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rfc:rfc7587

Internet Engineering Task Force (IETF) J. Spittka Request for Comments: 7587 Category: Standards Track K. Vos ISSN: 2070-1721 vocTone

                                                             JM. Valin
                                                               Mozilla
                                                             June 2015
       RTP Payload Format for the Opus Speech and Audio Codec

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.  It also
 provides an applicability statement for the use of Opus over RTP.
 Further, it describes media type registrations for the RTP payload
 format.

Status of This Memo

 This is an Internet Standards Track document.
 This document is a product of the Internet Engineering Task Force
 (IETF).  It represents the consensus of the IETF community.  It has
 received public review and has been approved for publication by the
 Internet Engineering Steering Group (IESG).  Further information on
 Internet Standards is available in Section 2 of RFC 5741.
 Information about the current status of this document, any errata,
 and how to provide feedback on it may be obtained at
 http://www.rfc-editor.org/info/rfc7587.

Copyright Notice

 Copyright (c) 2015 IETF Trust and the persons identified as the
 document authors.  All rights reserved.
 This document is subject to BCP 78 and the IETF Trust's Legal
 Provisions Relating to IETF Documents
 (http://trustee.ietf.org/license-info) in effect on the date of
 publication of this document.  Please review these documents
 carefully, as they describe your rights and restrictions with respect
 to this document.  Code Components extracted from this document must
 include Simplified BSD License text as described in Section 4.e of
 the Trust Legal Provisions and are provided without warranty as
 described in the Simplified BSD License.

Spittka, et al. Standards Track [Page 1] RFC 7587 RTP Payload Format for Opus June 2015

Table of Contents

 1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
 2.  Conventions, Definitions, and Acronyms Used in This Document    3
 3.  Opus Codec  . . . . . . . . . . . . . . . . . . . . . . . . .   4
   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)  . . . . . . . . . .   5
   3.2.  Complexity  . . . . . . . . . . . . . . . . . . . . . . .   6
   3.3.  Forward Error Correction (FEC)  . . . . . . . . . . . . .   6
   3.4.  Stereo Operation  . . . . . . . . . . . . . . . . . . . .   6
 4.  Opus RTP Payload Format . . . . . . . . . . . . . . . . . . .   7
   4.1.  RTP Header Usage  . . . . . . . . . . . . . . . . . . . .   7
   4.2.  Payload Structure . . . . . . . . . . . . . . . . . . . .   7
 5.  Congestion Control  . . . . . . . . . . . . . . . . . . . . .   8
 6.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   9
   6.1.  Opus Media Type Registration  . . . . . . . . . . . . . .   9
 7.  SDP Considerations  . . . . . . . . . . . . . . . . . . . . .  12
   7.1.  SDP Offer/Answer Considerations . . . . . . . . . . . . .  13
   7.2.  Declarative SDP Considerations for Opus . . . . . . . . .  15
 8.  Security Considerations . . . . . . . . . . . . . . . . . . .  15
 9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  16
   9.1.  Normative References  . . . . . . . . . . . . . . . . . .  16
   9.2.  Informative References  . . . . . . . . . . . . . . . . .  17
 Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .  18
 Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  18

1. Introduction

 Opus [RFC6716] 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 Opus in the most compatible way.
 It also provides an applicability statement for the use of Opus over
 RTP.  Further, it describes media type registrations for the RTP
 payload format.

Spittka, et al. Standards Track [Page 2] RFC 7587 RTP Payload Format for Opus June 2015

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].
 audio bandwidth:  The range of audio frequencies being coded
 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
 SSRC:  Synchronization source
 VBR:  Variable bitrate
 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      |
 +--------------+----------------+-----------------+-----------------+
                    Table 1: Audio Bandwidth Naming

Spittka, et al. Standards Track [Page 3] RFC 7587 RTP Payload Format for Opus June 2015

3. Opus Codec

 Opus 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.
 The voice mode allows efficient encoding of voice signals at lower
 bitrates while the audio mode is optimized for general audio signals
 at medium and higher bitrates.
 Opus is highly scalable in terms of audio bandwidth, bitrate, and
 complexity.  Further, Opus allows transmitting stereo signals with
 in-band signaling in the bitstream.

3.1. Network Bandwidth

 Opus supports bitrates from 6 kbit/s to 510 kbit/s.  The bitrate can
 be changed dynamically within that range.  All other parameters being
 equal, higher bitrates result in higher audio 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 kbit/s for NB speech,
 o  16-20 kbit/s for WB speech,
 o  28-40 kbit/s for FB speech,
 o  48-64 kbit/s for FB mono music, and
 o  64-128 kbit/s for FB stereo music.

3.1.2. Variable versus Constant Bitrate

 For the same average bitrate, variable bitrate (VBR) can achieve
 higher audio 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, the Opus padding mechanism

Spittka, et al. Standards Track [Page 4] RFC 7587 RTP Payload Format for Opus June 2015

 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 bandwidth.  If no target bitrate is specified, the bitrates
 specified in Section 3.1.1 are RECOMMENDED.

3.1.3. Discontinuous Transmission (DTX)

 Opus 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 silence.  When
 using continuous transmission, it will reduce the bitrate when the
 characteristics of the input signal permit, but it will never
 interrupt the transmission to the receiver.  Therefore, the received
 signal will maintain the same high level of audio quality over the
 full duration of a transmission while minimizing the average bitrate
 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.  Using Comfort Noise as defined in [RFC3389]
 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 constraints on available network bandwidth are
 severe.

Spittka, et al. Standards Track [Page 5] RFC 7587 RTP Payload Format for Opus June 2015

3.2. Complexity

 Complexity of the encoder 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 bitstream.  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; and (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;
 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.  Instead of performing
 loss concealment for a missing packet, the receiver can then
 configure its decoder to decode the FEC data from the next packet.
 Any compliant Opus decoder is capable of ignoring FEC information
 when it is not needed, so encoding with FEC cannot cause
 interoperability problems.  However, if FEC cannot be used on the
 receiving side, then 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 bitstream and no special arrangement
 is needed in the payload format.  An Opus decoder is capable of
 handling a stereo encoding, but an application might only be capable
 of consuming a single audio channel.
 If a decoder cannot 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.

Spittka, et al. Standards Track [Page 6] RFC 7587 RTP Payload Format for Opus June 2015

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; therefore,
 no padding is necessary.  The payload MAY be padded by an integer
 number of octets according to [RFC3550], although the Opus internal
 padding is preferred.
 The timestamp, sequence number, and marker bit (M) of the RTP header
 are used in accordance with Section 4.1 of [RFC3551].
 The RTP payload type for Opus is 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 it MUST discard the others.
 Opus supports 5 different audio bandwidths, which can be adjusted
 during a stream.  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.

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.

Spittka, et al. Standards Track [Page 7] RFC 7587 RTP Payload Format for Opus June 2015

                      +----------+--------------+
                      |RTP Header| Opus Payload |
                      +----------+--------------+
              Figure 1: Packet Structure with RTP Header
 Table 2 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 it 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  |  o  |  o  |  o   |  o   |
  |         |                 |     |     |     |     |      |      |
  |  audio  |   NB/WB/SWB/FB  |  o  |  o  |  o  |  o  |  x   |  x   |
  +---------+-----------------+-----+-----+-----+-----+------+------+
   Table 2: Supported Opus frame sizes and timestamp increments are
       marked with an o.  Unsupported ones are marked with an x.

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.
 Since UDP does not provide congestion control, applications that use
 RTP over UDP SHOULD implement their own congestion control above the
 UDP layer [RFC5405].  Work in the RMCAT working group [rmcat]
 describes the interactions and conceptual interfaces necessary
 between the application components that relate to congestion control,
 including the RTP layer, the higher-level media codec control layer,
 and the lower-level transport interface, as well as components
 dedicated to congestion control functions.

Spittka, et al. Standards Track [Page 8] RFC 7587 RTP Payload Format for Opus June 2015

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 [RFC6838] 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.
 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 prerecorded 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.

Spittka, et al. Standards Track [Page 9] RFC 7587 RTP Payload Format for Opus June 2015

 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.
 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.
 maxaveragebitrate:  specifies the maximum average receive bitrate of
    a session in bits per second (bit/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
    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
    prerecorded 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).

Spittka, et al. Standards Track [Page 10] RFC 7587 RTP Payload Format for Opus June 2015

 cbr:  specifies if the decoder prefers the use of a constant bitrate
    versus a 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 of [RFC6838].
 Security considerations:
    See Section 8 of this document.
 Interoperability considerations: none
 Published specification: RFC 7587
 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, and media
    streaming.
 Fragment identifier considerations: N/A
 Person & email address to contact for further information:
    SILK Support, silksupport@skype.net
    Jean-Marc Valin, jmvalin@jmvalin.ca
 Intended usage: COMMON

Spittka, et al. Standards Track [Page 11] RFC 7587 RTP Payload Format for Opus June 2015

 Restrictions on usage:
    For transfer over RTP, the RTP payload format (Section 4 of this
    document) SHALL be used.
 Authors:
    Julian Spittka, jspittka@gmail.com
    Koen Vos, koenvos74@gmail.com
    Jean-Marc Valin, jmvalin@jmvalin.ca
 Change controller: IETF Payload working group delegated from the IESG

7. SDP Considerations

 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 Opus, 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", "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.,
    maxplaybackrate=48000).  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

Spittka, et al. Standards Track [Page 12] RFC 7587 RTP Payload Format for Opus June 2015

    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 that 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
 bit/s, prefers to receive stereo but only plans to send mono, FEC is
 desired, DTX is not desired
     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

7.1. SDP Offer/Answer Considerations

 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

Spittka, et al. Standards Track [Page 13] RFC 7587 RTP Payload Format for Opus June 2015

 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 "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
    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.

Spittka, et al. Standards Track [Page 14] RFC 7587 RTP Payload Format for Opus June 2015

 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.
 The Opus parameters in an SDP offer/answer exchange are completely
 orthogonal, and there is no relationship between the SDP offer and
 the answer.

7.2. Declarative SDP Considerations for Opus

 For declarative use of SDP such as in the Session Announcement
 Protocol (SAP) [RFC2974] and the Real Time Streaming Protocol (RTSP)
 [RFC2326] for Opus, the following needs to be considered:
 o  The values for "maxptime", "ptime", "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 of the payload format
    configuration are recommendations by the decoding side to ensure
    the best performance for the decoder.
 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.

8. Security Considerations

 Use of VBR is subject to the security considerations in [RFC6562].
 RTP packets using the payload format defined in this specification
 are subject to the security considerations discussed in the RTP
 specification [RFC3550] and in any applicable RTP profile such as
 RTP/AVP [RFC3551], RTP/AVPF [RFC4585], RTP/SAVP [RFC3711], or RTP/
 SAVPF [RFC5124].  However, as "Securing the RTP Framework: Why RTP
 Does Not Mandate a Single Media Security Solution" [RFC7202]
 discusses, it is not an RTP payload format's responsibility to
 discuss or mandate what solutions are used to meet the basic security
 goals like confidentiality, integrity, and source authenticity for
 RTP in general.  This responsibility lies on anyone using RTP in an
 application.  They can find guidance on available security mechanisms

Spittka, et al. Standards Track [Page 15] RFC 7587 RTP Payload Format for Opus June 2015

 and important considerations in "Options for Securing RTP Sessions"
 [RFC7201].  Applications SHOULD use one or more appropriate strong
 security mechanisms.
 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.

9. References

9.1. Normative References

 [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
            Requirement Levels", BCP 14, RFC 2119,
            DOI 10.17487/RFC2119, March 1997,
            <http://www.rfc-editor.org/info/rfc2119>.
 [RFC2326]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
            Streaming Protocol (RTSP)", RFC 2326,
            DOI 10.17487/RFC2326, April 1998,
            <http://www.rfc-editor.org/info/rfc2326>.
 [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
            with Session Description Protocol (SDP)", RFC 3264,
            DOI 10.17487/RFC3264, June 2002,
            <http://www.rfc-editor.org/info/rfc3264>.
 [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
            Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389,
            September 2002, <http://www.rfc-editor.org/info/rfc3389>.
 [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
            Jacobson, "RTP: A Transport Protocol for Real-Time
            Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
            July 2003, <http://www.rfc-editor.org/info/rfc3550>.
 [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
            Video Conferences with Minimal Control", STD 65, RFC 3551,
            DOI 10.17487/RFC3551, July 2003,
            <http://www.rfc-editor.org/info/rfc3551>.
 [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
            Norrman, "The Secure Real-time Transport Protocol (SRTP)",
            RFC 3711, DOI 10.17487/RFC3711, March 2004,
            <http://www.rfc-editor.org/info/rfc3711>.

Spittka, et al. Standards Track [Page 16] RFC 7587 RTP Payload Format for Opus June 2015

 [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
            Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
            July 2006, <http://www.rfc-editor.org/info/rfc4566>.
 [RFC4855]  Casner, S., "Media Type Registration of RTP Payload
            Formats", RFC 4855, DOI 10.17487/RFC4855, February 2007,
            <http://www.rfc-editor.org/info/rfc4855>.
 [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
            Media Attributes in the Session Description Protocol
            (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
            <http://www.rfc-editor.org/info/rfc5576>.
 [RFC6562]  Perkins, C. and JM. Valin, "Guidelines for the Use of
            Variable Bit Rate Audio with Secure RTP", RFC 6562,
            DOI 10.17487/RFC6562, March 2012,
            <http://www.rfc-editor.org/info/rfc6562>.
 [RFC6716]  Valin, JM., Vos, K., and T. Terriberry, "Definition of the
            Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
            September 2012, <http://www.rfc-editor.org/info/rfc6716>.
 [RFC6838]  Freed, N., Klensin, J., and T. Hansen, "Media Type
            Specifications and Registration Procedures", BCP 13,
            RFC 6838, DOI 10.17487/RFC6838, January 2013,
            <http://www.rfc-editor.org/info/rfc6838>.

9.2. Informative References

 [RFC2974]  Handley, M., Perkins, C., and E. Whelan, "Session
            Announcement Protocol", RFC 2974, DOI 10.17487/RFC2974,
            October 2000, <http://www.rfc-editor.org/info/rfc2974>.
 [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
            "Extended RTP Profile for Real-time Transport Control
            Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
            DOI 10.17487/RFC4585, July 2006,
            <http://www.rfc-editor.org/info/rfc4585>.
 [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
            Real-time Transport Control Protocol (RTCP)-Based Feedback
            (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
            2008, <http://www.rfc-editor.org/info/rfc5124>.

Spittka, et al. Standards Track [Page 17] RFC 7587 RTP Payload Format for Opus June 2015

 [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
            for Application Designers", BCP 145, RFC 5405,
            DOI 10.17487/RFC5405, November 2008,
            <http://www.rfc-editor.org/info/rfc5405>.
 [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
            Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
            <http://www.rfc-editor.org/info/rfc7201>.
 [RFC7202]  Perkins, C. and M. Westerlund, "Securing the RTP
            Framework: Why RTP Does Not Mandate a Single Media
            Security Solution", RFC 7202, DOI 10.17487/RFC7202, April
            2014, <http://www.rfc-editor.org/info/rfc7202>.
 [rmcat]    "RTP Media Congestion Avoidance Techniques (rmcat)
            Documents", <https://datatracker.ietf.org/wg/rmcat/
            documents/>.

Acknowledgements

 Many people have made useful comments and suggestions contributing to
 this document.  In particular, we would like to thank Tina le Grand,
 Cullen Jennings, Jonathan Lennox, Gregory Maxwell, Colin Perkins, Jan
 Skoglund, Timothy B. Terriberry, Martin Thompson, Justin Uberti,
 Magnus Westerlund, and Mo Zanaty.

Authors' Addresses

 Julian Spittka
 Email: jspittka@gmail.com
 Koen Vos
 vocTone
 Email: koenvos74@gmail.com
 Jean-Marc Valin
 Mozilla
 331 E. Evelyn Avenue
 Mountain View, CA  94041
 United States
 Email: jmvalin@jmvalin.ca

Spittka, et al. Standards Track [Page 18]

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