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

Network Working Group                                         J. Spittka
Internet-Draft
Intended status: Informational                                    K. Vos
Expires: June 3, 2013                            Skype Technologies S.A.
                                                               JM. Valin
                                                                 Mozilla
                                                       November 30, 2012


           RTP Payload Format for Opus Speech and Audio Codec
                   draft-spittka-payload-rtp-opus-03

Abstract

   This document defines the Real-time Transport Protocol (RTP) payload
   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.

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 June 3, 2013.

Copyright Notice

   Copyright (c) 2012 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



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   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.  Conventions, Definitions and Acronyms used in this document  .  4
     2.1.  Audio Bandwidth  . . . . . . . . . . . . . . . . . . . . .  4
   3.  Opus Codec . . . . . . . . . . . . . . . . . . . . . . . . . .  5
     3.1.  Network Bandwidth  . . . . . . . . . . . . . . . . . . . .  5
       3.1.1.  Recommended Bitrate  . . . . . . . . . . . . . . . . .  5
       3.1.2.  Variable versus Constant Bit Rate  . . . . . . . . . .  5
       3.1.3.  Discontinuous Transmission (DTX) . . . . . . . . . . .  6
     3.2.  Complexity . . . . . . . . . . . . . . . . . . . . . . . .  6
     3.3.  Forward Error Correction (FEC) . . . . . . . . . . . . . .  6
     3.4.  Stereo Operation . . . . . . . . . . . . . . . . . . . . .  7
   4.  Opus RTP Payload Format  . . . . . . . . . . . . . . . . . . .  8
     4.1.  RTP Header Usage . . . . . . . . . . . . . . . . . . . . .  8
     4.2.  Payload Structure  . . . . . . . . . . . . . . . . . . . .  9
   5.  Congestion Control . . . . . . . . . . . . . . . . . . . . . . 11
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 12
     6.1.  Opus Media Type Registration . . . . . . . . . . . . . . . 12
     6.2.  Mapping to SDP Parameters  . . . . . . . . . . . . . . . . 15
       6.2.1.  Offer-Answer Model Considerations for Opus . . . . . . 17
       6.2.2.  Declarative SDP Considerations for Opus  . . . . . . . 18
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 19
   8.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 20
   9.  Normative References . . . . . . . . . . . . . . . . . . . . . 21
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22




















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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 it is highly scalable in terms of
   audio bandwidth, bitrate, 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 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.  More information on the Opus codec can be
   obtained from [RFC6716].



































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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].

   CBR:  Constant bitrate
   CPU:  Central Processing Unit
   DTX:  Discontinuous transmission
   FEC:  Forward error correction
   IP:  Internet Protocol
   samples:  Speech or audio samples (usually per channel)
   SDP:  Session Description Protocol
   VBR:  Variable bitrate

2.1.  Audio Bandwidth

   Throughout this document, we refer to the following definitions:

         +--------------+----------------+-----------+----------+
         | Abbreviation |      Name      | Bandwidth | Sampling |
         +--------------+----------------+-----------+----------+
         |      nb      |   Narrowband   |  0 - 4000 |   8000   |
         |              |                |           |          |
         |      mb      |   Mediumband   |  0 - 6000 |   12000  |
         |              |                |           |          |
         |      wb      |    Wideband    |  0 - 8000 |   16000  |
         |              |                |           |          |
         |      swb     | Super-wideband | 0 - 12000 |   24000  |
         |              |                |           |          |
         |      fb      |    Fullband    | 0 - 20000 |   48000  |
         +--------------+----------------+-----------+----------+

                          Audio bandwidth naming

                                  Table 1















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3.  Opus Codec

   The Opus [RFC6716] 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 efficient encoding of voice signals at lower
   bit rates while the audio mode is optimized for audio signals at
   medium and higher bitrates.

   The Opus speech and audio codec is highly scalable in terms of audio
   bandwidth, bitrate, and complexity.  Further, Opus allows
   transmitting stereo signals.

3.1.  Network Bandwidth

   Opus supports all bitrates from 6 kb/s to 510 kb/s.  The bitrate can
   be changed dynamically within that range.  All other parameters being
   equal, higher bitrate results in higher quality.

3.1.1.  Recommended Bitrate

   For a frame size of 20 ms, these are the bitrate "sweet spots" for
   Opus in various configurations:
   o  8-12 kb/s for NB speech,
   o  16-20 kb/s for WB speech,
   o  28-40 kb/s for FB speech,
   o  48-64 kb/s for FB mono music, and
   o  64-128 kb/s for FB stereo music.

3.1.2.  Variable versus Constant Bit Rate

   For the same average bitrate, variable bitrate (VBR) can achieve
   higher quality than constant bitrate (CBR).  For the majority of
   voice transmission application, VBR is the best choice.  One
   potential reason for choosing CBR is the potential information leak
   that _may_ occur when encrypting the compressed stream.  See
   [RFC6562] for guidelines on when VBR is appropriate for encrypted
   audio communications.  In the case where an existing VBR stream needs
   to be converted to CBR for security reasons, then the Opus padding
   mechanism described in [RFC6716] is the RECOMMENDED way to achieve
   padding because the RTP padding bit is unencrypted.

   The bitrate can be adjusted at any point in time.  To avoid
   congestion, the average bitrate SHOULD be adjusted to the available
   network capacity.  If no target bitrate is specified, the bitrates



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   specified in Section 3.1.1 are RECOMMENDED.

3.1.3.  Discontinuous Transmission (DTX)

   The Opus codec may, as described in Section 3.1.2, be operated with
   an adaptive bitrate.  In that case, the bitrate will automatically be
   reduced for certain input signals like periods of silence.  During
   continuous transmission the bitrate 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 bitrate of Opus needs to be reduced even further
   or in cases where only constant bitrate 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 capacity
   are severe.  The DTX mode can be engaged for operation in both
   adaptive or constant bitrate.

3.2.  Complexity

   Complexity can be scaled to optimize for CPU resources in real-time,
   mostly as a trade-off between audio quality and bitrate.  Also,
   different modes of Opus have different complexity.

3.3.  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
   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



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   the encoder and therefore no special precautions for the payload 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
   resources.  Decoder support for FEC SHOULD be indicated at the time a
   session is set up.

3.4.  Stereo Operation

   Opus allows for transmission of stereo audio signals.  This operation
   is signaled in-band in the Opus payload and no special arrangement is
   required in the payload format.  Any implementation of the Opus
   decoder MUST be capable of receiving stereo signals, although it MAY
   decode those signals as mono.

   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 the network.






















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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 is used in accordance with
   Section 4.1 of [RFC3551].

   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 audio bandwidths 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 per single (mono) channel.  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.

















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

                       Table 2: Timestamp multiplier

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 in milliseconds of encoded speech
   or audio data for speech and audio mode (Mode) and sampling rates
   (fs) of Opus and how the timestamp needs to be incremented for
   packetization (ts incr).  If the Opus encoder outputs multiple
   encoded frames into a single packet the timestamps have to be added
   up according to the combined frames.









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    +---------+-----------------+-----+-----+-----+-----+------+------+
    |   Mode  |        fs       | 2.5 |  5  |  10 |  20 |  40  |  60  |
    +---------+-----------------+-----+-----+-----+-----+------+------+
    | ts incr |       all       | 120 | 240 | 480 | 960 | 1920 | 2880 |
    |         |                 |     |     |     |     |      |      |
    |  voice  | nb/mb/wb/swb/fb |     |     |  x  |  x  |   x  |   x  |
    |         |                 |     |     |     |     |      |      |
    |  audio  |   nb/wb/swb/fb  |  x  |  x  |  x  |  x  |      |      |
    +---------+-----------------+-----+-----+-----+-----+------+------+

       Table 3: Supported Opus frame  sizes and timestamp increments








































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5.  Congestion Control

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

   The target bitrate 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.



































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6.  IANA Considerations

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

6.1.  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:

   maxplaybackrate:  a hint about the maximum output sampling rate that
      the receiver is capable of rendering in Hz.  The decoder MUST be
      capable of decoding any audio bandwidth but due to hardware
      limitations only signals up to the specified sampling rate can be
      played back.  Sending signals with higher audio bandwidth results
      in higher than necessary network usage and encoding complexity, so
      an encoder SHOULD NOT encode frequencies above the audio bandwidth
      specified by maxplaybackrate.  This parameter can take any value
      between 8000 and 48000, although commonly the value will match one
      of the Opus bandwidths (Table 1).  By default, the receiver is
      assumed to have no limitations, i.e. 48000.

   sprop-maxcapturerate:  a hint about the maximum input sampling rate
      that the sender is likely to produce.  This is not a guarantee
      that the sender will never send any higher bandwidth (e.g. it
      could send a pre-recorded prompt that uses a higher bandwidth),
      but it indicates to the receiver that frequencies above this
      maximum can safely be discarded.  This parameter is useful to
      avoid wasting receiver resources by operating the audio processing
      pipeline (e.g. echo cancellation) at a higher rate than necessary.
      This parameter can take any value between 8000 and 48000, although
      commonly the value will match one of the Opus bandwidths
      (Table 1).  By default, the sender is assumed to have no



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      limitations, i.e. 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.  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 of accepting any allowed packet sizes to
      ensure maximum compatibility.

   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.  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 of accepting 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.  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 bitrate of
      a session in bits per second (b/s).  The actual value of the
      bitrate may vary as it is dependent on the characteristics of the
      media in a packet.  Note that the maximum average bitrate 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 Section 3.1.1 for the corresponding mode of Opus and
      corresponding maxplaybackrate: will be the default.



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   stereo:  specifies whether the decoder prefers receiving stereo or
      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
      utilisation and encoding complexity.  If no value is specified,
      mono is assumed (stereo=0).

   sprop-stereo:  specifies whether the sender is likely to produce
      stereo audio.  Possible values are 1 and 0 where 1 specifies that
      stereo signals are likely to be sent, and 0 speficies that the
      sender will likely only send mono.  This is not a guarantee that
      the sender will never send stereo audio (e.g. it could send a pre-
      recorded prompt that uses stereo), but it indicates to the
      receiver that the received signal can be safely downmixed to mono.
      This parameter is useful to avoid wasting receiver resources by
      operating the audio processing pipeline (e.g. echo cancellation)
      in stereo when not necessary.  If no value is specified, mono is
      assumed (sprop-stereo=0).

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

   useinbandfec:  specifies that the decoder has the capability to take
      advantage of the Opus in-band FEC.  Possible values are 1 and 0.
      It is RECOMMENDED to provide 0 in case FEC cannot be utilized on
      the receiving side.  If no value is specified, useinbandfec is
      assumed to be 0.  This parameter is only a preference and the
      receiver MUST be able to process packets that include FEC
      information, even if it means the FEC part is discarded.

   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:








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

   Security considerations:

      See Section 7 of this document.

   Interoperability considerations: none


   Published specification: none


   Applications that use this media type:

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

   Person & email address to contact for further information:

      SILK Support silksupport@skype.net
      Jean-Marc Valin jmvalin@jmvalin.ca

   Intended usage: COMMON


   Restrictions on usage:


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

   Author:

      Julian Spittka jspittka@gmail.com

      Koen Vos koenvos74@gmail.com

      Jean-Marc Valin jmvalin@jmvalin.ca


   Change controller: TBD

6.2.  Mapping to SDP Parameters

   The information described in the media type specification has a
   specific mapping to fields in the Session Description Protocol (SDP)



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   [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 48000 and the
      number of channels MUST be 2.
   o  The OPTIONAL media type parameters "ptime" and "maxptime" are
      mapped to "a=ptime" and "a=maxptime" attributes, respectively, in
      the SDP.
   o  The OPTIONAL media type parameters "maxaveragebitrate",
      "maxplaybackrate", "minptime", "stereo", "cbr", "useinbandfec",
      and "usedtx", when present, MUST be included in the "a=fmtp"
      attribute in the SDP, expressed as a media type string in the form
      of a semicolon-separated list of parameter=value pairs (e.g.,
      maxaveragebitrate=20000).  They MUST NOT be specified in an SSRC-
      specific "fmtp" source-level attribute (as defined in Section 6.3
      of [RFC5576]).
   o  The OPTIONAL media type parameters "sprop-maxcapturerate", and
      "sprop-stereo" MAY be mapped to the "a=fmtp" SDP attribute by
      copying them directly from the media type parameter string as part
      of the semicolon-separated list of parameter=value pairs (e.g.,
      sprop-stereo=1).  These same OPTIONAL media type parameters MAY
      also be specified using an SSRC-specific "fmtp" source-level
      attribute as described in Section 6.3 of [RFC5576].  They MAY be
      specified in both places, in which case the parameter in the
      source-level attribute overrides the one found on the "a=fmtp"
      line.  The value of any parameter which is not specified in a
      source-level source attribute MUST be taken from the "a=fmtp"
      line, if it is present there.

   Below are some examples of SDP session descriptions for Opus:

   Example 1: Standard mono session with 48000 Hz clock rate


       m=audio 54312 RTP/AVP 101
       a=rtpmap:101 opus/48000/2


   Example 2: 16000 Hz clock rate, maximum packet size of 40 ms,
   recommended packet size of 40 ms, maximum average bitrate of 20000
   bps, prefers to receive stereo but only plans to send mono, FEC is
   allowed, DTX is not allowed






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       m=audio 54312 RTP/AVP 101
       a=rtpmap:101 opus/48000/2
       a=fmtp:101 maxplaybackrate=16000; sprop-maxcapturerate=16000;
       maxaveragebitrate=20000; stereo=1; useinbandfec=1; usedtx=0
       a=ptime:40
       a=maxptime:40


   Example 3: Two-way full-band stereo preferred


       m=audio 54312 RTP/AVP 101
       a=rtpmap:101 opus/48000/2
       a=fmtp:101 stereo=1; sprop-stereo=1


6.2.1.  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/48000/2


   o  The "ptime" and "maxptime" parameters 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.
      [RFC3264] defines the SDP offer-answer handling of the "ptime"
      parameter.  The "maxptime" parameter MUST be handled in the same
      way.
   o  The "minptime" parameter 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 "maxplaybackrate" parameter is a unidirectional receive-only
      parameter that reflects limitations of the local receiver.  The
      sender of the other side SHOULD NOT send with an audio bandwidth



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      higher than "maxplaybackrate" as this would lead to inefficient
      use of network resources.  The "maxplaybackrate" parameter does
      not affect interoperability.  Also, this parameter SHOULD NOT be
      used to adjust the audio bandwidth as a function of the bitrates,
      as this is the responsibility of the Opus encoder implementation.
   o  The "maxaveragebitrate" parameter 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 bitrate
      higher than "maxaveragebitrate" as it might overload the network
      and/or receiver.  The "maxaveragebitrate" parameter 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 "sprop-maxcapturerate" and "sprop-stereo" parameters are
      unidirectional sender-only parameters that reflect limitations of
      the sender side.  They allow the receiver to set up a reduced-
      complexity audio processing pipeline if the sender is not planning
      to use the full range of Opus's capabilities.  Neither "sprop-
      maxcapturerate" nor "sprop-stereo" affect interoperability and the
      receiver MUST be capable of receiving any signal.
   o  The "stereo" parameter is a unidirectional receive-only parameter.
   o  The "cbr" parameter is a unidirectional receive-only parameter.
   o  The "useinbandfec" parameter is a unidirectional receive-only
      parameter.
   o  The "usedtx" parameter 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.

6.2.2.  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", "maxplaybackrate",
      and "maxaveragebitrate" should be selected carefully to ensure
      that a reasonable performance can be achieved for the participants
      of a session.
   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.



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7.  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.


































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8.  Acknowledgements

   TBD
















































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9.  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.

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, June 2009.

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

   [RFC6716]  Valin, JM., Vos, K., and T. Terriberry, "Definition of the
              Opus Audio Codec", RFC 6716, September 2012.





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Authors' Addresses

   Julian Spittka

   Email: jspittka@gmail.com


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

   Email: koenvos74@gmail.com


   Jean-Marc Valin
   Mozilla
   650 Castro Street
   Mountain View, CA  94041
   USA

   Email: jmvalin@jmvalin.ca




























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