| Rfc | 7587 |
|---|
| Title | RTP Payload Format for the Opus Speech and Audio Codec |
|---|
| Author | J. Spittka,
K. Vos, JM. Valin |
|---|
| Date | June 2015 |
|---|
| Format: | TXT, HTML |
|---|
| Status: | PROPOSED
STANDARD |
|---|
|
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.
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.
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
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
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.
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.
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.
+----------+--------------+
|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.
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.
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).
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
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
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
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.
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
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>.
[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>.
[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