Sip plugin no audio in client

[Ahmad Jaber]

I'm using the sip plugin demo, hosted locally, and asterisk 18.

If a call is made where the both sip peers are janus, through asterisk, the call works perfectly, but if one of the two peers is not janus audio can't be heard on the client.

I tried playing hello world back:
exten = 100,1,Answer()
same = n,Wait(1)
same = n,Playback(hello-world)
same = n,Hangup()
but still no audio in the client.

I did pcap dump on the janus server and the rtp is reaching janus server, and I listened to it in wireshark.

I can't see any errors with --debug-level=7 in janus, and the logs in the client is the same in both cases.

Does anyone know what may be the cause of this issue?

[Ahmad Jaber]

I'm not sure if this is causing anything, but this appears in janus's logs in both cases.
No medium with a valid peer SSRC found for transport-wide CC feedback

[Lorenzo Miniero]

Check the Admin API to see if there's something wrong on the WebRTC side:

https://www.meetecho.com/blog/understanding-the-janus-admin-api/

Of course you may want to also check chrome://webrtc-internals (on Chrome) or about:webrtc (on Firefox) to see if there's anything that can explain the missing audio there (e.g., WebRTC errors, no incoming data, etc.).

L.

[Ahmad Jaber]

Thank you, for your help.