One way audio on webrtc until I press hold and resume

FreePBX Endpoints
1

Hi guys,

I’m trying to configure FreePBX with an WebRTC client, I tried a couple of prebuilt jsSIP phones. I’m using Twilio Elastic SIP trunk for testing purposes. I’m able to dial in and out, but the calls have one-way audio - there is silience from Twilio to me. This is supposed to be over the internet connectivity - so both my public and FreePBX public IP should be involved. Direct Media is turned off both for the SIP trunk and for the extension. When I run tcpdump on the server - the audio is there, but it gets lost on the way to my PC. The issue magically gets resolved if I press hold and resume - but for some time.

I’m able to get good calls using the CP phone, so the extension and trunk configuration are probably good. Any ideas - I’m fighting this for a couple of days and this is my first time playing around with FreePBX/Asterisk.

2

That symptom has been reported several times before, but I think it was for standard SIP to standard SIP. It was sufficiently long ago that it may have been fixed, but you haven’t said which version of Asterisk and even whether it is chan_sip or chan_pjsip (there is unlikely to have been an official fix in the former, assuming it occurs there).

You should probably search the forum to find the posts about the symptom and see if there is any solution given.

3

Hello David,

Thanks, your input actually helped - turns out it was some kind of a bug and upgrading from Asterisk 21 to 22 fixed the issue.