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draft-ietf-payload-rtp-opus
Network Working Group J. Spittka
Internet-Draft K. Vos
Intended status: Informational Skype Technologies S.A.
Expires: January 5, 2012 JM. Valin
Octasic Inc.
July 4, 2011
RTP Payload Format and File Storage Format for Opus Speech and Audio
Codec
draft-spittka-payload-rtp-opus-00
Abstract
This document defines the Real-time Transport Protocol (RTP) payload
format and file storage 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 and the file storage format.
Status of this Memo
This Internet-Draft is submitted to IETF in full conformance with the
provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on January 5, 2012.
Copyright Notice
Copyright (c) 2011 IETF Trust and the persons identified as the
document authors. All rights reserved.
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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
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described in the BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Conventions, Definitions and Acronyms used in this document . 4
3. Opus Codec . . . . . . . . . . . . . . . . . . . . . . . . . . 5
3.1. Modes . . . . . . . . . . . . . . . . . . . . . . . . . . 5
3.1.1. Audio Mode . . . . . . . . . . . . . . . . . . . . . . 5
3.1.2. Audio Mode . . . . . . . . . . . . . . . . . . . . . . 6
3.2. Network Bandwidth . . . . . . . . . . . . . . . . . . . . 6
3.2.1. Variable versus Constant Bit Rate . . . . . . . . . . 6
3.2.2. Discontinuous Transmission (DTX) . . . . . . . . . . . 7
3.3. Complexity . . . . . . . . . . . . . . . . . . . . . . . . 7
3.4. Forward Error Correction (FEC) . . . . . . . . . . . . . . 7
3.5. Stereo Operation . . . . . . . . . . . . . . . . . . . . . 8
4. Opus RTP Payload Format . . . . . . . . . . . . . . . . . . . 9
4.1. RTP Header Usage . . . . . . . . . . . . . . . . . . . . . 9
4.2. Payload Structure . . . . . . . . . . . . . . . . . . . . 10
5. Opus Storage Format . . . . . . . . . . . . . . . . . . . . . 12
5.1. Storage Header Structure . . . . . . . . . . . . . . . . . 12
5.2. Storage Block Structure . . . . . . . . . . . . . . . . . 12
6. Congestion Control . . . . . . . . . . . . . . . . . . . . . . 14
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15
7.1. Opus Media Type Registration . . . . . . . . . . . . . . . 15
7.2. Mapping to SDP Parameters . . . . . . . . . . . . . . . . 18
7.2.1. Offer-Answer Model Considerations for Opus . . . . . . 19
7.2.2. Declarative SDP Considerations for Opus . . . . . . . 20
8. Security Considerations . . . . . . . . . . . . . . . . . . . 22
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23
10. Normative References . . . . . . . . . . . . . . . . . . . . . 24
A. Informational References . . . . . . . . . . . . . . . . . . . 25
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 26
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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 is is highly scalable in terms of
audio bandwidth, network bit rate, 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 and file storage 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 and the file
storage format. More information on the Opus codec can be obtained
from the following IETF draft [Opus].
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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].
CPU: Central Processing Unit
IP: Internet Protocol
PSTN: Public Switched Telephone Network
samples: Speech or audio samples
SDP: Session Description Protocol
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3. Opus Codec
The Opus 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 to efficiently encode voice signals at lower
bit rates while the audio mode is optimized for audio signals at
medium and higher bit rates.
The Opus speech and audio codec is highly scalable in terms of audio
bandwidth, network bit rate, and complexity. Further, Opus allows to
transmit stereo signals.
The Opus speech and audio codec is based on the SILK codec [SILK] and
the CELT codec [CELT]. For more detailed information on how Opus
operates, also refer to [Opus].
3.1. Modes
Opus supports five different audio bandwidths, 8000, 12000, 16000,
24000, and 48000 Hz sampling frequency, for the voice mode and four
different audio bandwidths, 8000, 16000, 24000, and 48000 Hz sampling
frequency, for the audio mode.
3.1.1. Audio Mode
For low bit rate applications transmitting mostly speech signals the
voice mode of Opus SHOULD be used. The voice mode allows to encode
voice signals at 8000, 12000, 16000, 24000, and 48000 Hz sampling
frequency.
A sampling rate of 8000 Hz SHOULD only be used to interface to PSTN
networks or on low end devices that do not support greater than 8000
Hz sampling frequency. A sampling rate of 12000 Hz SHOULD be used
for lower end devices that do not support greater than 12000 Hz
sampling frequency or are under severe network bandwidth constrains
(e.g. wireless devices). A sampling rate of 16000 Hz SHOULD be used
for all-IP platforms that do not support greater than 16000 Hz
sampling frequency. Higher sampling rates are recommended for all
devices that support those high sampling rates and desire full-
bandwidth speech at medium bit rates.
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3.1.2. Audio Mode
For applications desiring very low delay speech transmission as well
as music transmission in trade off to a higher bit rate, the audio
mode SHOULD be used. This mode supports audio sampling rates of
8000, 16000, 24000, and 48000 Hz.
3.2. Network Bandwidth
The network bit rate is adaptive within the range specified in
Table 1 for corresponding modes and audio sampling rates. The
average target network bit rate can be defined and modified in real-
time while the actual bit rate will be dependent on the settings of
Opus and the input signal and may change over time.
+-------+---------+-----------+
| Mode | fs (Hz) | BR (kbps) |
+-------+---------+-----------+
| voice | 8000 | 6 - 20 |
| | | |
| voice | 12000 | 7 - 25 |
| | | |
| voice | 16000 | 8 - 30 |
| | | |
| voice | 24000 | 18 - 28 |
| | | |
| voice | 48000 | 24 - 32 |
| | | |
| audio | 8000 | 20 - 28 |
| | | |
| audio | 16000 | 24 - 32 |
| | | |
| audio | 24000 | 28 - 40 |
| | | |
| audio | 48000 | 32 - 128 |
+-------+---------+-----------+
Mode specifies the Opus mode of operation; fs specifies the audio
sampling frequency in Hertz (Hz); BR specifies the network bit rate
range in kilobits per second (kbps).
Table 1
3.2.1. Variable versus Constant Bit Rate
The voice mode will always use a variable bit rate at audio sampling
rates of 8000, 12000, and 16000 Hz. The average target bit rate can
be adjusted at any point in time. To avoid congestion of the
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connection the average target bit rate SHOULD be adjusted to the
available network bandwidth. If no target bit rate is specified the
average bit rate may go up to the highest bit rate specified in
Table 1.
In voice mode at audio sampling rates higher than 16000 Hz, i.e.
24000, and 48000 Hz, and audio mode Opus can be operated in both
variable and constant bit rate. The target bit rate can be adjusted
at any point in time.
3.2.2. Discontinuous Transmission (DTX)
The Opus codec may, as described in Section 3.2.1, be operated with
an adaptive bit rate. In that case, the bit rate will automatically
be reduced for certain input signals like periods of silence. During
continuous transmission the bit rate 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 bit rate of Opus needs to be reduced even further
or in cases where only constant bit rate 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
bandwidth are severe. The DTX mode can be engaged for operation in
both adaptive or constant bit rate.
3.3. Complexity
Complexity can be scaled to optimize for CPU resources in real-time,
mostly in trade-off to network bit rate. Also, different modes of
Opus have different complexity.
3.4. 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
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scheme adds redundant information about the previous packet (n-1) to
the current output packet n. For each frame, the encoder decides
whether to use FEC based on (1) an externally-provided estimate of
the channel's packet loss rate; (2) an externally-provided estimate
of the channel's capacity; (3) the sensitivity of the audio or speech
signal to packet loss; (4) whether the receiving decoder has
indicated it can take advantage of "in-band" FEC information. The
decision to send "in-band" FEC information is entirely controlled by
the encoder and therefore no special precautions for the payload or
storage format 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
bandwidth. Decoder support for FEC SHOULD be indicated at the time a
session is set up.
3.5. Stereo Operation
Opus allows for transmission of stereo audio signals. This operation
will be signaled in the Opus payload and no special arrangements have
to be made in the payload format. Any implementation of the Opus
decoder MUST be capable to receive stereo signals.
If a decoder can not take advantage of the benefits of a stereo
signal this SHOULD be indicated at the time a session is set up. In
that case the sending side SHOULD NOT send stereo signals as it leads
to an inefficient usage of network bandwidth.
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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 has no function in combination
with Opus and MAY be ignored.
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 sampling rates 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. 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 |
+---------+------------+
fs specifies the audio sampling frequency in Hertz (Hz); Multiplier
is the value that the number of samples have to be multiplied with to
calculate the RTP timestamp.
Table 2
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 for different modes and sampling
rates of Opus and how the timestamp needs to be incremented for
packetization.
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+------+------------------------+----+----+-----+-----+------+------+
| Mode | fs | 2. | 5 | 10 | 20 | 40 | 60 |
| | | 5 | | | | | |
+------+------------------------+----+----+-----+-----+------+------+
| ts | all | 12 | 24 | 480 | 960 | 1920 | 2880 |
| incr | | 0 | 0 | | | | |
| | | | | | | | |
| voic | 8000/12000/16000/24000 | | | x | x | x | x |
| e | /48000 | | | | | | |
| | | | | | | | |
| audi | 8000/16000/24000/48000 | x | x | x | x | | |
| o | | | | | | | |
+------+------------------------+----+----+-----+-----+------+------+
Mode specifies the Opus mode of operation; fs specifies the audio
sampling frequency in Hertz (Hz); 2.5, 5, 10, 20, 40, and 60
represent the duration of encoded speech or audio data in a packet;
ts incr specifies the value the timestamp needs to be incremented for
the representing packet size. For multiple frames in a packet these
values have to be multiplied with the respective number of frames.
Table 3
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5. Opus Storage Format
The Opus storage format allows to store Opus encoded data into e.g. a
file or an email attachment. The storage format consists of a header
and a series of blocks containing encoded speech or audio frames.
The storage format closely mimics the real-time payload format and
allows to easily convert packets, e.g. received by a voicemail
system, into a storage format and vice versa and therefore allowing
maximum flexibility and low overhead. Please note that this storage
format is not meant to be a robust storage format, nor the most
efficient storage format. For a robust storage format that allows
advanced functionality like e.g. seeking, a more advanced container
format should be used.
Figure 2 shows an example of an Opus encoded file. Note that due to
the potentially adaptive bit rate the packet length may be variable
and no fixed block size can be defined for blocks containing encoded
data.
+------------------+
| Header |
+-----------+------+
| block 1 |
+-----------+--+
| block 2 |
+--------------+--+
: ... :
+--------------+--+
| block n |
+-----------------+
Figure 2: Example of Opus file storage format showing different block
lengths due to potentially adaptive bit rate of Opus
5.1. Storage Header Structure
An Opus storage header contains the following ASCII character string
as a magic number:
"#!opus\n" (hexadecimal: 0x23 0x21 0x6f 0x70 0x75 0x73 0x0A)
5.2. Storage Block Structure
Following the storage header, blocks of encoded data are stored in
consecutive order in time according to Figure 2. Each block contains
a block header followed by a payload according to Figure 3.
The block header contains information that, for an RTP-based session,
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can be derived from the IP and RTP headers: The number of octets
contained in the subsequent payload and the RTP timestamp.
The number of octets in the payload is represented by 16 bits and the
timestamp is specified by 32 bits. For the first block, the
timestamp MAY be a random number. For the following blocks, the
timestamp MUST be incremented according to the way timestamps are
incremented when Opus payloads are transmitted over RTP.
0 16 48
+-------------------+----------------------------+-----------------
| # of octets | Timestamp | Payload
+-------------------+----------------------------+-----------------
Figure 3: Storage block header structure
The payload of each block in Figure 2 represents one packet of Opus
encoded data the way as originally encoded by the Opus encoder.
Information about frame size representing the duration of encoded
speech or audio data, number of encoded frames, stereo information,
and DTX is embedded into the payload of Opus and not subject to the
storage format. It can be extracted from the payload during decoding
of the encoded data.
During the usage of DTX no blocks are stored when the channel is
inactive. Timestamps MUST be used to reassemble the decoded signal
in a time-aligned way.
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6. Congestion Control
The adaptive nature of the Opus codec allows for an efficient
congestion control.
The voice mode of Opus at audio sampling rates of 8000, 12000, and
16000 always runs with a variable bit rate. The average bit rate in
that mode is dependent on the input signal and will especially
decrease during silent periods. The voice mode at audio sampling
rates of 24000 and 48000 Hz and the audio mode may run at a variable
or constant bit rate. In either way, the target bit rate 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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7. IANA Considerations
One media subtype (audio/opus) has been defined and registered as
described in the following section.
7.1. 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:
maxcodedaudiobandwidth: the decoder's maximum sampling frequency
specified in Hertz (Hz) that the application can take advantage
of. The decoder MUST be capable to receive any allowed sampling
frequency but due to hardware limitations only signals up to the
specified sampling frequency can be processed. Sending signals
with higher sampling frequency may result in higher than necessary
network bandwidth and encoding complexity. Possible values are
8000, 12000, 16000, 24000, 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 and Section 5 of this document. 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 to accept any
allowed packet sizes to ensure maximum compatibility.
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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 and Section 5 of this
document. 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 to accept 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 and Section 5 of this document. 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 bit rate of
a session in bits per second (bps). The actual value of the bit
rate may vary as it is dependent on the characteristics of the
media in a packet. Note that the maximum average bit rate 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 Table 1 for the corresponding mode of Opus and
corresponding clock rate will be the default.
stereo: specifies if the decoder prefers to receive stereo signals
versus 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 bandwidth and encoding complexity. If no value is
specified, stereo is assumed to be 0.
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cbr: specifies if the decoder prefers the use of a constant bit rate
versus variable bit rate. Possible values are 1 and 0 where 1
specifies constant bit rate and 0 specifies variable bit rate. If
no value is specified, cbr is assumed to be 0. Note that the
maximum average bit rate may still be changed, e.g. to adapt to
changing network conditions.
useinbandfec: specifies that Opus in-band FEC is supported by the
decoder and MAY be used during a session. Possible values are 1
and 0. It is RECOMMENDED to provide 0 in case FEC is not
implemented on the receiving side. If no value is specified,
useinbandfec is assumed to be 1.
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:
Opus media type is framed and consists of binary data according to
Section 4.8 in [RFC4288].
Security considerations:
See Section 8 of this document.
Interoperability considerations: none
Published specification: none
Applications that use this media type:
Any application that requires the transport or storage of speech
or audio data may use this media type. Some examples are, but not
limited to, audio and video conferencing, Voice over IP, voice
recording, media streaming, voice messaging.
Additional information:
For storage transfer methods the following applies:
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Magic number:"#!opus\n" (hexadecimal: 0x23 0x21 0x6f 0x70 0x75
0x73 0x0A)
File extension(s): ops, OPS
Macintosh file type code(s): "opus"
Person & email address to contact for further information:
SILK Support silksupport@skype.net
Jean-Marc Valin jean-marc.valin@octasic.com
Intended usage: COMMON
Restrictions on usage:
For transfer over RTP, the RTP payload format (Section 4 of this
document) SHALL be used. For storage usage, the storage format
(Section 5 of this document) SHALL be used.
Author:
Julian Spittka julian.spittka@skype.net
Koen Vos koen.vos@skype.net
Jean-Marc Valin jean-marc.valin@octasic.com
Change controller: TBD
7.2. Mapping to SDP Parameters
The information described in the media type specification has a
specific mapping to fields in the Session Description Protocol (SDP)
[RFC4566], which is commonly used to describe RTP sessions. When SDP
is used to specify sessions employing the Opus codec, the mapping is
as follows:
o The media type ("audio") goes in SDP "m=" as the media name.
o The media subtype ("opus") goes in SDP "a=rtpmap" as the encoding
name. The RTP clock rate in "a=rtpmap" MUST be mapped to the
required media type parameter "rate".
o The optional media type parameters "ptime" and "maxptime" are
mapped to "a=ptime" and "a=maxptime" attributes, respectively, in
the SDP.
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o All remaining media type parameters are mapped to the "a=fmtp"
attribute in the SDP by copying them directly from the media type
parameter string as a semicolon-separated list of parameter=value
pairs (e.g. maxaveragebitrate=20000).
Below are some examples of SDP session descriptions for Opus:
Example 1: Standard session with 48000 Hz clock rate
m=audio 54312 RTP/AVP 101
a=rtpmap:101 opus/48000
Example 2: 16000 Hz clock rate, maximum packet size of 40 ms,
recommended packet size of 40 ms, maximum average bit rate of 20000
bps, stereo signals are preferred, FEC is allowed, DTX is not allowed
m=audio 54312 RTP/AVP 101
a=rtpmap:101 opus/48000
a=fmtp:101 maxcodedaudiobandwidth=16000; maxaveragebitrate=20000;
stereo=1; useinbandfec=1; usedtx=0
a=ptime:40
a=maxptime:40
7.2.1. 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
o The parameters "ptime" and "maxptime" 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.
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[RFC3264] defines the SDP offer-answer handling of the "ptime"
parameter. The "maxptime" parameter MUST be handled in the same
way.
o The parameter "minptime" 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 parameter "maxcodedaudiobandwidth" is a unidirectional
receive-only parameter that reflects limitations of the local
receiver. The sender of the other side SHOULD NOT send with a
sampling rate higher than "maxcodedaudiobandwidth" as it
represents an inefficient use of network bandwidth resources and
CPU cycles on the encoding side. The parameter
"maxcodedaudiobandwidth" 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 parameter "maxaveragebitrate" 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 bit rate
higher than "maxaveragebitrate" as it might overload the network
and/or receiver. The parameter "maxaveragebitrate" 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 If the parameter "maxaveragebitrate" is below the range specified
in Table 1 the session MUST be rejected.
o The parameter "stereo" is a unidirectional receive-only parameter.
o The parameter "cbr" is a unidirectional receive-only parameter.
o The parameter "useinbandfec" is a unidirectional receive-only
parameter.
o The parameter "usedtx" 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.
7.2.2. 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",
"maxcodedaudiobandwidth", and "maxaveragebitrate" should be
selected carefully to ensure that a reasonable performance can be
achieved for the participants of a session.
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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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8. 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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9. Acknowledgements
TBD
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10. 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.
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Appendix A. Informational References
[codec] http://datatracker.ietf.org/wg/codec/
[SILK] https://developer.skype.com/silk
[CELT] http://www.celt-codec.org/
[Opus] http://datatracker.ietf.org/doc/draft-ietf-codec-opus/
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Authors' Addresses
Julian Spittka
Skype Technologies S.A.
3210 Porter Drive
Palo Alto, CA 94304
USA
Email: julian.spittka@skype.net
Koen Vos
Skype Technologies S.A.
3210 Porter Drive
Palo Alto, CA 94304
USA
Email: koen.vos@skype.net
Jean-Marc Valin
Octasic Inc.
4101 Molson Street
Montreal, Quebec
Canada
Email: jean-marc.valin@octasic.com
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