Skip to content
Snippets Groups Projects
Commit f7faa90b authored by Timothy B. Terriberry's avatar Timothy B. Terriberry Committed by Jean-Marc Valin
Browse files

RTP draft edits (no normative changes). 

This is the result of an editing pass for clarity and consistency.
parent 62351fa8
No related branches found
No related tags found
No related merge requests found
Hide whitespace changes
Inline Side-by-side
doc/draft-ietf-payload-rtp-opus.xml
+ 198
− 198
View file @ f7faa90b
......@@ -76,9 +76,9 @@
<t>
This document defines the Real-time Transport Protocol (RTP) payload
format for packetization of Opus encoded
speech and audio data that is essential to integrate the codec in the
most compatible way. Further, media type registrations
are described for the RTP payload format.
speech and audio data necessary to integrate the codec in the
most compatible way. Further, it describes media type registrations
for the RTP payload format.
</t>
</abstract>
</front>
......@@ -87,19 +87,19 @@
<section title='Introduction'>
<t>
The Opus codec is a speech and audio codec developed within the
IETF Internet Wideband Audio Codec working group (codec). The codec
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
as well as music signals, thus making it the codec of choice for
various applications using the Internet or similar networks.
</t>
<t>
This document defines the Real-time Transport Protocol (RTP)
<xref target="RFC3550"/> payload format for packetization
of Opus encoded speech and audio data that is essential to
of Opus encoded speech and audio data necessary to
integrate the Opus codec in the
most compatible way. Further, media type registrations are described for
most compatible way. Further, it describes media type registrations for
the RTP payload format. More information on the Opus
codec can be obtained from <xref target="RFC6716"/>.
</t>
......@@ -115,42 +115,42 @@
<t hangText="CPU:"> Central Processing Unit</t>
<t hangText="DTX:"> Discontinuous transmission</t>
<t hangText="FEC:"> Forward error correction</t>
<t hangText="IP:"> Internet Protocol</t>
<t hangText="samples:"> Speech or audio samples (usually per channel)</t>
<t hangText="SDP:"> Session Description Protocol</t>
<t hangText="IP:"> Internet Protocol</t>
<t hangText="samples:"> Speech or audio samples (per channel)</t>
<t hangText="SDP:"> Session Description Protocol</t>
<t hangText="VBR:"> Variable bitrate</t>
</list>
</t>
<section title='Audio Bandwidth'>
<t>
Throughout this document, we refer to the following definitions:
</t>
<t>
Throughout this document, we refer to the following definitions:
</t>
<texttable anchor='bandwidth_definitions'>
<ttcol align='center'>Abbreviation</ttcol>
<ttcol align='center'>Abbreviation</ttcol>
<ttcol align='center'>Name</ttcol>
<ttcol align='center'>Bandwidth</ttcol>
<ttcol align='center'>Sampling</ttcol>
<c>nb</c>
<ttcol align='center'>Audio Bandwidth (Hz)</ttcol>
<ttcol align='center'>Sampling Rate (Hz)</ttcol>
<c>NB</c>
<c>Narrowband</c>
<c>0 - 4000</c>
<c>8000</c>
<c>mb</c>
<c>MB</c>
<c>Mediumband</c>
<c>0 - 6000</c>
<c>12000</c>
<c>wb</c>
<c>WB</c>
<c>Wideband</c>
<c>0 - 8000</c>
<c>16000</c>
<c>swb</c>
<c>SWB</c>
<c>Super-wideband</c>
<c>0 - 12000</c>
<c>24000</c>
<c>fb</c>
<c>FB</c>
<c>Fullband</c>
<c>0 - 20000</c>
<c>48000</c>
......@@ -164,16 +164,16 @@
<section title='Opus Codec'>
<t>
The Opus <xref target="RFC6716"/> speech and audio codec has been developed to encode speech
signals as well as audio signals. Two different modes, a voice mode
or an audio mode, may be chosen to allow the most efficient coding
dependent on the type of input signal, the sampling frequency of the
input signal, and the specific application.
The Opus <xref target="RFC6716"/> codec 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.
</t>
<t>
The voice mode allows efficient encoding of voice signals at lower bit
rates while the audio mode is optimized for audio signals at medium and
rates while the audio mode is optimized for general audio signals at medium and
higher bitrates.
</t>
......@@ -185,40 +185,40 @@
<section title='Network Bandwidth'>
<t>
Opus supports all bitrates from 6&nbsp;kb/s to 510&nbsp;kb/s.
The bitrate can be changed dynamically within that range.
All
other parameters being
equal, higher bitrate results in higher quality.
</t>
<section title='Recommended Bitrate' anchor='bitrate_by_bandwidth'>
<t>
For a frame size of
20&nbsp;ms, these
are the bitrate "sweet spots" for Opus in various configurations:
Opus supports all bitrates from 6&nbsp;kb/s to 510&nbsp;kb/s.
The bitrate can be changed dynamically within that range.
All
other parameters being
equal, higher bitrates result in higher quality.
</t>
<section title='Recommended Bitrate' anchor='bitrate_by_bandwidth'>
<t>
For a frame size of
20&nbsp;ms, these
are the bitrate "sweet spots" for Opus in various configurations:
<list style="symbols">
<t>8-12 kb/s for NB speech,</t>
<t>16-20 kb/s for WB speech,</t>
<t>28-40 kb/s for FB speech,</t>
<t>48-64 kb/s for FB mono music, and</t>
<t>64-128 kb/s for FB stereo music.</t>
</list>
</t>
<t>8-12 kb/s for NB speech,</t>
<t>16-20 kb/s for WB speech,</t>
<t>28-40 kb/s for FB speech,</t>
<t>48-64 kb/s for FB mono music, and</t>
<t>64-128 kb/s for FB stereo music.</t>
</list>
</t>
</section>
<section title='Variable versus Constant Bit Rate' anchor='variable-vs-constant-bitrate'>
<section title='Variable versus Constant Bitrate' anchor='variable-vs-constant-bitrate'>
<t>
For the same average bitrate, variable bitrate (VBR) can achieve higher 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 <spanx style='emph'>might</spanx> occur when encrypting the
compressed stream. See <xref target="RFC6562"/> for guidelines on when VBR is
appropriate for encrypted audio communications. In the case where an existing
VBR stream needs to be converted to CBR for security reasons, then the Opus padding
mechanism described in <xref target="RFC6716"/> is the RECOMMENDED way to achieve padding
because the RTP padding bit is unencrypted.</t>
<t>
For the same average bitrate, variable bitrate (VBR) can achieve higher quality
than constant bitrate (CBR). For the majority of voice transmission application, VBR
is the best choice. One potential reason for choosing CBR is the potential
information leak that <spanx style='emph'>may</spanx> occur when encrypting the
compressed stream. See <xref target="RFC6562"/> for guidelines on when VBR is
appropriate for encrypted audio communications. In the case where an existing
VBR stream needs to be converted to CBR for security reasons, then the Opus padding
mechanism described in <xref target="RFC6716"/> is the RECOMMENDED way to achieve padding
because the RTP padding bit is unencrypted.</t>
<t>
The bitrate can be adjusted at any point in time. To avoid congestion,
the average bitrate SHOULD be adjusted to the available
network capacity. If no target bitrate is specified, the bitrates specified in
......@@ -230,12 +230,12 @@
<section title='Discontinuous Transmission (DTX)'>
<t>
The Opus codec may, as described in <xref target='variable-vs-constant-bitrate'/>,
be operated with an adaptive bitrate. In that case, the bitrate
will automatically be reduced for certain input signals like periods
of silence. During continuous transmission the bitrate will be
reduced, when the input signal allows to do so, but the transmission
to the receiver itself will never be interrupted. Therefore, the
The Opus codec can, as described in <xref target='variable-vs-constant-bitrate'/>,
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
will never interrupt the transmission to the receiver. 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.
......@@ -244,7 +244,7 @@
<t>
In cases where the bitrate of Opus needs to be reduced even
further or in cases where only constant bitrate is available,
the Opus encoder may be set to use discontinuous
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.
......@@ -258,11 +258,11 @@
</t>
<t>
The DTX mode of Opus will have a slightly lower speech or audio
quality than the continuous mode. Therefore, it is RECOMMENDED to
use Opus in the continuous mode unless restraints on network
capacity are severe. The DTX mode can be engaged for operation
in both adaptive or constant bitrate.
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 restraints on network capacity
are severe.
</t>
</section>
......@@ -281,10 +281,10 @@
<section title="Forward Error Correction (FEC)">
<t>
The voice mode of Opus allows for "in-band" forward error correction (FEC)
data to be embedded into the bit stream of Opus. This FEC scheme adds
redundant information about the previous packet (n-1) to the current
output packet n. For
The voice mode of Opus allows for embedding "in-band" forward error correction (FEC)
data into the Opus bit stream. 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
......@@ -297,12 +297,12 @@
<t>
On the receiving side, the decoder can take advantage of this
additional information when, in case of a packet loss, the next packet
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. 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.
will consider the frame lost and invoke frame loss concealment.
</t>
<t>
......@@ -319,15 +319,15 @@
<t>
Opus allows for transmission of stereo audio signals. This operation
is signaled in-band in the Opus payload and no special arrangement
is required in the payload format. Any implementation of the Opus
is needed in the payload format. Any implementation of the Opus
decoder MUST be capable of receiving stereo signals, although it MAY
decode those signals as mono.
decode those signals as mono.
</t>
<t>
If a decoder can not take advantage of the benefits of a stereo signal
this SHOULD be indicated at the time a session is set up. In that case
the sending side SHOULD NOT send stereo signals as it leads to an
inefficient usage of the network.
inefficient usage of network resources.
</t>
</section>
......@@ -338,36 +338,37 @@
<t>The payload format for Opus consists of the RTP header and Opus payload
data.</t>
<section title='RTP Header Usage'>
<t>The format of the RTP header is specified in <xref target="RFC3550"/>. The Opus
payload format uses the fields of the RTP header consistent with this
specification.</t>
<t>The format of the RTP header is specified in <xref target="RFC3550"/>.
The use of the fields of the RTP header by the Opus payload format is
consistent with that specification.</t>
<t>The payload length of Opus is a multiple number of octets and
therefore no padding is required. The payload MAY be padded by an
<t>The payload length of Opus is an integer number of octets and
therefore no padding is necessary. The payload MAY be padded by an
integer number of octets according to <xref target="RFC3550"/>.</t>
<t>The marker bit (M) of the RTP header is used in accordance with
Section 4.1 of <xref target="RFC3551"/>.</t>
Section 4.1 of <xref target="RFC3551"/>.</t>
<t>The RTP payload type for Opus has not been assigned statically and is
expected to be assigned dynamically.</t>
<t>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.</t>
<t>The receiving side MUST be prepared to receive duplicate RTP
packets. The receiver MUST provide only one of those payloads to the
Opus decoder for decoding, and MUST discard the others.</t>
<t>Opus supports 5 different audio bandwidths which may be adjusted during
the duration of a call. The RTP timestamp clock frequency is defined as
the highest supported sampling frequency of Opus, i.e. 48000 Hz, for all
modes and sampling rates of Opus. The unit
<t>Opus supports 5 different audio bandwidths, which can be adjusted during
a call.
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 sampling
rates lower than 48000 Hz the number of samples has to be multiplied with
a multiplier according to <xref target="fs-upsample-factors"/> to determine
the RTP timestamp.</t>
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 using the
corresponding multiplier in <xref target="fs-upsample-factors"/>.</t>
<texttable anchor='fs-upsample-factors' title="Timestamp multiplier">
<ttcol align='center'>fs (Hz)</ttcol>
<ttcol align='center'>Sampling Rate (Hz)</ttcol>
<ttcol align='center'>Multiplier</ttcol>
<c>8000</c>
<c>6</c>
......@@ -384,19 +385,19 @@
<section title='Payload Structure'>
<t>
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.
The Opus encoder can output encoded frames representing 2.5, 5, 10, 20,
40, or 60&nbsp;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&nbsp;ms of speech or audio data. The packetization of encoded data
is purely done by the Opus encoder, and therefore an RTP payload MUST
contain exactly one packet output from the Opus encoder.
</t>
<t><xref target='payload-structure'/> shows the structure combined with the RTP header.</t>
<figure anchor="payload-structure"
title="Payload Structure with RTP header">
<artwork>
<artwork align="center">
<![CDATA[
+----------+--------------+
|RTP Header| Opus Payload |
......@@ -407,11 +408,11 @@
<t>
<xref target='opus-packetization'/> shows supported frame sizes in
milliseconds of encoded speech or audio data for speech and audio mode
(Mode) and sampling rates (fs) of Opus and how the timestamp needs to
be incremented for packetization (ts incr). If the Opus encoder
outputs multiple encoded frames into a single packet the timestamps
have to be added up according to the combined frames.
milliseconds of encoded speech or audio data for the speech and audio modes
(Mode) and sampling rates (fs) of Opus and 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.
</t>
<texttable anchor='opus-packetization' title="Supported Opus frame
......@@ -433,7 +434,7 @@
<c>1920</c>
<c>2880</c>
<c>voice</c>
<c>nb/mb/wb/swb/fb</c>
<c>NB/MB/WB/SWB/FB</c>
<c></c>
<c></c>
<c>x</c>
......@@ -441,7 +442,7 @@
<c>x</c>
<c>x</c>
<c>audio</c>
<c>nb/wb/swb/fb</c>
<c>NB/WB/SWB/FB</c>
<c>x</c>
<c>x</c>
<c>x</c>
......@@ -456,19 +457,17 @@
<section title='Congestion Control'>
<t>The adaptive nature of the Opus codec allows for an efficient
congestion control.</t>
<t>The target bitrate of Opus can be adjusted at any point in time and
thus allowing for an efficient congestion control. Furthermore, the amount
<t>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 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.</t>
<t>It is RECOMMENDED that congestion control is applied during the
transmission of Opus encoded data.</t>
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.</t>
<t>It is RECOMMENDED that senders of Opus encoded data apply congestion
control.</t>
</section>
<section title='IANA Considerations'>
......@@ -485,10 +484,11 @@
<t>Required parameters:</t>
<t><list style="hanging">
<t hangText="rate:"> RTP timestamp clock rate is incremented with
<t hangText="rate:"> the RTP timestamp is incremented with a
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 <xref target="fs-upsample-factors"/>.
rates. For data encoded with sampling rates other than 48000 Hz,
the sampling rate has to be adjusted to 48000 Hz using the
corresponding multiplier in <xref target="fs-upsample-factors"/>.
</t>
</list></t>
......@@ -525,30 +525,30 @@
<vspace blankLines='1'/>
</t>
<t hangText="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
<xref target="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
<t hangText="maxptime:"> the maximum duration of media represented
by a packet (according to Section&nbsp;6 of
<xref target="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 <xref target='opus-rtp-payload-format'/>. If no value is
specified, 120 is assumed as default. This value is a recommendation
specified, the default is 120. This value is a recommendation
by the decoding side to ensure the best
performance for the decoder. The decoder MUST be
capable of accepting any allowed packet sizes to
ensure maximum compatibility.
<vspace blankLines='1'/></t>
<t hangText="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 <xref target="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 <xref
<t hangText="ptime:"> the preferred duration of media represented
by a packet (according to Section&nbsp;6 of
<xref target="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 <xref
target='opus-rtp-payload-format'/>. If no value is
specified, 20 is assumed as default. If ptime is greater than
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
......@@ -557,15 +557,15 @@
ensure maximum compatibility.
<vspace blankLines='1'/></t>
<t hangText="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 <xref
target="RFC4566"/>. Possible values are 3, 5, 10, 20, 40, and 60
<t hangText="minptime:"> the minimum duration of media represented
by a packet (according to Section&nbsp;6 of
<xref target="RFC4566"/>) that SHOULD be encapsulated in a received
packet, in milliseconds rounded up to the next full integer value.
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 <xref target='opus-rtp-payload-format'/>. If no value is
specified, 3 is assumed as default. This value is a recommendation
specified, the default is 3. 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
......@@ -573,65 +573,66 @@
<vspace blankLines='1'/></t>
<t hangText="maxaveragebitrate:"> specifies the maximum average
receive bitrate of a session in bits per second (b/s). The actual
value of the bitrate may vary as it is dependent on the
receive bitrate of a session in bits per second (b/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 between
6000 and 510000 SHOULD be ignored. If no value is specified, the
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 <xref target='bitrate_by_bandwidth'/>
for the corresponding mode of Opus and corresponding maxplaybackrate:
will be the default.<vspace blankLines='1'/></t>
for the corresponding mode of Opus and corresponding maxplaybackrate
is the default.<vspace blankLines='1'/></t>
<t hangText="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
Possible values are 1 and 0 where 1 specifies that stereo signals are preferred,
and 0 specifies that only mono signals are preferred.
Independent of the stereo parameter every receiver MUST be able to receive and
decode stereo signals but sending stereo signals to a receiver that signaled a
preference for mono signals may result in higher than necessary network
utilisation and encoding complexity. If no value is specified, mono
is assumed (stereo=0).<vspace blankLines='1'/>
utilization and encoding complexity. If no value is specified,
the default is 0 (mono).<vspace blankLines='1'/>
</t>
<t hangText="sprop-stereo:">
specifies whether the sender is likely to produce stereo audio.
Possible values are 1 and 0 where 1 specifies that stereo signals are likely to
be sent, and 0 speficies that the sender will likely only send mono.
This is not a guarantee that the sender will never send stereo audio
(e.g. it could send a pre-recorded prompt that uses stereo), but it
indicates to the receiver that the received signal can be safely downmixed to mono.
This parameter is useful to avoid wasting receiver resources by operating the audio
processing pipeline (e.g. echo cancellation) in stereo when not necessary.
If no value is specified, mono
is assumed (sprop-stereo=0).<vspace blankLines='1'/>
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 pre-recorded prompt that uses stereo), but it
indicates to the receiver that the received signal can be safely downmixed to mono.
This parameter is useful to avoid wasting receiver resources by operating the audio
processing pipeline (e.g. echo cancellation) in stereo when not necessary.
If no value is specified, the default is 0
(mono).<vspace blankLines='1'/>
</t>
<t hangText="cbr:">
specifies if the decoder prefers the use of a constant bitrate versus
variable bitrate. Possible values are 1 and 0 where 1 specifies constant
bitrate and 0 specifies variable bitrate. If no value is specified, cbr
is assumed to be 0. Note that the maximum average bitrate may still be
changed, e.g. to adapt to changing network conditions.<vspace blankLines='1'/>
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.<vspace blankLines='1'/>
</t>
<t hangText="useinbandfec:"> specifies that the decoder has the capability to
take advantage of the Opus in-band FEC. Possible values are 1 and 0. It is RECOMMENDED to provide
0 in case FEC cannot be utilized on the receiving side. If no
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.
<vspace blankLines='1'/></t>
<t hangText="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.<vspace blankLines='1'/></t>
DTX. Possible values are 1 and 0. If no value is specified, the
default is 0.<vspace blankLines='1'/></t>
</list></t>
<t>Encoding considerations:<vspace blankLines='1'/></t>
<t><list style="hanging">
<t>Opus media type is framed and consists of binary data according
to Section 4.8 in <xref target="RFC4288"/>.</t>
<t>The Opus media type is framed and consists of binary data according
to Section&nbsp;4.8 in <xref target="RFC4288"/>.</t>
</list></t>
<t>Security considerations: </t>
......@@ -645,7 +646,7 @@
<t>Applications that use this media type: </t>
<t><list style="hanging">
<t>Any application that requires the transport of
speech or audio data may use this media type. Some examples are,
speech or audio data can use this media type. Some examples are,
but not limited to, audio and video conferencing, Voice over IP,
media streaming.</t>
</list></t>
......@@ -689,7 +690,7 @@
<t>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.</t>
channels MUST be 2.</t>
<t>The OPTIONAL media type parameters "ptime" and "maxptime" are
mapped to "a=ptime" and "a=maxptime" attributes, respectively, in the
......@@ -775,8 +776,8 @@
<t>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
restricted by this payload format description. The decoder MUST be
capable of decoding every received clock rate. An example
is shown below:
<figure>
......@@ -791,8 +792,8 @@
<t>The "ptime" and "maxptime" parameters are unidirectional
receive-only parameters and typically will not compromise
interoperability; however, dependent on the set values of the
parameters the performance of the application may suffer. <xref
interoperability; however, some values might cause application
performance to suffer. <xref
target="RFC3264"/> defines the SDP offer-answer handling of the
"ptime" parameter. The "maxptime" parameter MUST be handled in the
same way.</t>
......@@ -800,9 +801,8 @@
<t>
The "minptime" parameter is a unidirectional
receive-only parameters and typically will not compromise
interoperability; however, dependent on the set values of the
parameter the performance of the application may suffer and should be
set with care.
interoperability; however, some values might cause application
performance to suffer and ought to be used with care.
</t>
<t>
......@@ -811,9 +811,9 @@
of the other side SHOULD NOT send with an audio bandwidth higher than
"maxplaybackrate" as this would lead to inefficient use of network resources.
The "maxplaybackrate" parameter does not
affect interoperability. Also, this parameter SHOULD NOT be used
to adjust the audio bandwidth as a function of the bitrates, as this
is the responsibility of the Opus encoder implementation.
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.
</t>
<t>The "maxaveragebitrate" parameter is a unidirectional receive-only
......@@ -821,9 +821,9 @@
of the other side MUST NOT send with an average bitrate higher than
"maxaveragebitrate" as it might overload the network and/or
receiver. The "maxaveragebitrate" parameter typically will not
compromise interoperability; however, dependent on the set value of
the parameter the performance of the application may suffer and should
be set with care.</t>
compromise interoperability; however, some values might cause
application performance to suffer, and ought to be set with
care.</t>
<t>The "sprop-maxcapturerate" and "sprop-stereo" parameters are
unidirectional sender-only parameters that reflect limitations of
......@@ -866,22 +866,22 @@
<t><list style="symbols">
<t>The values for "maxptime", "ptime", "minptime", "maxplaybackrate", and
"maxaveragebitrate" should be selected carefully to ensure that a
"maxaveragebitrate" ought to be selected carefully to ensure that a
reasonable performance can be achieved for the participants of a session.</t>
<t>
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
capable of accepting any allowed packet sizes to
ensure maximum compatibility.
</t>
<t>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
the session. More than one configuration can be provided if necessary
by declaring multiple RTP payload types; however, the number of types
should be kept small.</t>
ought to be kept small.</t>
</list></t>
</section>
</section>
......@@ -891,11 +891,11 @@
<t>All RTP packets using the payload format defined in this specification
are subject to the general security considerations discussed in the RTP
specification <xref target="RFC3550"/> and any profile from
e.g. <xref target="RFC3711"/> or <xref target="RFC3551"/>.</t>
specification <xref target="RFC3550"/> and any profile from,
e.g., <xref target="RFC3711"/> or <xref target="RFC3551"/>.</t>
<t>This payload format transports Opus encoded speech or audio data,
hence, security issues include confidentiality, integrity protection, and
<t>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 <xref target="RFC3711"/>, MAY be used.</t>
......
0% or .
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment