PJSIP Trunk one-way audio problem

Hi all,

Not what you think this is - usually, one way audio doesn’t defeat me but this looks a little non standard. The advantage is that I am also the SIP Provider, and PBX Maintainer. I have an Asterisk 13/FPBX 13 install up, and I’ve brought up a SIP trunk to our Provider FreeSwitch servers. For outbound calling, everything works well, and audio flows in both directions.

For an inbound call, this is where it gets into never-before-seen territory. From a pcap on the Provider servers everything looks excellent, they INVITE, Asterisk replies back with 100 Trying, then 183 Session Progress (with media SDP). After the 183, media negotiates with a-law and I can see two RTP flows (indicating, two way audio), then we see Asterisk sending 180 Ringing (twice) and I assume that with Early-Offer the media comes up to carry the “ringing audio”. All normal so far… After the two 180 Ringings, Asterisk sends a 200 OK (I guess this is when I pick up the call), and the provider sends an ACK. That 200 OK also has media SDP in it. After the Provider sends the ACK, Asterisk brings up a G722 RTP on the same source port, but the Provider doesn’t seem to bring up it’s RTP back to me (assuming the first a-law flow was torn down).

So, I know it’s normal to be able to change RTP streams and so forth during calls - is this a RE-INVITE? I do have canreinvite set to no in Asterisk, so it’s not bringing up a new RTP directly between the endpoint (or, it shouldn’t be).

Well, I suck at explaining so here’s a link to a stripped back overview of the pcap. As the arrows on it dictate … Right hand side is the Provider server, left hand side is the PBX. Trunk is PJSIP. Provider is FreeSwitch. Codec preference is a-law, u-law, g722. Both sides, and endpoint support these codecs.



Sometimes I question why I post things to forums… just makes me look like a fool!

So, I went back to Asterisk and figured - Hey, anything other than a-law and u-law on the trunk is useless, when it goes out to the PSTN it’s a-law anyway (yeah, I live down under). So, pull off g722, as well as GSM and g726 - not needed, no point, the PBX sits in the same DC as the provider servers, bandwidth is not an issue.

Ahhh, guess what? It stopped trying to bring up the g722 stream after the 200 OK!

Solved enough for me …


Hi I’m still new to this Freepbx my self i had a one way audio problem and someone in here told me i didn’t setup my External Address Detection which is in the Asterisk Sip settings. I clicked on the Detect External IP button and it saw my Address information i click applied and confirm and somehow my audio was working. Maybe this also can help you? I’m not sure but through i message. Have a good day.