diff --git a/doc/draft-spittka-payload-rtp-opus.xml b/doc/draft-spittka-payload-rtp-opus.xml
index 05dd3de56fdb7b6e75095e006209f16873a607e6..494229590aae03e0364c0017f23d2295039e9a5c 100644
--- a/doc/draft-spittka-payload-rtp-opus.xml
+++ b/doc/draft-spittka-payload-rtp-opus.xml
@@ -176,15 +176,15 @@
</t>
<t>
- The voice mode allows to efficiently encode voice signals at lower bit
+ The voice mode allows efficient encoding of voice signals at lower bit
rates while the audio mode is optimized for audio signals at medium and
higher bitrates.
</t>
<t>
The Opus speech and audio codec is highly scalable in terms of audio
- bandwidth and bitrate and complexity. Further, Opus allows to
- transmit stereo signals.
+ bandwidth, bitrate, and complexity. Further, Opus allows
+ transmitting stereo signals.
</t>
<section title='Network Bandwidth'>
@@ -325,7 +325,8 @@
Opus allows for transmission of stereo audio signals. This operation
is signaled in-band in the Opus payload and no special arrangement
is required in the payload format. Any implementation of the Opus
- decoder MUST be capable of receiving stereo signals.
+ decoder MUST be capable of receiving stereo signals, although it MAY
+ decode those signals as mono.
</t>
<t>
If a decoder can not take advantage of the benefits of a stereo signal
@@ -350,8 +351,8 @@
therefore no padding is required. 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 has no function in combination
- with Opus and MAY be ignored.</t>
+ <t>The marker bit (M) of the RTP header is used in accordance with
+ 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>
@@ -521,6 +522,18 @@
<vspace blankLines='1'/>
</t>
+ <t hangText="sprop-maxcodedaudiobandwidth:">
+ a hint about the maximum audio bandwidth that the sender is likely to produce.
+ This is not a guarantee that the sender will never send any higher bandwidth
+ (e.g. it could send a pre-recorded prompt that uses a higher bandwidth), but it
+ indicates to the receiver that frequencies above this maximum can safely be discarded.
+ This parameter is useful to avoid wasting receiver resources by operating the audio
+ processing pipeline (e.g. AEC) at a higher rate than necessary.
+ Possible values are nb, mb, wb, swb, fb. By default, the sender
+ is assumed to have no limitations, i.e. fb.
+ <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
@@ -590,6 +603,19 @@
is assumed (stereo=0).<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. AEC) in stereo when not necessary.
+ If no value is specified, mono
+ is assumed (stereo=0).<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
@@ -669,8 +695,8 @@
<t>The media type ("audio") goes in SDP "m=" as the media name.</t>
<t>The media subtype ("opus") goes in SDP "a=rtpmap" as the encoding
- name. The RTP clock rate in "a=rtpmap" MUST be mapped to the required
- media type parameter "rate".</t>
+ name. The RTP clock rate in "a=rtpmap" MUST be 48000 and the number of
+ 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
@@ -690,7 +716,7 @@
<artwork>
<![CDATA[
m=audio 54312 RTP/AVP 101
- a=rtpmap:101 opus/48000
+ a=rtpmap:101 opus/48000/2
]]>
</artwork>
</figure>
@@ -704,7 +730,7 @@
<artwork>
<![CDATA[
m=audio 54312 RTP/AVP 101
- a=rtpmap:101 opus/48000
+ a=rtpmap:101 opus/48000/2
a=fmtp:101 maxcodedaudiobandwidth=wb; maxaveragebitrate=20000;
stereo=1; useinbandfec=1; usedtx=0
a=ptime:40
@@ -731,8 +757,8 @@
<figure>
<artwork>
<![CDATA[
- m=audio 54312 RTP/AVP 100
- a=rtpmap:100 opus/48000
+ m=audio 54312 RTP/AVP 100
+ a=rtpmap:100 opus/48000/2
]]>
</artwork>
</figure>
@@ -777,19 +803,19 @@
in <xref target='bitrate_by_bandwidth'/> the session MUST be rejected.</t>
<t>
- The parameter "stereo" is a unidirectional receive-only
+ The "stereo" parameter is a unidirectional receive-only
parameter.
</t>
<t>
- The parameter "cbr" is a unidirectional receive-only
+ The "cbr" parameter is a unidirectional receive-only
parameter.
</t>
- <t>The parameter "useinbandfec" is a unidirectional receive-only
+ <t>The "useinbandfec" parameter is a unidirectional receive-only
parameter.</t>
- <t>The parameter "usedtx" is a unidirectional receive-only
+ <t>The "usedtx" parameter is a unidirectional receive-only
parameter.</t>
<t>Any unknown parameter in an offer MUST be ignored by the receiver