UCP Phone/WebRTC call connecting but sometimes no audio

I have setup UCP webrtc phone for our users. i managed to get it working.
However sometimes it has issues:
Ringing but not appearing on the other end
Ringing and once answered, no audio
When i did fwconsole restart it starts working again

Not sure if it is related but currently i use Microsip as our softphones, and it is working fine. I am setting up WebRTC so we can have more compatibilty in different OS.

Sorry newbie here please help

sounds like ports/firewall. You should run a Wireshark to see what is happening to the signaling and RTP on failed calls.

You can also enable SIP debugging and collect a trace of a failed call to help troubleshoot.

Providing Great Debug - Support Services - Documentation (freepbx.org)

hi comtech,

here are some details
[@pbx ~]# grep 1693402029.3969 /var/log/asterisk/full*
/var/log/asterisk/full-20230831:[2023-08-30 21:27:09] VERBOSE[2343][C-000007c9] pbx.c: Executing [s@macro-user-callerid:1] Set(“PJSIP/99206-00000799”, “TOUCH_MO NITOR=1693402029.3969”) in new stack
/var/log/asterisk/full-20230831:[2023-08-30 21:27:09] VERBOSE[2343][C-000007c9] pbx.c: Executing [recordcheck@sub-record-check:19] Set(“PJSIP/99206-00000799”, " __CALLFILENAME=out-+639089515811-206-20230830-212709-1693402029.3969") in new st ack
/var/log/asterisk/full-20230831:[2023-08-30 21:27:09] VERBOSE[2343][C-000007c9] pbx.c: Executing [recordcheck@sub-record-check:20] MixMonitor(“PJSIP/99206-00000 799”, “2023/08/30/out-+639089515811-206-20230830-212709-1693402029.3969.wav,abi( ),”) in new stack
/var/log/asterisk/full-20230831:[2023-08-30 21:27:09] VERBOSE[2343][C-000007c9] pbx.c: Executing [recordcheck@sub-record-check:24] Set(“PJSIP/99206-00000799”, " CDR(recordingfile)=out-+639089515811-206-20230830-212709-1693402029.3969.wav") i n new stack
/var/log/asterisk/full-20230831:[2023-08-30 21:27:31] VERBOSE[2343][C-000007c9] pbx.c: Executing [s@crm-hangup:4] NoOp(“PJSIP/99206-00000799”, “MASTER CHANNEL: 1693402029.3969 = 1693402029.3969”) in new stack
[@pbx ~]#

This is not what I recommended to you.

  1. Follow the instruction to turn sip debug on and capture a full log of a failed call, placing it in pastebin.
  2. Capture a Wireshark of the failed call.

For anyone to help you need to provide details on the failure.

This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.