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T4xs series opus codec - Printable Version

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T4xs series opus codec - volga629 - 03-20-2023 09:47 PM

Hello Everyone,
I been testing Yealink opus and ran into issue where are SDP in INVITE is not follow RFC 7587
It offer opus/8000, but should be opus/48000 MHz always and actual stream can be adjusted as need it.

How possible to open bug report on this


Code:
User-Agent: Yealink SIP-T41S 66.84.0.15

v=0
o=- 20011 20011 IN IP4 172.32.0.250
s=SDP data
c=IN IP4 172.32.0.250
t=0 0
m=audio 11780 RTP/AVP 106 101
[b]a=rtpmap:106 opus/8000[/b]
a=fmtp:106 sprop-maxcapturerate=16000; maxaveragebitrate=20000; maxplaybackrate=48000; useinbandfec=1
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Explanation

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.

and

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.

and

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.

The actual rate used is encoded as part of the Opus bit stream. It can be hinted towards with the various fmtp= parameters, but even that is optional.