draft-ietf-payload-rtp-opus-02.txt   draft-ietf-payload-rtp-opus-03.txt 
Network Working Group J. Spittka Network Working Group J. Spittka
Internet-Draft Internet-Draft
Intended status: Standards Track K. Vos Intended status: Standards Track K. Vos
Expires: January 01, 2015 vocTone Expires: January 31, 2015 vocTone
JM. Valin JM. Valin
Mozilla Mozilla
June 30, 2014 July 30, 2014
RTP Payload Format for Opus Speech and Audio Codec RTP Payload Format for Opus Speech and Audio Codec
draft-ietf-payload-rtp-opus-02 draft-ietf-payload-rtp-opus-03
Abstract Abstract
This document defines the Real-time Transport Protocol (RTP) payload This document defines the Real-time Transport Protocol (RTP) payload
format for packetization of Opus encoded speech and audio data that format for packetization of Opus encoded speech and audio data
is essential to integrate the codec in the most compatible way. necessary to integrate the codec in the most compatible way.
Further, media type registrations are described for the RTP payload Further, it describes media type registrations for the RTP payload
format. format.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on January 01, 2015. This Internet-Draft will expire on January 31, 2015.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
skipping to change at page 2, line 16 skipping to change at page 2, line 16
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Conventions, Definitions and Acronyms used in this document . 3 2. Conventions, Definitions and Acronyms used in this document . 3
2.1. Audio Bandwidth . . . . . . . . . . . . . . . . . . . . . 3 2.1. Audio Bandwidth . . . . . . . . . . . . . . . . . . . . . 3
3. Opus Codec . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Opus Codec . . . . . . . . . . . . . . . . . . . . . . . . . 3
3.1. Network Bandwidth . . . . . . . . . . . . . . . . . . . . 4 3.1. Network Bandwidth . . . . . . . . . . . . . . . . . . . . 4
3.1.1. Recommended Bitrate . . . . . . . . . . . . . . . . . 4 3.1.1. Recommended Bitrate . . . . . . . . . . . . . . . . . 4
3.1.2. Variable versus Constant Bit Rate . . . . . . . . . . 4 3.1.2. Variable versus Constant Bitrate . . . . . . . . . . 4
3.1.3. Discontinuous Transmission (DTX) . . . . . . . . . . 4 3.1.3. Discontinuous Transmission (DTX) . . . . . . . . . . 4
3.2. Complexity . . . . . . . . . . . . . . . . . . . . . . . 5 3.2. Complexity . . . . . . . . . . . . . . . . . . . . . . . 5
3.3. Forward Error Correction (FEC) . . . . . . . . . . . . . 5 3.3. Forward Error Correction (FEC) . . . . . . . . . . . . . 5
3.4. Stereo Operation . . . . . . . . . . . . . . . . . . . . 6 3.4. Stereo Operation . . . . . . . . . . . . . . . . . . . . 6
4. Opus RTP Payload Format . . . . . . . . . . . . . . . . . . . 6 4. Opus RTP Payload Format . . . . . . . . . . . . . . . . . . . 6
4.1. RTP Header Usage . . . . . . . . . . . . . . . . . . . . 6 4.1. RTP Header Usage . . . . . . . . . . . . . . . . . . . . 6
4.2. Payload Structure . . . . . . . . . . . . . . . . . . . . 7 4.2. Payload Structure . . . . . . . . . . . . . . . . . . . . 7
5. Congestion Control . . . . . . . . . . . . . . . . . . . . . 8 5. Congestion Control . . . . . . . . . . . . . . . . . . . . . 8
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9 6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
6.1. Opus Media Type Registration . . . . . . . . . . . . . . 9 6.1. Opus Media Type Registration . . . . . . . . . . . . . . 9
6.2. Mapping to SDP Parameters . . . . . . . . . . . . . . . . 13 6.2. Mapping to SDP Parameters . . . . . . . . . . . . . . . . 12
6.2.1. Offer-Answer Model Considerations for Opus . . . . . 15 6.2.1. Offer-Answer Model Considerations for Opus . . . . . 14
6.2.2. Declarative SDP Considerations for Opus . . . . . . . 16 6.2.2. Declarative SDP Considerations for Opus . . . . . . . 15
7. Security Considerations . . . . . . . . . . . . . . . . . . . 16 7. Security Considerations . . . . . . . . . . . . . . . . . . . 16
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 17 8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 16
9. Normative References . . . . . . . . . . . . . . . . . . . . 17 9. Normative References . . . . . . . . . . . . . . . . . . . . 16
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 18 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 17
1. Introduction 1. Introduction
The Opus codec is a speech and audio codec developed within the IETF The Opus codec is a speech and audio codec developed within the IETF
Internet Wideband Audio Codec working group (codec). The codec has a Internet Wideband Audio Codec working group. The codec has a very
very low algorithmic delay and it is highly scalable in terms of low algorithmic delay and it is highly scalable in terms of audio
audio bandwidth, bitrate, and complexity. Further, it provides bandwidth, bitrate, and complexity. Further, it provides different
different modes to efficiently encode speech signals as well as music modes to efficiently encode speech signals as well as music signals,
signals, thus, making it the codec of choice for various applications thus making it the codec of choice for various applications using the
using the Internet or similar networks. Internet or similar networks.
This document defines the Real-time Transport Protocol (RTP) This document defines the Real-time Transport Protocol (RTP)
[RFC3550] payload format for packetization of Opus encoded speech and [RFC3550] payload format for packetization of Opus encoded speech and
audio data that is essential to integrate the Opus codec in the most audio data necessary to integrate the Opus codec in the most
compatible way. Further, media type registrations are described for compatible way. Further, it describes media type registrations for
the RTP payload format. More information on the Opus codec can be the RTP payload format. More information on the Opus codec can be
obtained from [RFC6716]. obtained from [RFC6716].
2. Conventions, Definitions and Acronyms used in this document 2. Conventions, Definitions and Acronyms used in this document
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119]. document are to be interpreted as described in [RFC2119].
CBR: Constant bitrate CBR: Constant bitrate
CPU: Central Processing Unit CPU: Central Processing Unit
DTX: Discontinuous transmission DTX: Discontinuous transmission
FEC: Forward error correction FEC: Forward error correction
IP: Internet Protocol IP: Internet Protocol
samples: Speech or audio samples (usually per channel) samples: Speech or audio samples (per channel)
SDP: Session Description Protocol SDP: Session Description Protocol
VBR: Variable bitrate VBR: Variable bitrate
2.1. Audio Bandwidth 2.1. Audio Bandwidth
Throughout this document, we refer to the following definitions: Throughout this document, we refer to the following definitions:
+--------------+----------------+-----------+----------+ +--------------+----------------+-----------------+-----------------+
| Abbreviation | Name | Bandwidth | Sampling | | Abbreviation | Name | Audio Bandwidth | Sampling Rate |
+--------------+----------------+-----------+----------+ | | | (Hz) | (Hz) |
| nb | Narrowband | 0 - 4000 | 8000 | +--------------+----------------+-----------------+-----------------+
| | | | | | NB | Narrowband | 0 - 4000 | 8000 |
| mb | Mediumband | 0 - 6000 | 12000 | | | | | |
| | | | | | MB | Mediumband | 0 - 6000 | 12000 |
| wb | Wideband | 0 - 8000 | 16000 | | | | | |
| | | | | | WB | Wideband | 0 - 8000 | 16000 |
| swb | Super-wideband | 0 - 12000 | 24000 | | | | | |
| | | | | | SWB | Super-wideband | 0 - 12000 | 24000 |
| fb | Fullband | 0 - 20000 | 48000 | | | | | |
+--------------+----------------+-----------+----------+ | FB | Fullband | 0 - 20000 | 48000 |
+--------------+----------------+-----------------+-----------------+
Audio bandwidth naming Audio bandwidth naming
Table 1 Table 1
3. Opus Codec 3. Opus Codec
The Opus [RFC6716] speech and audio codec has been developed to The Opus [RFC6716] codec encodes speech signals as well as general
encode speech signals as well as audio signals. Two different modes, audio signals. Two different modes can be chosen, a voice mode or an
a voice mode or an audio mode, may be chosen to allow the most audio mode, to allow the most efficient coding depending on the type
efficient coding dependent on the type of input signal, the sampling of the input signal, the sampling frequency of the input signal, and
frequency of the input signal, and the specific application. the intended application.
The voice mode allows efficient encoding of voice signals at lower The voice mode allows efficient encoding of voice signals at lower
bit rates while the audio mode is optimized for audio signals at bit rates while the audio mode is optimized for general audio signals
medium and higher bitrates. at medium and higher bitrates.
The Opus speech and audio codec is highly scalable in terms of audio The Opus speech and audio codec is highly scalable in terms of audio
bandwidth, bitrate, and complexity. Further, Opus allows bandwidth, bitrate, and complexity. Further, Opus allows
transmitting stereo signals. transmitting stereo signals.
3.1. Network Bandwidth 3.1. Network Bandwidth
Opus supports all bitrates from 6 kb/s to 510 kb/s. The bitrate can 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 be changed dynamically within that range. All other parameters being
equal, a higher bitrate results in higher quality. equal, higher bitrates result in higher quality.
3.1.1. Recommended Bitrate 3.1.1. Recommended Bitrate
For a frame size of 20 ms, these are the bitrate "sweet spots" for For a frame size of 20 ms, these are the bitrate "sweet spots" for
Opus in various configurations: Opus in various configurations:
o 8-12 kb/s for NB speech, o 8-12 kb/s for NB speech,
o 16-20 kb/s for WB speech, o 16-20 kb/s for WB speech,
o 28-40 kb/s for FB speech, o 28-40 kb/s for FB speech,
o 48-64 kb/s for FB mono music, and o 48-64 kb/s for FB mono music, and
o 64-128 kb/s for FB stereo music. o 64-128 kb/s for FB stereo music.
3.1.2. Variable versus Constant Bit Rate 3.1.2. Variable versus Constant Bitrate
For the same average bitrate, variable bitrate (VBR) can achieve For the same average bitrate, variable bitrate (VBR) can achieve
higher quality than constant bitrate (CBR). For the majority of higher quality than constant bitrate (CBR). For the majority of
voice transmission application, VBR is the best choice. One voice transmission applications, VBR is the best choice. One reason
potential reason for choosing CBR is the potential information leak for choosing CBR is the potential information leak that _might_ occur
that _may_ occur when encrypting the compressed stream. See when encrypting the compressed stream. See [RFC6562] for guidelines
[RFC6562] for guidelines on when VBR is appropriate for encrypted on when VBR is appropriate for encrypted audio communications. In
audio communications. In the case where an existing VBR stream needs the case where an existing VBR stream needs to be converted to CBR
to be converted to CBR for security reasons, then the Opus padding for security reasons, then the Opus padding mechanism described in
mechanism described in [RFC6716] is the RECOMMENDED way to achieve [RFC6716] is the RECOMMENDED way to achieve padding because the RTP
padding because the RTP padding bit is unencrypted. padding bit is unencrypted.
The bitrate can be adjusted at any point in time. To avoid The bitrate can be adjusted at any point in time. To avoid
congestion, the average bitrate SHOULD be adjusted to the available congestion, the average bitrate SHOULD NOT exceed the available
network capacity. If no target bitrate is specified, the bitrates network capacity. If no target bitrate is specified, the bitrates
specified in Section 3.1.1 are RECOMMENDED. specified in Section 3.1.1 are RECOMMENDED.
3.1.3. Discontinuous Transmission (DTX) 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 The Opus codec can, as described in Section 3.1.2, be operated with a
reduced for certain input signals like periods of silence. During variable bitrate. In that case, the encoder will automatically
continuous transmission the bitrate will be reduced, when the input reduce the bitrate for certain input signals, like periods of
signal allows to do so, but the transmission to the receiver itself silence. When using continuous transmission, it will reduce the
will never be interrupted. Therefore, the received signal will bitrate when the characteristics of the input signal permit, but will
maintain the same high level of quality over the full duration of a never interrupt the transmission to the receiver. Therefore, the
transmission while minimizing the average bit rate over time. 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 In cases where the bitrate of Opus needs to be reduced even further
or in cases where only constant bitrate is available, the Opus or in cases where only constant bitrate is available, the Opus
encoder may be set to use discontinuous transmission (DTX), where encoder can use discontinuous transmission (DTX), where parts of the
parts of the encoded signal that correspond to periods of silence in encoded signal that correspond to periods of silence in the input
the input speech or audio signal are not transmitted to the receiver. speech or audio signal are not transmitted to the receiver. A
A receiver can distinguish between DTX and packet loss by looking for receiver can distinguish between DTX and packet loss by looking for
gaps in the sequence number, as described by Section 4.1 gaps in the sequence number, as described by Section 4.1
of [RFC3551]. of [RFC3551].
On the receiving side, the non-transmitted parts will be handled by a On the receiving side, the non-transmitted parts will be handled by a
frame loss concealment unit in the Opus decoder which generates a frame loss concealment unit in the Opus decoder which generates a
comfort noise signal to replace the non transmitted parts of the comfort noise signal to replace the non transmitted parts of the
speech or audio signal. Use of [RFC3389] Comfort Noise (CN) with speech or audio signal. Use of [RFC3389] Comfort Noise (CN) with
Opus is discouraged. The transmitter MUST drop whole frames only, Opus is discouraged. The transmitter MUST drop whole frames only,
based on the size of the last transmitted frame, to ensure successive 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 RTP timestamps differ by a multiple of 120 and to allow the receiver
to use whole frames for concealment. to use whole frames for concealment.
The DTX mode of Opus will have a slightly lower speech or audio DTX can be used with both variable and constant bitrate. It will
quality than the continuous mode. Therefore, it is RECOMMENDED to have a slightly lower speech or audio quality than continuous
use Opus in the continuous mode unless restraints on network capacity transmission. Therefore, using continuous transmission is
are severe. The DTX mode can be engaged for operation in both RECOMMENDED unless restraints on network capacity are severe.
adaptive or constant bitrate.
3.2. Complexity 3.2. Complexity
Complexity can be scaled to optimize for CPU resources in real-time, Complexity can be scaled to optimize for CPU resources in real-time,
mostly as a trade-off between audio quality and bitrate. Also, mostly as a trade-off between audio quality and bitrate. Also,
different modes of Opus have different complexity. different modes of Opus have different complexity.
3.3. Forward Error Correction (FEC) 3.3. Forward Error Correction (FEC)
The voice mode of Opus allows for "in-band" forward error correction The voice mode of Opus allows for embedding "in-band" forward error
(FEC) data to be embedded into the bit stream of Opus. This FEC correction (FEC) data into the Opus bit stream. This FEC scheme adds
scheme adds redundant information about the previous packet (n-1) to redundant information about the previous packet (N-1) to the current
the current output packet n. For each frame, the encoder decides output packet N. For each frame, the encoder decides whether to use
whether to use FEC based on (1) an externally-provided estimate of FEC based on (1) an externally-provided estimate of the channel's
the channel's packet loss rate; (2) an externally-provided estimate packet loss rate; (2) an externally-provided estimate of the
of the channel's capacity; (3) the sensitivity of the audio or speech channel's capacity; (3) the sensitivity of the audio or speech signal
signal to packet loss; (4) whether the receiving decoder has to packet loss; (4) whether the receiving decoder has indicated it
indicated it can take advantage of "in-band" FEC information. The can take advantage of "in-band" FEC information. The decision to
decision to send "in-band" FEC information is entirely controlled by send "in-band" FEC information is entirely controlled by the encoder
the encoder and therefore no special precautions for the payload have and therefore no special precautions for the payload have to be
to be taken. taken.
On the receiving side, the decoder can take advantage of this On the receiving side, the decoder can take advantage of this
additional information when, in case of a packet loss, the next additional information when it loses a packet and the next packet is
packet is available. In order to use the FEC data, the jitter buffer available. In order to use the FEC data, the jitter buffer needs to
needs to provide access to payloads with the FEC data. The decoder provide access to payloads with the FEC data. The receiver can then
API function has a flag to indicate that a FEC frame rather than a configure its decoder to decode the FEC data from the packet rather
regular frame should be decoded. If no FEC data is available for the than the regular audio data. If no FEC data is available for the
current frame, the decoder will consider the frame lost and invokes current frame, the decoder will consider the frame lost and invoke
the frame loss concealment. frame loss concealment.
If the FEC scheme is not implemented on the receiving side, FEC 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 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 resources. Decoder support for FEC SHOULD be indicated at the time a
session is set up. session is set up.
3.4. Stereo Operation 3.4. Stereo Operation
Opus allows for transmission of stereo audio signals. This operation Opus allows for transmission of stereo audio signals. This operation
is signaled in-band in the Opus payload and no special arrangement is is signaled in-band in the Opus payload and no special arrangement is
required in the payload format. Any implementation of the Opus needed in the payload format. Any implementation of the Opus decoder
decoder MUST be capable of receiving stereo signals, although it MAY MUST be capable of receiving stereo signals, although it MAY decode
decode those signals as mono. those signals as mono.
If a decoder can not take advantage of the benefits of a stereo 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 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 that case the sending side SHOULD NOT send stereo signals as it leads
to an inefficient usage of the network. to an inefficient usage of network resources.
4. Opus RTP Payload Format 4. Opus RTP Payload Format
The payload format for Opus consists of the RTP header and Opus The payload format for Opus consists of the RTP header and Opus
payload data. payload data.
4.1. RTP Header Usage 4.1. RTP Header Usage
The format of the RTP header is specified in [RFC3550]. The Opus The format of the RTP header is specified in [RFC3550]. The use of
payload format uses the fields of the RTP header consistent with this the fields of the RTP header by the Opus payload format is consistent
specification. with that specification.
The payload length of Opus is a multiple number of octets and The payload length of Opus is an integer number of octets and
therefore no padding is required. The payload MAY be padded by an therefore no padding is necessary. The payload MAY be padded by an
integer number of octets according to [RFC3550]. integer number of octets according to [RFC3550].
The timestamp, sequence number, and marker bit (M) of the RTP header The timestamp, sequence number, and marker bit (M) of the RTP header
are used in accordance with Section 4.1 of [RFC3551]. are used in accordance with Section 4.1 of [RFC3551].
The RTP payload type for Opus has not been assigned statically and is The RTP payload type for Opus has not been assigned statically and is
expected to be assigned dynamically. expected to be assigned dynamically.
The receiving side MUST be prepared to receive duplicates of RTP The receiving side MUST be prepared to receive duplicate RTP packets.
packets. Only one of those payloads MUST be provided to the Opus The receiver MUST provide at most one of those payloads to the Opus
decoder for decoding and others MUST be discarded. decoder for decoding, and MUST discard the others.
Opus supports 5 different audio bandwidths which may be adjusted Opus supports 5 different audio bandwidths, which can be adjusted
during the duration of a call. The RTP timestamp clock frequency is during a call. The RTP timestamp is incremented with a 48000 Hz
defined as the highest supported sampling frequency of Opus, i.e. clock rate for all modes of Opus and all sampling rates. The unit
48000 Hz, for all modes and sampling rates of Opus. The unit for the for the timestamp is samples per single (mono) channel. The RTP
timestamp is samples per single (mono) channel. The RTP timestamp timestamp corresponds to the sample time of the first encoded sample
corresponds to the sample time of the first encoded sample in the in the encoded frame. For data encoded with sampling rates other
encoded frame. For sampling rates lower than 48000 Hz the number of than 48000 Hz, the sampling rate has to be adjusted to 48000 Hz using
samples has to be multiplied with a multiplier according to Table 2 the corresponding multiplier in Table 2.
to determine the RTP timestamp.
+---------+------------+ +--------------------+------------+
| fs (Hz) | Multiplier | | Sampling Rate (Hz) | Multiplier |
+---------+------------+ +--------------------+------------+
| 8000 | 6 | | 8000 | 6 |
| | | | | |
| 12000 | 4 | | 12000 | 4 |
| | | | | |
| 16000 | 3 | | 16000 | 3 |
| | | | | |
| 24000 | 2 | | 24000 | 2 |
| | | | | |
| 48000 | 1 | | 48000 | 1 |
+---------+------------+ +--------------------+------------+
Table 2: Timestamp multiplier Table 2: Timestamp multiplier
4.2. Payload Structure 4.2. Payload Structure
The Opus encoder can be set to output encoded frames representing The Opus encoder can output encoded frames representing 2.5, 5, 10,
2.5, 5, 10, 20, 40, or 60 ms of speech or audio data. Further, an 20, 40, or 60 ms of speech or audio data. Further, an arbitrary
arbitrary number of frames can be combined into a packet. The number of frames can be combined into a packet, up to a maximum
maximum packet length is limited to the amount of encoded data packet duration representing 120 ms of speech or audio data. The
representing 120 ms of speech or audio data. The packetization of grouping of one or more Opus frames into a single Opus packet is
encoded data is purely done by the Opus encoder and therefore only defined in Section 3 of [RFC6716]. An RTP payload MUST contain
one packet output from the Opus encoder MUST be used as a payload. exactly one Opus packet as defined by that document.
Figure 1 shows the structure combined with the RTP header. Figure 1 shows the structure combined with the RTP header.
+----------+--------------+ +----------+--------------+
|RTP Header| Opus Payload | |RTP Header| Opus Payload |
+----------+--------------+ +----------+--------------+
Figure 1: Payload Structure with RTP header Figure 1: Payload Structure with RTP header
Table 3 shows supported frame sizes in milliseconds of encoded speech Table 3 shows supported frame sizes in milliseconds of encoded speech
or audio data for speech and audio mode (Mode) and sampling rates or audio data for the speech and audio modes (Mode) and sampling
(fs) of Opus and how the timestamp needs to be incremented for rates (fs) of Opus and shows how the timestamp is incremented for
packetization (ts incr). If the Opus encoder outputs multiple packetization (ts incr). If the Opus encoder outputs multiple
encoded frames into a single packet the timestamps have to be added encoded frames into a single packet, the timestamp increment is the
up according to the combined frames. sum of the increments for the individual frames.
+---------+-----------------+-----+-----+-----+-----+------+------+ +---------+-----------------+-----+-----+-----+-----+------+------+
| Mode | fs | 2.5 | 5 | 10 | 20 | 40 | 60 | | Mode | fs | 2.5 | 5 | 10 | 20 | 40 | 60 |
+---------+-----------------+-----+-----+-----+-----+------+------+ +---------+-----------------+-----+-----+-----+-----+------+------+
| ts incr | all | 120 | 240 | 480 | 960 | 1920 | 2880 | | ts incr | all | 120 | 240 | 480 | 960 | 1920 | 2880 |
| | | | | | | | | | | | | | | | | |
| voice | nb/mb/wb/swb/fb | | | x | x | x | x | | voice | NB/MB/WB/SWB/FB | | | x | x | x | x |
| | | | | | | | | | | | | | | | | |
| audio | nb/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 Table 3: Supported Opus frame sizes and timestamp increments
5. Congestion Control 5. Congestion Control
The adaptive nature of the Opus codec allows for an efficient The target bitrate of Opus can be adjusted at any point in time, thus
congestion control. allowing efficient congestion control. Furthermore, the amount of
encoded speech or audio data encoded in a single packet can be used
The target bitrate of Opus can be adjusted at any point in time and for congestion control, since the transmission rate is inversely
thus allowing for an efficient congestion control. Furthermore, the proportional to the packet duration. A lower packet transmission
amount of encoded speech or audio data encoded in a single packet can rate reduces the amount of header overhead, but at the same time
be used for congestion control since the transmission rate is increases latency and loss sensitivity, so it ought to be used with
inversely proportional to these frame sizes. A lower packet care.
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 It is RECOMMENDED that senders of Opus encoded data apply congestion
transmission of Opus encoded data. control.
6. IANA Considerations 6. IANA Considerations
One media subtype (audio/opus) has been defined and registered as One media subtype (audio/opus) has been defined and registered as
described in the following section. described in the following section.
6.1. Opus Media Type Registration 6.1. Opus Media Type Registration
Media type registration is done according to [RFC4288] and [RFC4855]. Media type registration is done according to [RFC4288] and [RFC4855].
Type name: audio Type name: audio
Subtype name: opus Subtype name: opus
Required parameters: Required parameters:
rate: RTP timestamp clock rate is incremented with 48000 Hz clock rate: the RTP timestamp is incremented with a 48000 Hz clock rate
rate for all modes of Opus and all sampling frequencies. For for all modes of Opus and all sampling rates. For data encoded
audio sampling rates other than 48000 Hz the rate has to be with sampling rates other than 48000 Hz, the sampling rate has to
adjusted to 48000 Hz according to Table 2. be adjusted to 48000 Hz using the corresponding multiplier in
Table 2.
Optional parameters: Optional parameters:
maxplaybackrate: a hint about the maximum output sampling rate that maxplaybackrate: a hint about the maximum output sampling rate that
the receiver is capable of rendering in Hz. The decoder MUST be the receiver is capable of rendering in Hz. The decoder MUST be
capable of decoding any audio bandwidth but due to hardware capable of decoding any audio bandwidth but due to hardware
limitations only signals up to the specified sampling rate can be limitations only signals up to the specified sampling rate can be
played back. Sending signals with higher audio bandwidth results played back. Sending signals with higher audio bandwidth results
in higher than necessary network usage and encoding complexity, so in higher than necessary network usage and encoding complexity, so
an encoder SHOULD NOT encode frequencies above the audio bandwidth an encoder SHOULD NOT encode frequencies above the audio bandwidth
skipping to change at page 10, line 14 skipping to change at page 9, line 44
sprop-maxcapturerate: a hint about the maximum input sampling rate 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 is likely to produce. This is not a guarantee
that the sender will never send any higher bandwidth (e.g. it that the sender will never send any higher bandwidth (e.g. it
could send a pre-recorded prompt that uses a higher bandwidth), could send a pre-recorded prompt that uses a higher bandwidth),
but it indicates to the receiver that frequencies above this but it indicates to the receiver that frequencies above this
maximum can safely be discarded. This parameter is useful to maximum can safely be discarded. This parameter is useful to
avoid wasting receiver resources by operating the audio processing avoid wasting receiver resources by operating the audio processing
pipeline (e.g. echo cancellation) at a higher rate than necessary. pipeline (e.g. echo cancellation) at a higher rate than necessary.
This parameter can take any value between 8000 and 48000, although This parameter can take any value between 8000 and 48000, although
commonly the value will match one of the Opus bandwidths (Table commonly the value will match one of the Opus bandwidths
1). By default, the sender is assumed to have no limitations, (Table 1). By default, the sender is assumed to have no
i.e. 48000. limitations, i.e. 48000.
maxptime: the decoder's maximum length of time in milliseconds maxptime: the maximum duration of media represented by a packet
rounded up to the next full integer value represented by the media (according to Section 6 of [RFC4566]) that a decoder wants to
in a packet that can be encapsulated in a received packet receive, in milliseconds rounded up to the next full integer
according to Section 6 of [RFC4566]. Possible values are 3, 5, value. Possible values are 3, 5, 10, 20, 40, 60, or an arbitrary
10, 20, 40, and 60 or an arbitrary multiple of Opus frame sizes multiple of an Opus frame size rounded up to the next full integer
rounded up to the next full integer value up to a maximum value of value, up to a maximum value of 120, as defined in Section 4. If
120 as defined in Section 4. If no value is specified, 120 is no value is specified, the default is 120. This value is a
assumed as default. This value is a recommendation by the recommendation by the decoding side to ensure the best performance
for the decoder. The decoder MUST be capable of accepting any
allowed packet sizes to ensure maximum compatibility.
ptime: the preferred duration of media represented by a packet
(according to Section 6 of [RFC4566]) that a decoder wants to
receive, in milliseconds rounded up to the next full integer
value. Possible values are 3, 5, 10, 20, 40, 60, or an arbitrary
multiple of an Opus frame size rounded up to the next full integer
value, up to a maximum value of 120, as defined in Section 4. If
no value is specified, the default is 20. If ptime is greater
than maxptime, ptime MUST be ignored. This parameter MAY be
changed during a session. This value is a recommendation by the
decoding side to ensure the best performance for the decoder. The decoding side to ensure the best performance for the decoder. The
decoder MUST be capable of accepting any allowed packet sizes to decoder MUST be capable of accepting any allowed packet sizes to
ensure maximum compatibility. ensure maximum compatibility.
ptime: the decoder's recommended length of time in milliseconds minptime: the minimum duration of media represented by a packet
rounded up to the next full integer value represented by the media (according to Section 6 of [RFC4566]) that SHOULD be encapsulated
in a packet according to Section 6 of [RFC4566]. Possible values in a received packet, in milliseconds rounded up to the next full
are 3, 5, 10, 20, 40, or 60 or an arbitrary multiple of Opus frame integer value. Possible values are 3, 5, 10, 20, 40, and 60 or an
sizes rounded up to the next full integer value up to a maximum arbitrary multiple of Opus frame sizes rounded up to the next full
value of 120 as defined in Section 4. If no value is specified, integer value up to a maximum value of 120 as defined in
20 is assumed as default. If ptime is greater than maxptime, Section 4. If no value is specified, the default is 3. This
ptime MUST be ignored. This parameter MAY be changed during a value is a recommendation by the decoding side to ensure the best
session. This value is a recommendation by the decoding side to performance for the decoder. The decoder MUST be capable to
ensure the best performance for the decoder. The decoder MUST be accept any allowed packet sizes to ensure maximum compatibility.
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 maxaveragebitrate: specifies the maximum average receive bitrate of
a session in bits per second (b/s). The actual value of the 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 bitrate can vary, as it is dependent on the characteristics of the
media in a packet. Note that the maximum average bitrate MAY be media in a packet. Note that the maximum average bitrate MAY be
modified dynamically during a session. Any positive integer is modified dynamically during a session. Any positive integer is
allowed but values outside the range between 6000 and 510000 allowed, but values outside the range 6000 to 510000 SHOULD be
SHOULD be ignored. If no value is specified, the maximum value ignored. If no value is specified, the maximum value specified in
specified in Section 3.1.1 for the corresponding mode of Opus and Section 3.1.1 for the corresponding mode of Opus and corresponding
corresponding maxplaybackrate: will be the default. maxplaybackrate is the default.
stereo: specifies whether the decoder prefers receiving stereo or stereo: specifies whether the decoder prefers receiving stereo or
mono signals. Possible values are 1 and 0 where 1 specifies that mono signals. Possible values are 1 and 0 where 1 specifies that
stereo signals are preferred and 0 specifies that only mono stereo signals are preferred, and 0 specifies that only mono
signals are preferred. Independent of the stereo parameter every signals are preferred. Independent of the stereo parameter every
receiver MUST be able to receive and decode stereo signals but receiver MUST be able to receive and decode stereo signals but
sending stereo signals to a receiver that signaled a preference sending stereo signals to a receiver that signaled a preference
for mono signals may result in higher than necessary network for mono signals may result in higher than necessary network
utilisation and encoding complexity. If no value is specified, utilization and encoding complexity. If no value is specified,
mono is assumed (stereo=0). the default is 0 (mono).
sprop-stereo: specifies whether the sender is likely to produce sprop-stereo: specifies whether the sender is likely to produce
stereo audio. Possible values are 1 and 0 where 1 specifies that stereo audio. Possible values are 1 and 0, where 1 specifies that
stereo signals are likely to be sent, and 0 speficies that the stereo signals are likely to be sent, and 0 specifies that the
sender will likely only send mono. This is not a guarantee that 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- the sender will never send stereo audio (e.g. it could send a pre-
recorded prompt that uses stereo), but it indicates to the recorded prompt that uses stereo), but it indicates to the
receiver that the received signal can be safely downmixed to mono. receiver that the received signal can be safely downmixed to mono.
This parameter is useful to avoid wasting receiver resources by This parameter is useful to avoid wasting receiver resources by
operating the audio processing pipeline (e.g. echo cancellation) operating the audio processing pipeline (e.g. echo cancellation)
in stereo when not necessary. If no value is specified, mono is in stereo when not necessary. If no value is specified, the
assumed (sprop-stereo=0). default is 0 (mono).
cbr: specifies if the decoder prefers the use of a constant bitrate cbr: specifies if the decoder prefers the use of a constant bitrate
versus variable bitrate. Possible values are 1 and 0 where 1 versus variable bitrate. Possible values are 1 and 0, where 1
specifies constant bitrate and 0 specifies variable bitrate. If specifies constant bitrate and 0 specifies variable bitrate. If
no value is specified, cbr is assumed to be 0. Note that the no value is specified, the default is 0 (vbr). When cbr is 1, the
maximum average bitrate may still be changed, e.g. to adapt to maximum average bitrate can still change, e.g. to adapt to
changing network conditions. changing network conditions.
useinbandfec: specifies that the decoder has the capability to take useinbandfec: specifies that the decoder has the capability to take
advantage of the Opus in-band FEC. Possible values are 1 and 0. 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 Providing 0 when FEC cannot be used on the receiving side is
the receiving side. If no value is specified, useinbandfec is RECOMMENDED. If no value is specified, useinbandfec is assumed to
assumed to be 0. This parameter is only a preference and the be 0. This parameter is only a preference and the receiver MUST
receiver MUST be able to process packets that include FEC be able to process packets that include FEC information, even if
information, even if it means the FEC part is discarded. it means the FEC part is discarded.
usedtx: specifies if the decoder prefers the use of DTX. Possible usedtx: specifies if the decoder prefers the use of DTX. Possible
values are 1 and 0. If no value is specified, usedtx is assumed values are 1 and 0. If no value is specified, the default is 0.
to be 0.
Encoding considerations: Encoding considerations:
Opus media type is framed and consists of binary data according to The Opus media type is framed and consists of binary data
Section 4.8 in [RFC4288]. according to Section 4.8 in [RFC4288].
Security considerations: Security considerations:
See Section 7 of this document. See Section 7 of this document.
Interoperability considerations: none Interoperability considerations: none
Published specification: none Published specification: none
Applications that use this media type: Applications that use this media type:
Any application that requires the transport of speech or audio Any application that requires the transport of speech or audio
data may use this media type. Some examples are, but not limited data can use this media type. Some examples are, but not limited
to, audio and video conferencing, Voice over IP, media streaming. to, audio and video conferencing, Voice over IP, media streaming.
Person & email address to contact for further information: Person & email address to contact for further information:
SILK Support silksupport@skype.net SILK Support silksupport@skype.net
Jean-Marc Valin jmvalin@jmvalin.ca Jean-Marc Valin jmvalin@jmvalin.ca
Intended usage: COMMON Intended usage: COMMON
Restrictions on usage: Restrictions on usage:
skipping to change at page 14, line 28 skipping to change at page 13, line 42
Below are some examples of SDP session descriptions for Opus: Below are some examples of SDP session descriptions for Opus:
Example 1: Standard mono session with 48000 Hz clock rate Example 1: Standard mono session with 48000 Hz clock rate
m=audio 54312 RTP/AVP 101 m=audio 54312 RTP/AVP 101
a=rtpmap:101 opus/48000/2 a=rtpmap:101 opus/48000/2
Example 2: 16000 Hz clock rate, maximum packet size of 40 ms, Example 2: 16000 Hz clock rate, maximum packet size of 40 ms,
recommended packet size of 40 ms, maximum average bitrate of 20000 recommended packet size of 40 ms, maximum average bitrate of 20000
bps, prefers to receive stereo but only plans to send mono, FEC is bps, prefers to receive stereo but only plans to send mono, FEC is
allowed, DTX is not desired desired, DTX is not desired
m=audio 54312 RTP/AVP 101 m=audio 54312 RTP/AVP 101
a=rtpmap:101 opus/48000/2 a=rtpmap:101 opus/48000/2
a=fmtp:101 maxplaybackrate=16000; sprop-maxcapturerate=16000; a=fmtp:101 maxplaybackrate=16000; sprop-maxcapturerate=16000;
maxaveragebitrate=20000; stereo=1; useinbandfec=1; usedtx=0 maxaveragebitrate=20000; stereo=1; useinbandfec=1; usedtx=0
a=ptime:40 a=ptime:40
a=maxptime:40 a=maxptime:40
Example 3: Two-way full-band stereo preferred Example 3: Two-way full-band stereo preferred
m=audio 54312 RTP/AVP 101 m=audio 54312 RTP/AVP 101
skipping to change at page 15, line 13 skipping to change at page 14, line 25
a=fmtp:101 stereo=1; sprop-stereo=1 a=fmtp:101 stereo=1; sprop-stereo=1
6.2.1. Offer-Answer Model Considerations for Opus 6.2.1. Offer-Answer Model Considerations for Opus
When using the offer-answer procedure described in [RFC3264] to When using the offer-answer procedure described in [RFC3264] to
negotiate the use of Opus, the following considerations apply: negotiate the use of Opus, the following considerations apply:
o Opus supports several clock rates. For signaling purposes only o Opus supports several clock rates. For signaling purposes only
the highest, i.e. 48000, is used. The actual clock rate of the the highest, i.e. 48000, is used. The actual clock rate of the
corresponding media is signaled inside the payload and is not corresponding media is signaled inside the payload and is not
subject to this payload format description. The decoder MUST be restricted by this payload format description. The decoder MUST
capable to decode every received clock rate. An example is shown be capable of decoding every received clock rate. An example is
below: shown below:
m=audio 54312 RTP/AVP 100 m=audio 54312 RTP/AVP 100
a=rtpmap:100 opus/48000/2 a=rtpmap:100 opus/48000/2
o The "ptime" and "maxptime" parameters are unidirectional receive- o The "ptime" and "maxptime" parameters are unidirectional receive-
only parameters and typically will not compromise only parameters and typically will not compromise
interoperability; however, dependent on the set values of the interoperability; however, some values might cause application
parameters the performance of the application may suffer. performance to suffer. [RFC3264] defines the SDP offer-answer
[RFC3264] defines the SDP offer-answer handling of the "ptime" handling of the "ptime" parameter. The "maxptime" parameter MUST
parameter. The "maxptime" parameter MUST be handled in the same be handled in the same way.
way.
o The "minptime" parameter is a unidirectional receive-only o The "minptime" parameter is a unidirectional receive-only
parameters and typically will not compromise interoperability; parameters and typically will not compromise interoperability;
however, dependent on the set values of the parameter the however, some values might cause application performance to suffer
performance of the application may suffer and should be set with and ought to be used with care.
care.
o The "maxplaybackrate" parameter is a unidirectional receive-only o The "maxplaybackrate" parameter is a unidirectional receive-only
parameter that reflects limitations of the local receiver. The parameter that reflects limitations of the local receiver. When
sender of the other side SHOULD NOT send with an audio bandwidth sending to a single destination, a sender MUST NOT use an audio
higher than "maxplaybackrate" as this would lead to inefficient bandwidth higher than necessary to make full use of audio sampled
use of network resources. The "maxplaybackrate" parameter does at a sampling rate of "maxplaybackrate". Gateways or senders that
not affect interoperability. Also, this parameter SHOULD NOT be are sending the same encoded audio to multiple destinations SHOULD
used to adjust the audio bandwidth as a function of the bitrates, NOT use an audio bandwidth higher than necessary to represent
as this is the responsibility of the Opus encoder implementation. 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 o The "maxaveragebitrate" parameter is a unidirectional receive-only
parameter that reflects limitations of the local receiver. The parameter that reflects limitations of the local receiver. The
sender of the other side MUST NOT send with an average bitrate sender of the other side MUST NOT send with an average bitrate
higher than "maxaveragebitrate" as it might overload the network higher than "maxaveragebitrate" as it might overload the network
and/or receiver. The "maxaveragebitrate" parameter typically will and/or receiver. The "maxaveragebitrate" parameter typically will
not compromise interoperability; however, dependent on the set not compromise interoperability; however, some values might cause
value of the parameter the performance of the application may application performance to suffer, and ought to be set with care.
suffer and should be set with care.
o The "sprop-maxcapturerate" and "sprop-stereo" parameters are o The "sprop-maxcapturerate" and "sprop-stereo" parameters are
unidirectional sender-only parameters that reflect limitations of unidirectional sender-only parameters that reflect limitations of
the sender side. They allow the receiver to set up a reduced- the sender side. They allow the receiver to set up a reduced-
complexity audio processing pipeline if the sender is not planning complexity audio processing pipeline if the sender is not planning
to use the full range of Opus's capabilities. Neither "sprop- to use the full range of Opus's capabilities. Neither "sprop-
maxcapturerate" nor "sprop-stereo" affect interoperability and the maxcapturerate" nor "sprop-stereo" affect interoperability and the
receiver MUST be capable of receiving any signal. receiver MUST be capable of receiving any signal.
o The "stereo" parameter is a unidirectional receive-only parameter. 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 "cbr" parameter is a unidirectional receive-only parameter.
o The "useinbandfec" parameter is a unidirectional receive-only o The "useinbandfec" parameter is a unidirectional receive-only
parameter. parameter.
o The "usedtx" 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 o Any unknown parameter in an offer MUST be ignored by the receiver
and MUST be removed from the answer. and MUST be removed from the answer.
6.2.2. Declarative SDP Considerations for Opus 6.2.2. Declarative SDP Considerations for Opus
For declarative use of SDP such as in Session Announcement Protocol For declarative use of SDP such as in Session Announcement Protocol
(SAP), [RFC2974], and RTSP, [RFC2326], for Opus, the following needs (SAP), [RFC2974], and RTSP, [RFC2326], for Opus, the following needs
to be considered: to be considered:
o The values for "maxptime", "ptime", "minptime", "maxplaybackrate", o The values for "maxptime", "ptime", "minptime", "maxplaybackrate",
and "maxaveragebitrate" should be selected carefully to ensure and "maxaveragebitrate" ought to be selected carefully to ensure
that a reasonable performance can be achieved for the participants that a reasonable performance can be achieved for the participants
of a session. of a session.
o The values for "maxptime", "ptime", and "minptime" of the payload o The values for "maxptime", "ptime", and "minptime" of the payload
format configuration are recommendations by the decoding side to format configuration are recommendations by the decoding side to
ensure the best performance for the decoder. The decoder MUST be ensure the best performance for the decoder. The decoder MUST be
capable to accept any allowed packet sizes to ensure maximum capable of accepting any allowed packet sizes to ensure maximum
compatibility. compatibility.
o All other parameters of the payload format configuration are o All other parameters of the payload format configuration are
declarative and a participant MUST use the configurations that are declarative and a participant MUST use the configurations that are
provided for the session. More than one configuration may be provided for the session. More than one configuration can be
provided if necessary by declaring multiple RTP payload types; provided if necessary by declaring multiple RTP payload types;
however, the number of types should be kept small. however, the number of types ought to be kept small.
7. Security Considerations 7. Security Considerations
All RTP packets using the payload format defined in this All RTP packets using the payload format defined in this
specification are subject to the general security considerations specification are subject to the general security considerations
discussed in the RTP specification [RFC3550] and any profile from discussed in the RTP specification [RFC3550] and any profile from,
e.g. [RFC3711] or [RFC3551]. e.g., [RFC3711] or [RFC3551].
This payload format transports Opus encoded speech or audio data, This payload format transports Opus encoded speech or audio data.
hence, security issues include confidentiality, integrity protection, Hence, security issues include confidentiality, integrity protection,
and authentication of the speech or audio itself. The Opus payload and authentication of the speech or audio itself. The Opus payload
format does not have any built-in security mechanisms. Any suitable format does not have any built-in security mechanisms. Any suitable
external mechanisms, such as SRTP [RFC3711], MAY be used. external mechanisms, such as SRTP [RFC3711], MAY be used.
This payload format and the Opus encoding do not exhibit any This payload format and the Opus encoding do not exhibit any
significant non-uniformity in the receiver-end computational load and significant non-uniformity in the receiver-end computational load and
thus are unlikely to pose a denial-of-service threat due to the thus are unlikely to pose a denial-of-service threat due to the
receipt of pathological datagrams. receipt of pathological datagrams.
8. Acknowledgements 8. Acknowledgements
skipping to change at page 18, line 26 skipping to change at page 18, line 4
Authors' Addresses Authors' Addresses
Julian Spittka Julian Spittka
Email: jspittka@gmail.com Email: jspittka@gmail.com
Koen Vos Koen Vos
vocTone vocTone
Email: koenvos74@gmail.com Email: koenvos74@gmail.com
Jean-Marc Valin Jean-Marc Valin
Mozilla Mozilla
2 Harrison Street 331 E. Evelyn Avenue
San Francisco, CA 94105 Mountain View, CA 94041
USA USA
Email: jmvalin@jmvalin.ca Email: jmvalin@jmvalin.ca
 End of changes. 70 change blocks. 
252 lines changed or deleted 254 lines changed or added

This html diff was produced by rfcdiff 1.41. The latest version is available from http://tools.ietf.org/tools/rfcdiff/