Problem with PJSIP endpoint and CHAN_SIP trunk

Hello all!

We are running FreePBX 14+Asterisk 13 and I’m having one way audio issues when using PJSIP on endpoints but no issues when i use CHAN_SIP in the same endpoint.

We have a SIP Trunk (from Vivo, a brazilian PSTN and SIP Trunk provider) phisically connected (eth1 with a IP address assigned) registered via CHAN_SIP.

CHAN_SIP is responding at the 5060 port (for old local IP phones) and PJSIP is in the other port (for remote workers with softphones).

These tests where made from the softphone in my LAN, so there’s no firewall between my softphone and my PBX:

When i register my softphone using CHAN_SIP, i can make calls for internal extensions and i can use the SIP trunk for outbound calls without any problem.

When i register my softphone using PJSIP, i can make calls for other extensions but when i make a outbound call (using the SIP Trunk), i can’t get two way audio. I can’t hear the other person.

It’s not a codec issue since i can use my PJSIP softphone with OPUS to call another CHAN_SIP extension using G.711. I can call *43 (echo test) and i hear myself too.

I have no idea of what is causing this issue since CHAN_SIP is working well.

Hey Fernando,

Codec negotiation is how the phones decide which codec they can both use to accomplish the call. In cisco land if there is a mismatch you get a media not acceptable error when placing the call. While I believe your softphone supports the OPUS codec if the other device does not then the call will not work.

If Chan SIP is working fine then in my book that normally rules out a networking issue.

I think your issue may lie in your PJSIP configuration or PJSIP trunk configuration. Chan SIP normally uses UDP 5160 and PJSIP 5060. Are you able to provide a packet capture of a test call?

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