Internet Engineering Task Force (IETF) J. Spittka Request for Comments: 7587 Category: Standards Track K. Vos ISSN: 2070-1721 vocTone JM. Valin Mozilla June 2015
RTP Payload Format for the Opus Speech and Audio Codec
Abstract
This document defines the Real-time Transport Protocol (RTP) payload format for packetization of Opus-encoded speech and audio data necessary to integrate the codec in the most compatible way. It also provides an applicability statement for the use of Opus over RTP. Further, it describes media type registrations for the RTP payload format.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Further information on Internet Standards is available in Section 2 of RFC 5741.
Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at http://www.rfc-editor.org/info/rfc7587.
Copyright Notice
Copyright (c) 2015 IETF Trust and the persons identified as the document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
Opus [RFC6716] is a speech and audio codec developed within the IETF Internet Wideband Audio Codec working group. The codec has a very low algorithmic delay, and it is highly scalable in terms of audio bandwidth, bitrate, and complexity. Further, it provides different modes to efficiently encode speech signals as well as music signals, thus making it the codec of choice for various applications using the Internet or similar networks.
This document defines the Real-time Transport Protocol (RTP) [RFC3550] payload format for packetization of Opus-encoded speech and audio data necessary to integrate Opus in the most compatible way. It also provides an applicability statement for the use of Opus over RTP. Further, it describes media type registrations for the RTP payload format.
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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].
audio bandwidth: The range of audio frequencies being coded
CBR: Constant bitrate
CPU: Central Processing Unit
DTX: Discontinuous Transmission
FEC: Forward Error Correction
IP: Internet Protocol
samples: Speech or audio samples (per channel)
SDP: Session Description Protocol
SSRC: Synchronization source
VBR: Variable bitrate
Throughout this document, we refer to the following definitions:
Opus encodes speech signals as well as general audio signals. Two different modes can be chosen, a voice mode or an audio mode, to allow the most efficient coding depending on the type of the input signal, the sampling frequency of the input signal, and the intended application.
The voice mode allows efficient encoding of voice signals at lower bitrates while the audio mode is optimized for general audio signals at medium and higher bitrates.
Opus is highly scalable in terms of audio bandwidth, bitrate, and complexity. Further, Opus allows transmitting stereo signals with in-band signaling in the bitstream.
Opus supports bitrates from 6 kbit/s to 510 kbit/s. The bitrate can be changed dynamically within that range. All other parameters being equal, higher bitrates result in higher audio quality.
For the same average bitrate, variable bitrate (VBR) can achieve higher audio quality than constant bitrate (CBR). For the majority of voice transmission applications, VBR is the best choice. One reason for choosing CBR is the potential information leak that _might_ occur when encrypting the compressed stream. See [RFC6562] for guidelines on when VBR is appropriate for encrypted audio communications. In the case where an existing VBR stream needs to be converted to CBR for security reasons, the Opus padding mechanism
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described in [RFC6716] is the RECOMMENDED way to achieve padding because the RTP padding bit is unencrypted.
The bitrate can be adjusted at any point in time. To avoid congestion, the average bitrate SHOULD NOT exceed the available network bandwidth. If no target bitrate is specified, the bitrates specified in Section 3.1.1 are RECOMMENDED.
Opus can, as described in Section 3.1.2, be operated with a variable bitrate. In that case, the encoder will automatically reduce the bitrate for certain input signals, like periods of silence. When using continuous transmission, it will reduce the bitrate when the characteristics of the input signal permit, but it will never interrupt the transmission to the receiver. Therefore, the received signal will maintain the same high level of audio quality over the full duration of a transmission while minimizing the average bitrate over time.
In cases where the bitrate of Opus needs to be reduced even further or in cases where only constant bitrate is available, the Opus encoder can use Discontinuous Transmission (DTX), where parts of the encoded signal that correspond to periods of silence in the input speech or audio signal are not transmitted to the receiver. A receiver can distinguish between DTX and packet loss by looking for gaps in the sequence number, as described by Section 4.1 of [RFC3551].
On the receiving side, the non-transmitted parts will be handled by a frame loss concealment unit in the Opus decoder, which generates a comfort noise signal to replace the non-transmitted parts of the speech or audio signal. Using Comfort Noise as defined in [RFC3389] with Opus is discouraged. The transmitter MUST drop whole frames only, based on the size of the last transmitted frame, to ensure successive RTP timestamps differ by a multiple of 120 and to allow the receiver to use whole frames for concealment.
DTX can be used with both variable and constant bitrate. It will have a slightly lower speech or audio quality than continuous transmission. Therefore, using continuous transmission is RECOMMENDED unless constraints on available network bandwidth are severe.
Complexity of the encoder can be scaled to optimize for CPU resources in real time, mostly as a trade-off between audio quality and bitrate. Also, different modes of Opus have different complexity.
The voice mode of Opus allows for embedding in-band Forward Error Correction (FEC) data into the Opus bitstream. This FEC scheme adds redundant information about the previous packet (N-1) to the current output packet N. For each frame, the encoder decides whether to use FEC based on (1) an externally provided estimate of the channel's packet loss rate; (2) an externally provided estimate of the channel's capacity; (3) the sensitivity of the audio or speech signal to packet loss; and (4) whether the receiving decoder has indicated it can take advantage of in-band FEC information. The decision to send in-band FEC information is entirely controlled by the encoder; therefore, no special precautions for the payload have to be taken.
On the receiving side, the decoder can take advantage of this additional information when it loses a packet and the next packet is available. In order to use the FEC data, the jitter buffer needs to provide access to payloads with the FEC data. Instead of performing loss concealment for a missing packet, the receiver can then configure its decoder to decode the FEC data from the next packet.
Any compliant Opus decoder is capable of ignoring FEC information when it is not needed, so encoding with FEC cannot cause interoperability problems. However, if FEC cannot be used on the receiving side, then FEC SHOULD NOT be used, as it leads to an inefficient usage of network resources. Decoder support for FEC SHOULD be indicated at the time a session is set up.
Opus allows for transmission of stereo audio signals. This operation is signaled in-band in the Opus bitstream and no special arrangement is needed in the payload format. An Opus decoder is capable of handling a stereo encoding, but an application might only be capable of consuming a single audio channel.
If a decoder cannot take advantage of the benefits of a stereo signal, this SHOULD be indicated at the time a session is set up. In that case, the sending side SHOULD NOT send stereo signals as it leads to an inefficient usage of network resources.
The format of the RTP header is specified in [RFC3550]. The use of the fields of the RTP header by the Opus payload format is consistent with that specification.
The payload length of Opus is an integer number of octets; therefore, no padding is necessary. The payload MAY be padded by an integer number of octets according to [RFC3550], although the Opus internal padding is preferred.
The timestamp, sequence number, and marker bit (M) of the RTP header are used in accordance with Section 4.1 of [RFC3551].
The RTP payload type for Opus is to be assigned dynamically.
The receiving side MUST be prepared to receive duplicate RTP packets. The receiver MUST provide at most one of those payloads to the Opus decoder for decoding, and it MUST discard the others.
Opus supports 5 different audio bandwidths, which can be adjusted during a stream. The RTP timestamp is incremented with a 48000 Hz clock rate for all modes of Opus and all sampling rates. The unit for the timestamp is samples per single (mono) channel. The RTP timestamp corresponds to the sample time of the first encoded sample in the encoded frame. For data encoded with sampling rates other than 48000 Hz, the sampling rate has to be adjusted to 48000 Hz.
The Opus encoder can output encoded frames representing 2.5, 5, 10, 20, 40, or 60 ms of speech or audio data. Further, an arbitrary number of frames can be combined into a packet, up to a maximum packet duration representing 120 ms of speech or audio data. The grouping of one or more Opus frames into a single Opus packet is defined in Section 3 of [RFC6716]. An RTP payload MUST contain exactly one Opus packet as defined by that document.
Figure 1 shows the structure combined with the RTP header.
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+----------+--------------+ |RTP Header| Opus Payload | +----------+--------------+
Figure 1: Packet Structure with RTP Header
Table 2 shows supported frame sizes in milliseconds of encoded speech or audio data for the speech and audio modes (Mode) and sampling rates (fs) of Opus, and it shows how the timestamp is incremented for packetization (ts incr). If the Opus encoder outputs multiple encoded frames into a single packet, the timestamp increment is the sum of the increments for the individual frames.
+---------+-----------------+-----+-----+-----+-----+------+------+ | Mode | fs | 2.5 | 5 | 10 | 20 | 40 | 60 | +---------+-----------------+-----+-----+-----+-----+------+------+ | ts incr | all | 120 | 240 | 480 | 960 | 1920 | 2880 | | | | | | | | | | | voice | NB/MB/WB/SWB/FB | x | x | o | o | o | o | | | | | | | | | | | audio | NB/WB/SWB/FB | o | o | o | o | x | x | +---------+-----------------+-----+-----+-----+-----+------+------+
Table 2: Supported Opus frame sizes and timestamp increments are marked with an o. Unsupported ones are marked with an x.
The target bitrate of Opus can be adjusted at any point in time, thus allowing efficient congestion control. Furthermore, the amount of encoded speech or audio data encoded in a single packet can be used for congestion control, since the transmission rate is inversely proportional to the packet duration. A lower packet transmission rate reduces the amount of header overhead, but at the same time increases latency and loss sensitivity, so it ought to be used with care.
Since UDP does not provide congestion control, applications that use RTP over UDP SHOULD implement their own congestion control above the UDP layer [RFC5405]. Work in the RMCAT working group [rmcat] describes the interactions and conceptual interfaces necessary between the application components that relate to congestion control, including the RTP layer, the higher-level media codec control layer, and the lower-level transport interface, as well as components dedicated to congestion control functions.
Media type registration is done according to [RFC6838] and [RFC4855].
Type name: audio
Subtype name: opus
Required parameters:
rate: the RTP timestamp is incremented with a 48000 Hz clock rate for all modes of Opus and all sampling rates. For data encoded with sampling rates other than 48000 Hz, the sampling rate has to be adjusted to 48000 Hz.
Optional parameters:
maxplaybackrate: a hint about the maximum output sampling rate that the receiver is capable of rendering in Hz. The decoder MUST be capable of decoding any audio bandwidth, but, due to hardware limitations, only signals up to the specified sampling rate can be played back. Sending signals with higher audio bandwidth results in higher than necessary network usage and encoding complexity, so an encoder SHOULD NOT encode frequencies above the audio bandwidth specified by maxplaybackrate. This parameter can take any value between 8000 and 48000, although commonly the value will match one of the Opus bandwidths (Table 1). By default, the receiver is assumed to have no limitations, i.e., 48000.
sprop-maxcapturerate: a hint about the maximum input sampling rate that the sender is likely to produce. This is not a guarantee that the sender will never send any higher bandwidth (e.g., it could send a prerecorded prompt that uses a higher bandwidth), but it indicates to the receiver that frequencies above this maximum can safely be discarded. This parameter is useful to avoid wasting receiver resources by operating the audio processing pipeline (e.g., echo cancellation) at a higher rate than necessary. This parameter can take any value between 8000 and 48000, although commonly the value will match one of the Opus bandwidths (Table 1). By default, the sender is assumed to have no limitations, i.e., 48000.
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maxptime: the maximum duration of media represented by a packet (according to Section 6 of [RFC4566]) that a decoder wants to receive, in milliseconds rounded up to the next full integer value. Possible values are 3, 5, 10, 20, 40, 60, or an arbitrary multiple of an Opus frame size rounded up to the next full integer value, up to a maximum value of 120, as defined in Section 4. If no value is specified, the default is 120.
ptime: the preferred duration of media represented by a packet (according to Section 6 of [RFC4566]) that a decoder wants to receive, in milliseconds rounded up to the next full integer value. Possible values are 3, 5, 10, 20, 40, 60, or an arbitrary multiple of an Opus frame size rounded up to the next full integer value, up to a maximum value of 120, as defined in Section 4. If no value is specified, the default is 20.
maxaveragebitrate: specifies the maximum average receive bitrate of a session in bits per second (bit/s). The actual value of the bitrate can vary, as it is dependent on the characteristics of the media in a packet. Note that the maximum average bitrate MAY be modified dynamically during a session. Any positive integer is allowed, but values outside the range 6000 to 510000 SHOULD be ignored. If no value is specified, the maximum value specified in Section 3.1.1 for the corresponding mode of Opus and corresponding maxplaybackrate is the default.
stereo: specifies whether the decoder prefers receiving stereo or mono signals. Possible values are 1 and 0, where 1 specifies that stereo signals are preferred, and 0 specifies that only mono signals are preferred. Independent of the stereo parameter, every receiver MUST be able to receive and decode stereo signals, but sending stereo signals to a receiver that signaled a preference for mono signals may result in higher than necessary network utilization and encoding complexity. If no value is specified, the default is 0 (mono).
sprop-stereo: specifies whether the sender is likely to produce stereo audio. Possible values are 1 and 0, where 1 specifies that stereo signals are likely to be sent, and 0 specifies that the sender will likely only send mono. This is not a guarantee that the sender will never send stereo audio (e.g., it could send a prerecorded prompt that uses stereo), but it indicates to the receiver that the received signal can be safely downmixed to mono. This parameter is useful to avoid wasting receiver resources by operating the audio processing pipeline (e.g., echo cancellation) in stereo when not necessary. If no value is specified, the default is 0 (mono).
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cbr: specifies if the decoder prefers the use of a constant bitrate versus a variable bitrate. Possible values are 1 and 0, where 1 specifies constant bitrate, and 0 specifies variable bitrate. If no value is specified, the default is 0 (vbr). When cbr is 1, the maximum average bitrate can still change, e.g., to adapt to changing network conditions.
useinbandfec: specifies that the decoder has the capability to take advantage of the Opus in-band FEC. Possible values are 1 and 0. Providing 0 when FEC cannot be used on the receiving side is RECOMMENDED. If no value is specified, useinbandfec is assumed to be 0. This parameter is only a preference, and the receiver MUST be able to process packets that include FEC information, even if it means the FEC part is discarded.
usedtx: specifies if the decoder prefers the use of DTX. Possible values are 1 and 0. If no value is specified, the default is 0.
Encoding considerations:
The Opus media type is framed and consists of binary data according to Section 4.8 of [RFC6838].
Any application that requires the transport of speech or audio data can use this media type. Some examples are, but not limited to, audio and video conferencing, Voice over IP, and media streaming.
Fragment identifier considerations: N/A
Person & email address to contact for further information:
SILK Support, silksupport@skype.net
Jean-Marc Valin, jmvalin@jmvalin.ca
Intended usage: COMMON
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Restrictions on usage:
For transfer over RTP, the RTP payload format (Section 4 of this document) SHALL be used.
Authors:
Julian Spittka, jspittka@gmail.com
Koen Vos, koenvos74@gmail.com
Jean-Marc Valin, jmvalin@jmvalin.ca
Change controller: IETF Payload working group delegated from the IESG
The information described in the media type specification has a specific mapping to fields in the Session Description Protocol (SDP) [RFC4566], which is commonly used to describe RTP sessions. When SDP is used to specify sessions employing Opus, the mapping is as follows:
o The media type ("audio") goes in SDP "m=" as the media name.
o The media subtype ("opus") goes in SDP "a=rtpmap" as the encoding name. The RTP clock rate in "a=rtpmap" MUST be 48000, and the number of channels MUST be 2.
o The OPTIONAL media type parameters "ptime" and "maxptime" are mapped to "a=ptime" and "a=maxptime" attributes, respectively, in the SDP.
o The OPTIONAL media type parameters "maxaveragebitrate", "maxplaybackrate", "stereo", "cbr", "useinbandfec", and "usedtx", when present, MUST be included in the "a=fmtp" attribute in the SDP, expressed as a media type string in the form of a semicolon- separated list of parameter=value pairs (e.g., maxplaybackrate=48000). They MUST NOT be specified in an SSRC- specific "fmtp" source-level attribute (as defined in Section 6.3 of [RFC5576]).
o The OPTIONAL media type parameters "sprop-maxcapturerate" and "sprop-stereo" MAY be mapped to the "a=fmtp" SDP attribute by copying them directly from the media type parameter string as part of the semicolon-separated list of parameter=value pairs (e.g., sprop-stereo=1). These same OPTIONAL media type parameters MAY also be specified using an SSRC-specific "fmtp" source-level
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attribute as described in Section 6.3 of [RFC5576]. They MAY be specified in both places, in which case the parameter in the source-level attribute overrides the one found on the "a=fmtp" line. The value of any parameter that is not specified in a source-level source attribute MUST be taken from the "a=fmtp" line, if it is present there.
Below are some examples of SDP session descriptions for Opus:
Example 1: Standard mono session with 48000 Hz clock rate
Example 2: 16000 Hz clock rate, maximum packet size of 40 ms, recommended packet size of 40 ms, maximum average bitrate of 20000 bit/s, prefers to receive stereo but only plans to send mono, FEC is desired, DTX is not desired
When using the offer/answer procedure described in [RFC3264] to negotiate the use of Opus, the following considerations apply:
o Opus supports several clock rates. For signaling purposes, only the highest, i.e., 48000, is used. The actual clock rate of the corresponding media is signaled inside the payload and is not restricted by this payload format description. The decoder MUST be capable of decoding every received clock rate. An example is shown below:
o The "ptime" and "maxptime" parameters are unidirectional receive- only parameters and typically will not compromise interoperability; however, some values might cause application performance to suffer. [RFC3264] defines the SDP offer/answer handling of the "ptime" parameter. The "maxptime" parameter MUST be handled in the same way.
o The "maxplaybackrate" parameter is a unidirectional receive-only parameter that reflects limitations of the local receiver. When sending to a single destination, a sender MUST NOT use an audio bandwidth higher than necessary to make full use of audio sampled at a sampling rate of "maxplaybackrate". Gateways or senders that are sending the same encoded audio to multiple destinations SHOULD NOT use an audio bandwidth higher than necessary to represent audio sampled at "maxplaybackrate", as this would lead to inefficient use of network resources. The "maxplaybackrate" parameter does not affect interoperability. Also, this parameter SHOULD NOT be used to adjust the audio bandwidth as a function of the bitrate, as this is the responsibility of the Opus encoder implementation.
o The "maxaveragebitrate" parameter is a unidirectional receive-only parameter that reflects limitations of the local receiver. The sender of the other side MUST NOT send with an average bitrate higher than "maxaveragebitrate" as it might overload the network and/or receiver. The "maxaveragebitrate" parameter typically will not compromise interoperability; however, some values might cause application performance to suffer and ought to be set with care.
o The "sprop-maxcapturerate" and "sprop-stereo" parameters are unidirectional sender-only parameters that reflect limitations of the sender side. They allow the receiver to set up a reduced- complexity audio processing pipeline if the sender is not planning to use the full range of Opus's capabilities. Neither "sprop- maxcapturerate" nor "sprop-stereo" affect interoperability, and the receiver MUST be capable of receiving any signal.
o The "stereo" parameter is a unidirectional receive-only parameter. When sending to a single destination, a sender MUST NOT use stereo when "stereo" is 0. Gateways or senders that are sending the same encoded audio to multiple destinations SHOULD NOT use stereo when "stereo" is 0, as this would lead to inefficient use of network resources. The "stereo" parameter does not affect interoperability.
o The "cbr" parameter is a unidirectional receive-only parameter.
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o The "useinbandfec" parameter is a unidirectional receive-only parameter.
o The "usedtx" parameter is a unidirectional receive-only parameter.
o Any unknown parameter in an offer MUST be ignored by the receiver and MUST be removed from the answer.
The Opus parameters in an SDP offer/answer exchange are completely orthogonal, and there is no relationship between the SDP offer and the answer.
For declarative use of SDP such as in the Session Announcement Protocol (SAP) [RFC2974] and the Real Time Streaming Protocol (RTSP) [RFC2326] for Opus, the following needs to be considered:
o The values for "maxptime", "ptime", "maxplaybackrate", and "maxaveragebitrate" ought to be selected carefully to ensure that a reasonable performance can be achieved for the participants of a session.
o The values for "maxptime", "ptime", and of the payload format configuration are recommendations by the decoding side to ensure the best performance for the decoder.
o All other parameters of the payload format configuration are declarative and a participant MUST use the configurations that are provided for the session. More than one configuration can be provided if necessary by declaring multiple RTP payload types; however, the number of types ought to be kept small.
Use of VBR is subject to the security considerations in [RFC6562].
RTP packets using the payload format defined in this specification are subject to the security considerations discussed in the RTP specification [RFC3550] and in any applicable RTP profile such as RTP/AVP [RFC3551], RTP/AVPF [RFC4585], RTP/SAVP [RFC3711], or RTP/ SAVPF [RFC5124]. However, as "Securing the RTP Framework: Why RTP Does Not Mandate a Single Media Security Solution" [RFC7202] discusses, it is not an RTP payload format's responsibility to discuss or mandate what solutions are used to meet the basic security goals like confidentiality, integrity, and source authenticity for RTP in general. This responsibility lies on anyone using RTP in an application. They can find guidance on available security mechanisms
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and important considerations in "Options for Securing RTP Sessions" [RFC7201]. Applications SHOULD use one or more appropriate strong security mechanisms.
This payload format and the Opus encoding do not exhibit any significant non-uniformity in the receiver-end computational load and thus are unlikely to pose a denial-of-service threat due to the receipt of pathological datagrams.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, DOI 10.17487/RFC4585, July 2006, <http://www.rfc-editor.org/info/rfc4585>.
Many people have made useful comments and suggestions contributing to this document. In particular, we would like to thank Tina le Grand, Cullen Jennings, Jonathan Lennox, Gregory Maxwell, Colin Perkins, Jan Skoglund, Timothy B. Terriberry, Martin Thompson, Justin Uberti, Magnus Westerlund, and Mo Zanaty.
Authors' Addresses
Julian Spittka
Email: jspittka@gmail.com
Koen Vos vocTone
Email: koenvos74@gmail.com
Jean-Marc Valin Mozilla 331 E. Evelyn Avenue Mountain View, CA 94041 United States