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Versions: (draft-spittka-payload-rtp-opus) 00 01 02 03

Network Working Group                                         J. Spittka
Internet-Draft
Intended status: Standards Track                                  K. Vos
Expires: January 31, 2015                                        vocTone
                                                               JM. Valin
                                                                 Mozilla
                                                           July 30, 2014


           RTP Payload Format for Opus Speech and Audio Codec
                     draft-ietf-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
   necessary to integrate the codec in the most compatible way.
   Further, it describes media type registrations 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 January 31, 2015.

Copyright Notice

   Copyright (c) 2014 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  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Conventions, Definitions and Acronyms used in this document .   3
     2.1.  Audio Bandwidth . . . . . . . . . . . . . . . . . . . . .   3
   3.  Opus Codec  . . . . . . . . . . . . . . . . . . . . . . . . .   3
     3.1.  Network Bandwidth . . . . . . . . . . . . . . . . . . . .   4
       3.1.1.  Recommended Bitrate . . . . . . . . . . . . . . . . .   4
       3.1.2.  Variable versus Constant Bitrate  . . . . . . . . . .   4
       3.1.3.  Discontinuous Transmission (DTX)  . . . . . . . . . .   4
     3.2.  Complexity  . . . . . . . . . . . . . . . . . . . . . . .   5
     3.3.  Forward Error Correction (FEC)  . . . . . . . . . . . . .   5
     3.4.  Stereo Operation  . . . . . . . . . . . . . . . . . . . .   6
   4.  Opus RTP Payload Format . . . . . . . . . . . . . . . . . . .   6
     4.1.  RTP Header Usage  . . . . . . . . . . . . . . . . . . . .   6
     4.2.  Payload Structure . . . . . . . . . . . . . . . . . . . .   7
   5.  Congestion Control  . . . . . . . . . . . . . . . . . . . . .   8
   6.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   8
     6.1.  Opus Media Type Registration  . . . . . . . . . . . . . .   9
     6.2.  Mapping to SDP Parameters . . . . . . . . . . . . . . . .  12
       6.2.1.  Offer-Answer Model Considerations for Opus  . . . . .  14
       6.2.2.  Declarative SDP Considerations for Opus . . . . . . .  15
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  16
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  16
   9.  Normative References  . . . . . . . . . . . . . . . . . . . .  16
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  17

1.  Introduction

   The Opus codec is a speech and audio codec developed within the IETF
   Internet Wideband Audio Codec working group.  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 necessary to integrate the Opus codec in the most
   compatible way.  Further, it describes media type registrations 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 (per channel)
   SDP:  Session Description Protocol
   VBR:  Variable bitrate

2.1.  Audio Bandwidth

   Throughout this document, we refer to the following definitions:

   +--------------+----------------+-----------------+-----------------+
   | Abbreviation |      Name      | Audio Bandwidth |  Sampling Rate  |
   |              |                |       (Hz)      |       (Hz)      |
   +--------------+----------------+-----------------+-----------------+
   |      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

3.  Opus Codec

   The Opus [RFC6716] codec encodes speech signals as well as general
   audio signals.  Two different modes can be chosen, a voice mode or an
   audio mode, to allow the most efficient coding depending on the type
   of the input signal, the sampling frequency of the input signal, and
   the intended application.






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   The voice mode allows efficient encoding of voice signals at lower
   bit rates while the audio mode is optimized for general 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 bitrates result 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 Bitrate

   For the same average bitrate, variable bitrate (VBR) can achieve
   higher quality than constant bitrate (CBR).  For the majority of
   voice transmission applications, VBR is the best choice.  One reason
   for choosing CBR is the potential information leak that _might_ 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 NOT exceed the available
   network capacity.  If no target bitrate is specified, the bitrates
   specified in Section 3.1.1 are RECOMMENDED.

3.1.3.  Discontinuous Transmission (DTX)

   The Opus codec can, as described in Section 3.1.2, be operated with a
   variable bitrate.  In that case, the encoder will automatically
   reduce the bitrate for certain input signals, like periods of



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   silence.  When using continuous transmission, it will reduce the
   bitrate when the characteristics of the input signal permit, but will
   never interrupt the transmission to the receiver.  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 can 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.  A
   receiver can distinguish between DTX and packet loss by looking for
   gaps in the sequence number, as described by Section 4.1
   of [RFC3551].

   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.  Use of [RFC3389] Comfort Noise (CN) with
   Opus is discouraged.  The transmitter MUST drop whole frames only,
   based on the size of the last transmitted frame, to ensure successive
   RTP timestamps differ by a multiple of 120 and to allow the receiver
   to use whole frames for concealment.

   DTX can be used with both variable and constant bitrate.  It will
   have a slightly lower speech or audio quality than continuous
   transmission.  Therefore, using continuous transmission is
   RECOMMENDED unless restraints on network capacity are severe.

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 embedding "in-band" forward error
   correction (FEC) data into the Opus bit stream.  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 the encoder



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   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 it loses a packet and 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 receiver can then
   configure its decoder to decode the FEC data from the packet rather
   than the regular audio data.  If no FEC data is available for the
   current frame, the decoder will consider the frame lost and invoke
   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
   needed 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 network resources.

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 use of
   the fields of the RTP header by the Opus payload format is consistent
   with that specification.

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

   The timestamp, sequence number, and marker bit (M) of the RTP header
   are used in accordance with Section 4.1 of [RFC3551].




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   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 duplicate RTP packets.
   The receiver MUST provide at most one of those payloads to the Opus
   decoder for decoding, and MUST discard the others.

   Opus supports 5 different audio bandwidths, which can be adjusted
   during a call.  The RTP timestamp is incremented with a 48000 Hz
   clock rate for all modes of Opus and all sampling rates.  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 data encoded with sampling rates other
   than 48000 Hz, the sampling rate has to be adjusted to 48000 Hz using
   the corresponding multiplier in Table 2.

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

                       Table 2: Timestamp multiplier

4.2.  Payload Structure

   The Opus encoder can 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, up to a maximum
   packet duration representing 120 ms of speech or audio data.  The
   grouping of one or more Opus frames into a single Opus packet is
   defined in Section 3 of [RFC6716].  An RTP payload MUST contain
   exactly one Opus packet as defined by that document.

   Figure 1 shows the structure combined with the RTP header.








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                        +----------+--------------+
                        |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 the speech and audio modes (Mode) and sampling
   rates (fs) of Opus and shows how the timestamp is incremented for
   packetization (ts incr).  If the Opus encoder outputs multiple
   encoded frames into a single packet, the timestamp increment is the
   sum of the increments for the individual frames.

    +---------+-----------------+-----+-----+-----+-----+------+------+
    |   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

5.  Congestion Control

   The target bitrate of Opus can be adjusted at any point in time, thus
   allowing 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 the packet duration.  A lower packet transmission
   rate reduces the amount of header overhead, but at the same time
   increases latency and loss sensitivity, so it ought to be used with
   care.

   It is RECOMMENDED that senders of Opus encoded data apply congestion
   control.

6.  IANA Considerations

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







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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:  the RTP timestamp is incremented with a 48000 Hz clock rate
      for all modes of Opus and all sampling rates.  For data encoded
      with sampling rates other than 48000 Hz, the sampling rate has to
      be adjusted to 48000 Hz using the corresponding multiplier in
      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
      limitations, i.e. 48000.





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   maxptime:  the maximum duration of media represented by a packet
      (according to Section 6 of [RFC4566]) that a decoder wants to
      receive, in milliseconds rounded up to the next full integer
      value.  Possible values are 3, 5, 10, 20, 40, 60, or an arbitrary
      multiple of an Opus frame size 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, the default is 120.  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 preferred duration of media represented by a packet
      (according to Section 6 of [RFC4566]) that a decoder wants to
      receive, in milliseconds rounded up to the next full integer
      value.  Possible values are 3, 5, 10, 20, 40, 60, or an arbitrary
      multiple of an Opus frame size 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, the default is 20.  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 minimum duration of media represented by a packet
      (according to Section 6 of [RFC4566]) that SHOULD be encapsulated
      in a received packet, in milliseconds rounded up to the next full
      integer value.  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, the default is 3.  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 can 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 6000 to 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 is the default.

   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



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      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
      utilization and encoding complexity.  If no value is specified,
      the default is 0 (mono).


   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 specifies 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, the
      default is 0 (mono).


   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, the default is 0 (vbr).  When cbr is 1, the
      maximum average bitrate can still change, 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.
      Providing 0 when FEC cannot be used on the receiving side is
      RECOMMENDED.  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, the default is 0.


   Encoding considerations:


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

   Security considerations:



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      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 can 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)
   [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.



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   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
   desired, DTX is not desired











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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
      restricted by this payload format description.  The decoder MUST
      be capable of decoding 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, some values might cause application
      performance to 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, some values might cause application performance to suffer
      and ought to be used with care.
   o  The "maxplaybackrate" parameter is a unidirectional receive-only
      parameter that reflects limitations of the local receiver.  When
      sending to a single destination, a sender MUST NOT use an audio
      bandwidth higher than necessary to make full use of audio sampled
      at a sampling rate of "maxplaybackrate".  Gateways or senders that
      are sending the same encoded audio to multiple destinations SHOULD



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      NOT use an audio bandwidth higher than necessary to represent
      audio sampled at "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 bitrate, 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, some values might cause
      application performance to suffer, and ought to 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.
      When sending to a single destination, a sender MUST NOT use stereo
      when "stereo" is 0.  Gateways or senders that are sending the same
      encoded audio to multiple destinations SHOULD NOT use stereo when
      "stereo" is 0, as this would lead to inefficient use of network
      resources.  The "stereo" parameter does not affect
      interoperability.
   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" ought to 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



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      capable of accepting 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 can be
      provided if necessary by declaring multiple RTP payload types;
      however, the number of types ought to be kept small.

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.

8.  Acknowledgements

   TBD

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.

   [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
              Comfort Noise (CN)", RFC 3389, September 2002.




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   [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", 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.

Authors' Addresses

   Julian Spittka

   Email: jspittka@gmail.com


   Koen Vos
   vocTone

   Email: koenvos74@gmail.com








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   Jean-Marc Valin
   Mozilla
   331 E. Evelyn Avenue
   Mountain View, CA  94041
   USA

   Email: jmvalin@jmvalin.ca












































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