We are testing testing VitalPBX 3.1.1-1 . We managed to get everything we needed up and running using chan_sip (local calls using Opus, call through our trunk using g.722), incomming and outgoing calls, queues etc…
We are equiped with Yealink T53 SIP phones and have a 10 Mbps symmetrical dedicated line for VOIP.
We have been working with pure asterisk and chan_sip for years.
Our Yealink phones are using PJSIP as SIP stack for their firmwares, and also PJSIP is the stack privileged now by the Asterisk project so we gave a try to PJSIP. But this is where we face issues when using Opus.
Something easy to reproduce is configuring a new extension and association a new device using a PJSIP profile (on which we have enabled G722 and Opus and set Opus as prefered codec, followed by G722). If we call the voicemail, we don’t hear. If I use a Linphone device (Opus enabled), I hear a very choppy announce. It’s a little bit better for G722 but not so much. However, if I set ulaw or alaw as prefered codec, everything works correctly.
This happens only when using the pjsip chan driver, the same tests with sip chan driver give normal results.