Asterisk 18 Opus problems with Freeswitch


I am using Asterisk 18.8.0 with Opus codec codec_opus-18.0_1.3.0-x86_64.tar.gz from Index of /pub/telephony/codec_opus

When calling via chan_pjsip to a Freeswitch (BigBlueButton) installation, audio via Opus works perfectly.
But sometimes a call is disconnected at some time in the middle of the call. The Freeswitch log says that the Opus PTIME was changed from 20ms to 1ms, which is invalid for Freeswitch. Therefore the call is disconnected.
I tried a lot of settings in codec.conf as described on Codec Opus - Asterisk Project - Asterisk Project Wiki
but without success. It seems Asterisk-Opus changes the session parameters during the connection, but I am not sure.
Is there are setting for Opus to not to change PTIME? Or am I looking at the wrong place for this issue?


Any opus settings would be configured in codecs.conf, there is none for this. As well the codec_opus you are using is a binary module distributed by Sangoma, any issues with it would need to be filed on the issue tracker[1]. There is no time frame on them.

[1] System Dashboard - Digium/Asterisk JIRA

If there is another Opus module for Asterisk available, I would like to test it.
Is there another module?

I can’t comment on any such things.

There you go …
However, your issue sounds like either an SDP or remote implementation bug. To check the SDP theory, a SIP trace around the time of that FreeSWITCH log entry would be one approach. Anyway, with the linked Opus-Codec module, you should be able to debug any Opus-Codec related cause because you can not only debug the module but Opus-Codec itself.

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