Network Working Group J. Spittka
Internet-Draft
Intended status: Standards Track K. Vos
Expires: July 14, 2013 Skype Technologies S.A.
JM. Valin
Mozilla
January 10, 2013
RTP Payload Format for Opus Speech and Audio Codec
draft-ietf-payload-rtp-opus-00
Abstract
This document defines the Real-time Transport Protocol (RTP) payload
format for packetization of Opus encoded speech and audio data that
is essential to integrate the codec in the most compatible way.
Further, media type registrations are described for the RTP payload
format.
Status of this Memo
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Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved.
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described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Conventions, Definitions and Acronyms used in this document . 4
2.1. Audio Bandwidth . . . . . . . . . . . . . . . . . . . . . 4
3. Opus Codec . . . . . . . . . . . . . . . . . . . . . . . . . . 5
3.1. Network Bandwidth . . . . . . . . . . . . . . . . . . . . 5
3.1.1. Recommended Bitrate . . . . . . . . . . . . . . . . . 5
3.1.2. Variable versus Constant Bit Rate . . . . . . . . . . 5
3.1.3. Discontinuous Transmission (DTX) . . . . . . . . . . . 6
3.2. Complexity . . . . . . . . . . . . . . . . . . . . . . . . 6
3.3. Forward Error Correction (FEC) . . . . . . . . . . . . . . 6
3.4. Stereo Operation . . . . . . . . . . . . . . . . . . . . . 7
4. Opus RTP Payload Format . . . . . . . . . . . . . . . . . . . 8
4.1. RTP Header Usage . . . . . . . . . . . . . . . . . . . . . 8
4.2. Payload Structure . . . . . . . . . . . . . . . . . . . . 9
5. Congestion Control . . . . . . . . . . . . . . . . . . . . . . 11
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12
6.1. Opus Media Type Registration . . . . . . . . . . . . . . . 12
6.2. Mapping to SDP Parameters . . . . . . . . . . . . . . . . 15
6.2.1. Offer-Answer Model Considerations for Opus . . . . . . 17
6.2.2. Declarative SDP Considerations for Opus . . . . . . . 18
7. Security Considerations . . . . . . . . . . . . . . . . . . . 19
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 20
9. Normative References . . . . . . . . . . . . . . . . . . . . . 21
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22
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1. Introduction
The Opus codec is a speech and audio codec developed within the IETF
Internet Wideband Audio Codec working group (codec). The codec has a
very low algorithmic delay and it is highly scalable in terms of
audio bandwidth, bitrate, and complexity. Further, it provides
different modes to efficiently encode speech signals as well as music
signals, thus, making it the codec of choice for various applications
using the Internet or similar networks.
This document defines the Real-time Transport Protocol (RTP)
[RFC3550] payload format for packetization of Opus encoded speech and
audio data that is essential to integrate the Opus codec in the most
compatible way. Further, media type registrations are described for
the RTP payload format. More information on the Opus codec can be
obtained from [RFC6716].
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2. Conventions, Definitions and Acronyms used in this document
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
CBR: Constant bitrate
CPU: Central Processing Unit
DTX: Discontinuous transmission
FEC: Forward error correction
IP: Internet Protocol
samples: Speech or audio samples (usually per channel)
SDP: Session Description Protocol
VBR: Variable bitrate
2.1. Audio Bandwidth
Throughout this document, we refer to the following definitions:
+--------------+----------------+-----------+----------+
| Abbreviation | Name | Bandwidth | Sampling |
+--------------+----------------+-----------+----------+
| nb | Narrowband | 0 - 4000 | 8000 |
| | | | |
| mb | Mediumband | 0 - 6000 | 12000 |
| | | | |
| wb | Wideband | 0 - 8000 | 16000 |
| | | | |
| swb | Super-wideband | 0 - 12000 | 24000 |
| | | | |
| fb | Fullband | 0 - 20000 | 48000 |
+--------------+----------------+-----------+----------+
Audio bandwidth naming
Table 1
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3. Opus Codec
The Opus [RFC6716] speech and audio codec has been developed to
encode speech signals as well as audio signals. Two different modes,
a voice mode or an audio mode, may be chosen to allow the most
efficient coding dependent on the type of input signal, the sampling
frequency of the input signal, and the specific application.
The voice mode allows efficient encoding of voice signals at lower
bit rates while the audio mode is optimized for audio signals at
medium and higher bitrates.
The Opus speech and audio codec is highly scalable in terms of audio
bandwidth, bitrate, and complexity. Further, Opus allows
transmitting stereo signals.
3.1. Network Bandwidth
Opus supports all bitrates from 6 kb/s to 510 kb/s. The bitrate can
be changed dynamically within that range. All other parameters being
equal, higher bitrate results in higher quality.
3.1.1. Recommended Bitrate
For a frame size of 20 ms, these are the bitrate "sweet spots" for
Opus in various configurations:
o 8-12 kb/s for NB speech,
o 16-20 kb/s for WB speech,
o 28-40 kb/s for FB speech,
o 48-64 kb/s for FB mono music, and
o 64-128 kb/s for FB stereo music.
3.1.2. Variable versus Constant Bit Rate
For the same average bitrate, variable bitrate (VBR) can achieve
higher quality than constant bitrate (CBR). For the majority of
voice transmission application, VBR is the best choice. One
potential reason for choosing CBR is the potential information leak
that _may_ occur when encrypting the compressed stream. See
[RFC6562] for guidelines on when VBR is appropriate for encrypted
audio communications. In the case where an existing VBR stream needs
to be converted to CBR for security reasons, then the Opus padding
mechanism described in [RFC6716] is the RECOMMENDED way to achieve
padding because the RTP padding bit is unencrypted.
The bitrate can be adjusted at any point in time. To avoid
congestion, the average bitrate SHOULD be adjusted to the available
network capacity. If no target bitrate is specified, the bitrates
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specified in Section 3.1.1 are RECOMMENDED.
3.1.3. Discontinuous Transmission (DTX)
The Opus codec may, as described in Section 3.1.2, be operated with
an adaptive bitrate. In that case, the bitrate will automatically be
reduced for certain input signals like periods of silence. During
continuous transmission the bitrate will be reduced, when the input
signal allows to do so, but the transmission to the receiver itself
will never be interrupted. Therefore, the received signal will
maintain the same high level of quality over the full duration of a
transmission while minimizing the average bit rate over time.
In cases where the bitrate of Opus needs to be reduced even further
or in cases where only constant bitrate is available, the Opus
encoder may be set to use discontinuous transmission (DTX), where
parts of the encoded signal that correspond to periods of silence in
the input speech or audio signal are not transmitted to the receiver.
On the receiving side, the non-transmitted parts will be handled by a
frame loss concealment unit in the Opus decoder which generates a
comfort noise signal to replace the non transmitted parts of the
speech or audio signal.
The DTX mode of Opus will have a slightly lower speech or audio
quality than the continuous mode. Therefore, it is RECOMMENDED to
use Opus in the continuous mode unless restraints on network capacity
are severe. The DTX mode can be engaged for operation in both
adaptive or constant bitrate.
3.2. Complexity
Complexity can be scaled to optimize for CPU resources in real-time,
mostly as a trade-off between audio quality and bitrate. Also,
different modes of Opus have different complexity.
3.3. Forward Error Correction (FEC)
The voice mode of Opus allows for "in-band" forward error correction
(FEC) data to be embedded into the bit stream of Opus. This FEC
scheme adds redundant information about the previous packet (n-1) to
the current output packet n. For each frame, the encoder decides
whether to use FEC based on (1) an externally-provided estimate of
the channel's packet loss rate; (2) an externally-provided estimate
of the channel's capacity; (3) the sensitivity of the audio or speech
signal to packet loss; (4) whether the receiving decoder has
indicated it can take advantage of "in-band" FEC information. The
decision to send "in-band" FEC information is entirely controlled by
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the encoder and therefore no special precautions for the payload have
to be taken.
On the receiving side, the decoder can take advantage of this
additional information when, in case of a packet loss, the next
packet is available. In order to use the FEC data, the jitter buffer
needs to provide access to payloads with the FEC data. The decoder
API function has a flag to indicate that a FEC frame rather than a
regular frame should be decoded. If no FEC data is available for the
current frame, the decoder will consider the frame lost and invokes
the frame loss concealment.
If the FEC scheme is not implemented on the receiving side, FEC
SHOULD NOT be used, as it leads to an inefficient usage of network
resources. Decoder support for FEC SHOULD be indicated at the time a
session is set up.
3.4. Stereo Operation
Opus allows for transmission of stereo audio signals. This operation
is signaled in-band in the Opus payload and no special arrangement is
required in the payload format. Any implementation of the Opus
decoder MUST be capable of receiving stereo signals, although it MAY
decode those signals as mono.
If a decoder can not take advantage of the benefits of a stereo
signal this SHOULD be indicated at the time a session is set up. In
that case the sending side SHOULD NOT send stereo signals as it leads
to an inefficient usage of the network.
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4. Opus RTP Payload Format
The payload format for Opus consists of the RTP header and Opus
payload data.
4.1. RTP Header Usage
The format of the RTP header is specified in [RFC3550]. The Opus
payload format uses the fields of the RTP header consistent with this
specification.
The payload length of Opus is a multiple number of octets and
therefore no padding is required. The payload MAY be padded by an
integer number of octets according to [RFC3550].
The marker bit (M) of the RTP header is used in accordance with
Section 4.1 of [RFC3551].
The RTP payload type for Opus has not been assigned statically and is
expected to be assigned dynamically.
The receiving side MUST be prepared to receive duplicates of RTP
packets. Only one of those payloads MUST be provided to the Opus
decoder for decoding and others MUST be discarded.
Opus supports 5 different audio bandwidths which may be adjusted
during the duration of a call. The RTP timestamp clock frequency is
defined as the highest supported sampling frequency of Opus, i.e.
48000 Hz, for all modes and sampling rates of Opus. The unit for the
timestamp is samples per single (mono) channel. The RTP timestamp
corresponds to the sample time of the first encoded sample in the
encoded frame. For sampling rates lower than 48000 Hz the number of
samples has to be multiplied with a multiplier according to Table 2
to determine the RTP timestamp.
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+---------+------------+
| fs (Hz) | Multiplier |
+---------+------------+
| 8000 | 6 |
| | |
| 12000 | 4 |
| | |
| 16000 | 3 |
| | |
| 24000 | 2 |
| | |
| 48000 | 1 |
+---------+------------+
Table 2: Timestamp multiplier
4.2. Payload Structure
The Opus encoder can be set to output encoded frames representing
2.5, 5, 10, 20, 40, or 60 ms of speech or audio data. Further, an
arbitrary number of frames can be combined into a packet. The
maximum packet length is limited to the amount of encoded data
representing 120 ms of speech or audio data. The packetization of
encoded data is purely done by the Opus encoder and therefore only
one packet output from the Opus encoder MUST be used as a payload.
Figure 1 shows the structure combined with the RTP header.
+----------+--------------+
|RTP Header| Opus Payload |
+----------+--------------+
Figure 1: Payload Structure with RTP header
Table 3 shows supported frame sizes in milliseconds of encoded speech
or audio data for speech and audio mode (Mode) and sampling rates
(fs) of Opus and how the timestamp needs to be incremented for
packetization (ts incr). If the Opus encoder outputs multiple
encoded frames into a single packet the timestamps have to be added
up according to the combined frames.
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+---------+-----------------+-----+-----+-----+-----+------+------+
| Mode | fs | 2.5 | 5 | 10 | 20 | 40 | 60 |
+---------+-----------------+-----+-----+-----+-----+------+------+
| ts incr | all | 120 | 240 | 480 | 960 | 1920 | 2880 |
| | | | | | | | |
| voice | nb/mb/wb/swb/fb | | | x | x | x | x |
| | | | | | | | |
| audio | nb/wb/swb/fb | x | x | x | x | | |
+---------+-----------------+-----+-----+-----+-----+------+------+
Table 3: Supported Opus frame sizes and timestamp increments
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5. Congestion Control
The adaptive nature of the Opus codec allows for an efficient
congestion control.
The target bitrate of Opus can be adjusted at any point in time and
thus allowing for an efficient congestion control. Furthermore, the
amount of encoded speech or audio data encoded in a single packet can
be used for congestion control since the transmission rate is
inversely proportional to these frame sizes. A lower packet
transmission rate reduces the amount of header overhead but at the
same time increases latency and error sensitivity and should be done
with care.
It is RECOMMENDED that congestion control is applied during the
transmission of Opus encoded data.
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6. IANA Considerations
One media subtype (audio/opus) has been defined and registered as
described in the following section.
6.1. Opus Media Type Registration
Media type registration is done according to [RFC4288] and [RFC4855].
Type name: audio
Subtype name: opus
Required parameters:
rate: RTP timestamp clock rate is incremented with 48000 Hz clock
rate for all modes of Opus and all sampling frequencies. For
audio sampling rates other than 48000 Hz the rate has to be
adjusted to 48000 Hz according to Table 2.
Optional parameters:
maxplaybackrate: a hint about the maximum output sampling rate that
the receiver is capable of rendering in Hz. The decoder MUST be
capable of decoding any audio bandwidth but due to hardware
limitations only signals up to the specified sampling rate can be
played back. Sending signals with higher audio bandwidth results
in higher than necessary network usage and encoding complexity, so
an encoder SHOULD NOT encode frequencies above the audio bandwidth
specified by maxplaybackrate. This parameter can take any value
between 8000 and 48000, although commonly the value will match one
of the Opus bandwidths (Table 1). By default, the receiver is
assumed to have no limitations, i.e. 48000.
sprop-maxcapturerate: a hint about the maximum input sampling rate
that the sender is likely to produce. This is not a guarantee
that the sender will never send any higher bandwidth (e.g. it
could send a pre-recorded prompt that uses a higher bandwidth),
but it indicates to the receiver that frequencies above this
maximum can safely be discarded. This parameter is useful to
avoid wasting receiver resources by operating the audio processing
pipeline (e.g. echo cancellation) at a higher rate than necessary.
This parameter can take any value between 8000 and 48000, although
commonly the value will match one of the Opus bandwidths
(Table 1). By default, the sender is assumed to have no
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limitations, i.e. 48000.
maxptime: the decoder's maximum length of time in milliseconds
rounded up to the next full integer value represented by the media
in a packet that can be encapsulated in a received packet
according to Section 6 of [RFC4566]. Possible values are 3, 5,
10, 20, 40, and 60 or an arbitrary multiple of Opus frame sizes
rounded up to the next full integer value up to a maximum value of
120 as defined in Section 4. If no value is specified, 120 is
assumed as default. This value is a recommendation by the
decoding side to ensure the best performance for the decoder. The
decoder MUST be capable of accepting any allowed packet sizes to
ensure maximum compatibility.
ptime: the decoder's recommended length of time in milliseconds
rounded up to the next full integer value represented by the media
in a packet according to Section 6 of [RFC4566]. Possible values
are 3, 5, 10, 20, 40, or 60 or an arbitrary multiple of Opus frame
sizes rounded up to the next full integer value up to a maximum
value of 120 as defined in Section 4. If no value is specified,
20 is assumed as default. If ptime is greater than maxptime,
ptime MUST be ignored. This parameter MAY be changed during a
session. This value is a recommendation by the decoding side to
ensure the best performance for the decoder. The decoder MUST be
capable of accepting any allowed packet sizes to ensure maximum
compatibility.
minptime: the decoder's minimum length of time in milliseconds
rounded up to the next full integer value represented by the media
in a packet that SHOULD be encapsulated in a received packet
according to Section 6 of [RFC4566]. Possible values are 3, 5,
10, 20, 40, and 60 or an arbitrary multiple of Opus frame sizes
rounded up to the next full integer value up to a maximum value of
120 as defined in Section 4. If no value is specified, 3 is
assumed as default. This value is a recommendation by the
decoding side to ensure the best performance for the decoder. The
decoder MUST be capable to accept any allowed packet sizes to
ensure maximum compatibility.
maxaveragebitrate: specifies the maximum average receive bitrate of
a session in bits per second (b/s). The actual value of the
bitrate may vary as it is dependent on the characteristics of the
media in a packet. Note that the maximum average bitrate MAY be
modified dynamically during a session. Any positive integer is
allowed but values outside the range between 6000 and 510000
SHOULD be ignored. If no value is specified, the maximum value
specified in Section 3.1.1 for the corresponding mode of Opus and
corresponding maxplaybackrate: will be the default.
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stereo: specifies whether the decoder prefers receiving stereo or
mono signals. Possible values are 1 and 0 where 1 specifies that
stereo signals are preferred and 0 specifies that only mono
signals are preferred. Independent of the stereo parameter every
receiver MUST be able to receive and decode stereo signals but
sending stereo signals to a receiver that signaled a preference
for mono signals may result in higher than necessary network
utilisation and encoding complexity. If no value is specified,
mono is assumed (stereo=0).
sprop-stereo: specifies whether the sender is likely to produce
stereo audio. Possible values are 1 and 0 where 1 specifies that
stereo signals are likely to be sent, and 0 speficies that the
sender will likely only send mono. This is not a guarantee that
the sender will never send stereo audio (e.g. it could send a pre-
recorded prompt that uses stereo), but it indicates to the
receiver that the received signal can be safely downmixed to mono.
This parameter is useful to avoid wasting receiver resources by
operating the audio processing pipeline (e.g. echo cancellation)
in stereo when not necessary. If no value is specified, mono is
assumed (sprop-stereo=0).
cbr: specifies if the decoder prefers the use of a constant bitrate
versus variable bitrate. Possible values are 1 and 0 where 1
specifies constant bitrate and 0 specifies variable bitrate. If
no value is specified, cbr is assumed to be 0. Note that the
maximum average bitrate may still be changed, e.g. to adapt to
changing network conditions.
useinbandfec: specifies that the decoder has the capability to take
advantage of the Opus in-band FEC. Possible values are 1 and 0.
It is RECOMMENDED to provide 0 in case FEC cannot be utilized on
the receiving side. If no value is specified, useinbandfec is
assumed to be 0. This parameter is only a preference and the
receiver MUST be able to process packets that include FEC
information, even if it means the FEC part is discarded.
usedtx: specifies if the decoder prefers the use of DTX. Possible
values are 1 and 0. If no value is specified, usedtx is assumed
to be 0.
Encoding considerations:
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Opus media type is framed and consists of binary data according to
Section 4.8 in [RFC4288].
Security considerations:
See Section 7 of this document.
Interoperability considerations: none
Published specification: none
Applications that use this media type:
Any application that requires the transport of speech or audio
data may use this media type. Some examples are, but not limited
to, audio and video conferencing, Voice over IP, media streaming.
Person & email address to contact for further information:
SILK Support silksupport@skype.net
Jean-Marc Valin jmvalin@jmvalin.ca
Intended usage: COMMON
Restrictions on usage:
For transfer over RTP, the RTP payload format (Section 4 of this
document) SHALL be used.
Author:
Julian Spittka jspittka@gmail.com
Koen Vos koenvos74@gmail.com
Jean-Marc Valin jmvalin@jmvalin.ca
Change controller: TBD
6.2. Mapping to SDP Parameters
The information described in the media type specification has a
specific mapping to fields in the Session Description Protocol (SDP)
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[RFC4566], which is commonly used to describe RTP sessions. When SDP
is used to specify sessions employing the Opus codec, the mapping is
as follows:
o The media type ("audio") goes in SDP "m=" as the media name.
o The media subtype ("opus") goes in SDP "a=rtpmap" as the encoding
name. The RTP clock rate in "a=rtpmap" MUST be 48000 and the
number of channels MUST be 2.
o The OPTIONAL media type parameters "ptime" and "maxptime" are
mapped to "a=ptime" and "a=maxptime" attributes, respectively, in
the SDP.
o The OPTIONAL media type parameters "maxaveragebitrate",
"maxplaybackrate", "minptime", "stereo", "cbr", "useinbandfec",
and "usedtx", when present, MUST be included in the "a=fmtp"
attribute in the SDP, expressed as a media type string in the form
of a semicolon-separated list of parameter=value pairs (e.g.,
maxaveragebitrate=20000). They MUST NOT be specified in an SSRC-
specific "fmtp" source-level attribute (as defined in Section 6.3
of [RFC5576]).
o The OPTIONAL media type parameters "sprop-maxcapturerate", and
"sprop-stereo" MAY be mapped to the "a=fmtp" SDP attribute by
copying them directly from the media type parameter string as part
of the semicolon-separated list of parameter=value pairs (e.g.,
sprop-stereo=1). These same OPTIONAL media type parameters MAY
also be specified using an SSRC-specific "fmtp" source-level
attribute as described in Section 6.3 of [RFC5576]. They MAY be
specified in both places, in which case the parameter in the
source-level attribute overrides the one found on the "a=fmtp"
line. The value of any parameter which is not specified in a
source-level source attribute MUST be taken from the "a=fmtp"
line, if it is present there.
Below are some examples of SDP session descriptions for Opus:
Example 1: Standard mono session with 48000 Hz clock rate
m=audio 54312 RTP/AVP 101
a=rtpmap:101 opus/48000/2
Example 2: 16000 Hz clock rate, maximum packet size of 40 ms,
recommended packet size of 40 ms, maximum average bitrate of 20000
bps, prefers to receive stereo but only plans to send mono, FEC is
allowed, DTX is not allowed
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m=audio 54312 RTP/AVP 101
a=rtpmap:101 opus/48000/2
a=fmtp:101 maxplaybackrate=16000; sprop-maxcapturerate=16000;
maxaveragebitrate=20000; stereo=1; useinbandfec=1; usedtx=0
a=ptime:40
a=maxptime:40
Example 3: Two-way full-band stereo preferred
m=audio 54312 RTP/AVP 101
a=rtpmap:101 opus/48000/2
a=fmtp:101 stereo=1; sprop-stereo=1
6.2.1. Offer-Answer Model Considerations for Opus
When using the offer-answer procedure described in [RFC3264] to
negotiate the use of Opus, the following considerations apply:
o Opus supports several clock rates. For signaling purposes only
the highest, i.e. 48000, is used. The actual clock rate of the
corresponding media is signaled inside the payload and is not
subject to this payload format description. The decoder MUST be
capable to decode every received clock rate. An example is shown
below:
m=audio 54312 RTP/AVP 100
a=rtpmap:100 opus/48000/2
o The "ptime" and "maxptime" parameters are unidirectional receive-
only parameters and typically will not compromise
interoperability; however, dependent on the set values of the
parameters the performance of the application may suffer.
[RFC3264] defines the SDP offer-answer handling of the "ptime"
parameter. The "maxptime" parameter MUST be handled in the same
way.
o The "minptime" parameter is a unidirectional receive-only
parameters and typically will not compromise interoperability;
however, dependent on the set values of the parameter the
performance of the application may suffer and should be set with
care.
o The "maxplaybackrate" parameter is a unidirectional receive-only
parameter that reflects limitations of the local receiver. The
sender of the other side SHOULD NOT send with an audio bandwidth
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higher than "maxplaybackrate" as this would lead to inefficient
use of network resources. The "maxplaybackrate" parameter does
not affect interoperability. Also, this parameter SHOULD NOT be
used to adjust the audio bandwidth as a function of the bitrates,
as this is the responsibility of the Opus encoder implementation.
o The "maxaveragebitrate" parameter is a unidirectional receive-only
parameter that reflects limitations of the local receiver. The
sender of the other side MUST NOT send with an average bitrate
higher than "maxaveragebitrate" as it might overload the network
and/or receiver. The "maxaveragebitrate" parameter typically will
not compromise interoperability; however, dependent on the set
value of the parameter the performance of the application may
suffer and should be set with care.
o The "sprop-maxcapturerate" and "sprop-stereo" parameters are
unidirectional sender-only parameters that reflect limitations of
the sender side. They allow the receiver to set up a reduced-
complexity audio processing pipeline if the sender is not planning
to use the full range of Opus's capabilities. Neither "sprop-
maxcapturerate" nor "sprop-stereo" affect interoperability and the
receiver MUST be capable of receiving any signal.
o The "stereo" parameter is a unidirectional receive-only parameter.
o The "cbr" parameter is a unidirectional receive-only parameter.
o The "useinbandfec" parameter is a unidirectional receive-only
parameter.
o The "usedtx" parameter is a unidirectional receive-only parameter.
o Any unknown parameter in an offer MUST be ignored by the receiver
and MUST be removed from the answer.
6.2.2. Declarative SDP Considerations for Opus
For declarative use of SDP such as in Session Announcement Protocol
(SAP), [RFC2974], and RTSP, [RFC2326], for Opus, the following needs
to be considered:
o The values for "maxptime", "ptime", "minptime", "maxplaybackrate",
and "maxaveragebitrate" should be selected carefully to ensure
that a reasonable performance can be achieved for the participants
of a session.
o The values for "maxptime", "ptime", and "minptime" of the payload
format configuration are recommendations by the decoding side to
ensure the best performance for the decoder. The decoder MUST be
capable to accept any allowed packet sizes to ensure maximum
compatibility.
o All other parameters of the payload format configuration are
declarative and a participant MUST use the configurations that are
provided for the session. More than one configuration may be
provided if necessary by declaring multiple RTP payload types;
however, the number of types should be kept small.
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7. Security Considerations
All RTP packets using the payload format defined in this
specification are subject to the general security considerations
discussed in the RTP specification [RFC3550] and any profile from
e.g. [RFC3711] or [RFC3551].
This payload format transports Opus encoded speech or audio data,
hence, security issues include confidentiality, integrity protection,
and authentication of the speech or audio itself. The Opus payload
format does not have any built-in security mechanisms. Any suitable
external mechanisms, such as SRTP [RFC3711], MAY be used.
This payload format and the Opus encoding do not exhibit any
significant non-uniformity in the receiver-end computational load and
thus are unlikely to pose a denial-of-service threat due to the
receipt of pathological datagrams.
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8. Acknowledgements
TBD
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9. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
Streaming Protocol (RTSP)", RFC 2326, April 1998.
[RFC2974] Handley, M., Perkins, C., and E. Whelan, "Session
Announcement Protocol", RFC 2974, October 2000.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC4288] Freed, N. and J. Klensin, "Media Type Specifications and
Registration Procedures", BCP 13, RFC 4288, December 2005.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4855] Casner, S., "Media Type Registration of RTP Payload
Formats", RFC 4855, February 2007.
[RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
Media Attributes in the Session Description Protocol
(SDP)", RFC 5576, June 2009.
[RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of
Variable Bit Rate Audio with Secure RTP", RFC 6562,
March 2012.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, September 2012.
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Authors' Addresses
Julian Spittka
Email: jspittka@gmail.com
Koen Vos
Skype Technologies S.A.
3210 Porter Drive
Palo Alto, CA 94304
USA
Email: koenvos74@gmail.com
Jean-Marc Valin
Mozilla
650 Castro Street
Mountain View, CA 94041
USA
Email: jmvalin@jmvalin.ca
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