CODEWITHSUNDEEP
webrtcasteriskpjsip
Dmytro Golovchenko
Dmytro Golovchenko

Reputation: 56

Asterisk 13.7.2 pjsip 2.4.5 unexpected BYE with SIP cause 58 while answering

I'm upgrading Asterisk 11 to 13 and testing new configuration with WebRTC enpoints. For some reason PJSIP is dropping call sending BYE to replying endpoint after receiving OK packet from that end. No any errors in the console, only == Everyone is busy/congested at this time (1:0/0/1) populated between sending ACK and BYE

At the same time clients are able to place a call and listen messages or perform calls routed via iax

With same scenario Asterisk 11 just establishing connection:

SIP/920-00000009 answered SIP/929-00000008
Audio is at 10292
Adding codec 100003 (ulaw) to SDP

Could you please advise what could be a reason for such behaviour? P.S with the same config Asterisk 13 is sending/receiving SIP MESSAGE perfectly

Because this forum is not happy to populate SIP debug output as code, you can refer my post on the same at Asterisk community forum for SIP debug, pjsip.conf endpoint configuration and pjsip show contacts output

Upvotes: 1

Views: 704

Answers (1)

Dmytro Golovchenko
Dmytro Golovchenko

Reputation: 56

I found the reason, it was option "force_avp = yes" in an endpoint configuration. Could someone advise why it was a case, when media details as "m=audio 31118 UDP/TLS/RTP/SAVPF 0" are present in all INVITE and OK packets?

Another question, is there any reasonable way to debug such issues? I meant to get readable message in console/log with reason of channel unavailability? I guess it should be inside of pjsip library, but is there any way to get it?

P.S. Also I found, that without option "use_avpf = yes" (and without "force_avp = yes" as well) connection has been established, however no audio is going between endpoints.

Upvotes: 1

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