than maxptime, ptime MUST be ignored.  This
      parameter MAY be changed during a session.  This value is a
      recommendation by the decoding side to ensure the best performance
      for the decoder.  The decoder MUST be capable to accept any
      allowed packet sizes to ensure maximum compatibility.

   minptime:  the decoder's minimum length of time in milliseconds
      rounded up to the next full integer value represented by the media
      in a packet that SHOULD be encapsulated in a received packet
      according to Section 6 of [RFC4566].  Possible values are 3, 5,
      10, 20, 40, and 60 or an arbitrary multiple of Opus frame sizes
      rounded up to the next full integer value up to a maximum value of
      120 as defined in Section 4 and Section 5 of this document.  If no
      value is specified, 3 is assumed as default.  This value is a
      recommendation by the decoding side to ensure the best performance
      for the decoder.  The decoder MUST be capable to accept any
      allowed packet sizes to ensure maximum compatibility.

   maxaveragebitrate:  specifies the maximum average receive bit rate of
      a session in bits per second (bps).  The actual value of the bit
      rate may vary as it is dependent on the characteristics of the
      media in a packet.  Note that the maximum average bit rate MAY be
      modified dynamically during a session.  Any positive integer is
      allowed but values outside the range between 6000 and 510000
      SHOULD be ignored.  If no value is specified, the maximum value
      specified in Table 1 for the corresponding mode of Opus and
      corresponding clock rate will be the default.

   stereo:  specifies if the decoder prefers to receive stereo signals
      versus mono signals.  Possible values are 1 and 0 where 1
      specifies that stereo signals are preferred and 0 specifies that
      only mono signals are preferred.  Independent of the stereo
      parameter every receiver MUST be able to receive and decode stereo
      signals but sending stereo signals to a receiver that signaled a
      preference for mono signals may result in higher than necessary
      network bandwidth and encoding complexity.  If no value is
      specified, stereo is assumed to be 0.







Spittka, et al.          Expires January 5, 2012               [Page 16]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


   cbr:  specifies if the decoder prefers the use of a constant bit rate
      versus variable bit rate.  Possible values are 1 and 0 where 1
      specifies constant bit rate and 0 specifies variable bit rate.  If
      no value is specified, cbr is assumed to be 0.  Note that the
      maximum average bit rate may still be changed, e.g. to adapt to
      changing network conditions.

   useinbandfec:  specifies that Opus in-band FEC is supported by the
      decoder and MAY be used during a session.  Possible values are 1
      and 0.  It is RECOMMENDED to provide 0 in case FEC is not
      implemented on the receiving side.  If no value is specified,
      useinbandfec is assumed to be 1.

   usedtx:  specifies if the decoder prefers the use of DTX.  Possible
      values are 1 and 0.  If no value is specified, usedtx is assumed
      to be 0.


   Encoding considerations:


      Opus media type is framed and consists of binary data according to
      Section 4.8 in [RFC4288].

   Security considerations:

      See Section 8 of this document.

   Interoperability considerations: none


   Published specification: none


   Applications that use this media type:

      Any application that requires the transport or storage of speech
      or audio data may use this media type.  Some examples are, but not
      limited to, audio and video conferencing, Voice over IP, voice
      recording, media streaming, voice messaging.

   Additional information:

      For storage transfer methods the following applies:







Spittka, et al.          Expires January 5, 2012               [Page 17]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


      Magic number:"#!opus\n" (hexadecimal: 0x23 0x21 0x6f 0x70 0x75
      0x73 0x0A)

      File extension(s): ops, OPS

      Macintosh file type code(s): "opus"

   Person & email address to contact for further information:

      SILK Support silksupport@skype.net
      Jean-Marc Valin jean-marc.valin@octasic.com

   Intended usage: COMMON


   Restrictions on usage:


      For transfer over RTP, the RTP payload format (Section 4 of this
      document) SHALL be used.  For storage usage, the storage format
      (Section 5 of this document) SHALL be used.

   Author:

      Julian Spittka julian.spittka@skype.net

      Koen Vos koen.vos@skype.net

      Jean-Marc Valin jean-marc.valin@octasic.com


   Change controller: TBD

7.2.  Mapping to SDP Parameters

   The information described 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 Opus codec, the mapping is
   as follows:

   o  The media type ("audio") goes in SDP "m=" as the media name.
   o  The media subtype ("opus") goes in SDP "a=rtpmap" as the encoding
      name.  The RTP clock rate in "a=rtpmap" MUST be mapped to the
      required media type parameter "rate".
   o  The optional media type parameters "ptime" and "maxptime" are
      mapped to "a=ptime" and "a=maxptime" attributes, respectively, in
      the SDP.



Spittka, et al.          Expires January 5, 2012               [Page 18]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


   o  All remaining media type parameters are mapped to the "a=fmtp"
      attribute in the SDP by copying them directly from the media type
      parameter string as a semicolon-separated list of parameter=value
      pairs (e.g. maxaveragebitrate=20000).

   Below are some examples of SDP session descriptions for Opus:

   Example 1: Standard session with 48000 Hz clock rate


       m=audio 54312 RTP/AVP 101
       a=rtpmap:101 opus/48000


   Example 2: 16000 Hz clock rate, maximum packet size of 40 ms,
   recommended packet size of 40 ms, maximum average bit rate of 20000
   bps, stereo signals are preferred, FEC is allowed, DTX is not allowed


       m=audio 54312 RTP/AVP 101
       a=rtpmap:101 opus/48000
       a=fmtp:101 maxcodedaudiobandwidth=16000; maxaveragebitrate=20000;
       stereo=1; useinbandfec=1; usedtx=0
       a=ptime:40
       a=maxptime:40


7.2.1.  Offer-Answer Model Considerations for Opus

   When using the offer-answer procedure described in [RFC3264] to
   negotiate the use of Opus, the following considerations apply:

   o  Opus supports several clock rates.  For signaling purposes only
      the highest, i.e. 48000, is used.  The actual clock rate of the
      corresponding media is signaled inside the payload and is not
      subject to this payload format description.  The decoder MUST be
      capable to decode every received clock rate.  An example is shown
      below:


           m=audio 54312 RTP/AVP 100
           a=rtpmap:100 opus/48000


   o  The parameters "ptime" and "maxptime" are unidirectional receive-
      only parameters and typically will not compromise
      interoperability; however, dependent on the set values of the
      parameters the performance of the application may suffer.



Spittka, et al.          Expires January 5, 2012               [Page 19]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


      [RFC3264] defines the SDP offer-answer handling of the "ptime"
      parameter.  The "maxptime" parameter MUST be handled in the same
      way.
   o  The parameter "minptime" is a unidirectional receive-only
      parameters and typically will not compromise interoperability;
      however, dependent on the set values of the parameter the
      performance of the application may suffer and should be set with
      care.
   o  The parameter "maxcodedaudiobandwidth" is a unidirectional
      receive-only parameter that reflects limitations of the local
      receiver.  The sender of the other side SHOULD NOT send with a
      sampling rate higher than "maxcodedaudiobandwidth" as it
      represents an inefficient use of network bandwidth resources and
      CPU cycles on the encoding side.  The parameter
      "maxcodedaudiobandwidth" typically will not compromise
      interoperability; however, dependent on the set value of the
      parameter the performance of the application may suffer and should
      be set with care.
   o  The parameter "maxaveragebitrate" is a unidirectional receive-only
      parameter that reflects limitations of the local receiver.  The
      sender of the other side MUST NOT send with an average bit rate
      higher than "maxaveragebitrate" as it might overload the network
      and/or receiver.  The parameter "maxaveragebitrate" typically will
      not compromise interoperability; however, dependent on the set
      value of the parameter the performance of the application may
      suffer and should be set with care.
   o  If the parameter "maxaveragebitrate" is below the range specified
      in Table 1 the session MUST be rejected.
   o  The parameter "stereo" is a unidirectional receive-only parameter.
   o  The parameter "cbr" is a unidirectional receive-only parameter.
   o  The parameter "useinbandfec" is a unidirectional receive-only
      parameter.
   o  The parameter "usedtx" is a unidirectional receive-only parameter.
   o  Any unknown parameter in an offer MUST be ignored by the receiver
      and MUST be removed from the answer.

7.2.2.  Declarative SDP Considerations for Opus

   For declarative use of SDP such as in Session Announcement Protocol
   (SAP), [RFC2974], and RTSP, [RFC2326], for Opus, the following needs
   to be considered:

   o  The values for "maxptime", "ptime", "minptime",
      "maxcodedaudiobandwidth", and "maxaveragebitrate" should be
      selected carefully to ensure that a reasonable performance can be
      achieved for the participants of a session.





Spittka, et al.          Expires January 5, 2012               [Page 20]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


   o  The values for "maxptime", "ptime", and "minptime" of the payload
      format configuration are recommendations by the decoding side to
      ensure the best performance for the decoder.  The decoder MUST be
      capable to accept any allowed packet sizes to ensure maximum
      compatibility.
   o  All other parameters of the payload format configuration are
      declarative and a participant MUST use the configurations that are
      provided for the session.  More than one configuration may be
      provided if necessary by declaring multiple RTP payload types;
      however, the number of types should be kept small.









































Spittka, et al.          Expires January 5, 2012               [Page 21]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


8.  Security Considerations

   All RTP packets using the payload format defined in this
   specification are subject to the general security considerations
   discussed in the RTP specification [RFC3550] and any profile from
   e.g.  [RFC3711] or [RFC3551].

   This payload format transports Opus encoded speech or audio data,
   hence, security issues include confidentiality, integrity protection,
   and authentication of the speech or audio itself.  The Opus payload
   format does not have any built-in security mechanisms.  Any suitable
   external mechanisms, such as SRTP [RFC3711], MAY be used.

   This payload format and the Opus encoding do not exhibit any
   significant non-uniformity in the receiver-end computational load and
   thus are unlikely to pose a denial-of-service threat due to the
   receipt of pathological datagrams.


































Spittka, et al.          Expires January 5, 2012               [Page 22]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


9.  Acknowledgements

   TBD
















































Spittka, et al.          Expires January 5, 2012               [Page 23]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


10.  Normative References

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

   [RFC2326]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
              Streaming Protocol (RTSP)", RFC 2326, April 1998.

   [RFC2974]  Handley, M., Perkins, C., and E. Whelan, "Session
              Announcement Protocol", RFC 2974, October 2000.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              June 2002.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              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.

   [RFC4288]  Freed, N. and J. Klensin, "Media Type Specifications and
              Registration Procedures", BCP 13, RFC 4288, December 2005.

   [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.
















Spittka, et al.          Expires January 5, 2012               [Page 24]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


Appendix A.  Informational References

      [codec] http://datatracker.ietf.org/wg/codec/
      [SILK] https://developer.skype.com/silk
      [CELT] http://www.celt-codec.org/
      [Opus] http://datatracker.ietf.org/doc/draft-ietf-codec-opus/













































Spittka, et al.          Expires January 5, 2012               [Page 25]


Internet-Draft      RTP Payload Format for Opus Codec          July 2011


Authors' Addresses

   Julian Spittka
   Skype Technologies S.A.
   3210 Porter Drive
   Palo Alto, CA  94304
   USA

   Email: julian.spittka@skype.net


   Koen Vos
   Skype Technologies S.A.
   3210 Porter Drive
   Palo Alto, CA  94304
   USA

   Email: koen.vos@skype.net


   Jean-Marc Valin
   Octasic Inc.
   4101 Molson Street
   Montreal, Quebec
   Canada

   Email: jean-marc.valin@octasic.com
























Spittka, et al.          Expires January 5, 2012               [Page 26]


Html markup produced by rfcmarkup 1.129b, available from https://tools.ietf.org/tools/rfcmarkup/