UCP Phone/WebRTC call connecting but sometimes no audio

FreePBX Applications / Modules
1

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

2

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)

3

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 ~]#

4

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.

5

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