Hi,
I have a question about configuring the opus codec in Asterisk. I am currently on Asterisk version 20.7.0 but I plan to update to a newer version.
Background:
According to the RFC7587: RTP Payload Format for the Opus codec section 7.
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.
I have old Linphone clients that sends the SDP packets not following this standard.
a=rtpmap:107 opus/48000
It is missing the channel (2). It has been fixed in this Linphone commit but I don’t have any good way to update the version of the Linphone clients.
The incorrectly formatted rtpmap causes the Asterisk to report:
NOTICE[1087644]: res_pjsip_sdp_rtp.c:533 set_caps: No joint capabilities for 'audio' media stream between our configuration((opus)) and incoming SDP((nothing))
which is fair since it is incorrectly formatted.
Now to my question: Is there any way to make the Asterisk recognize and accept the incorrectly formatted messages? I have checked the configuration options for opus but cant find any obvious way to do so.
I don’t really expect it to be possible (and maybe it is a bad idea anyway), but if anyone knows a temporary workaround it would be greatly appreciated.
I should mention that our old Asterisk (version 13.21) is accepting and handling the incorrect messages but I really don’t want to go back to chan_sip and an unnecessarily old Asterisk version.