No sound and video from/to SIP client

[callmemaxim]

Hello!

Thank you so much for your useful project, but there are some difficulties during deployment process.
I am using renater/sipmediagw:1.4.5 (with 1.1.2 tag from SIPMediaDeploy repo - the same issue)
And actual docker-compose.yml file.

So during the call from SIP client (tested with zoiper/linphone) I have no incoming/outcoming sound and video from and to SIP client. The client is connecting to conference and I can see him in the list of members of current conference but I can't hear and see him and the same from the client's side.

There are several errors at startup in gateway's log:
xrandr: Failed to get size of gamma for output screen
xrandr: Failed to get size of gamma for output screen
Baresip: pulse: pa_context_connect failed: (Connection refused)
Baresip: pulse: pa_context_connect failed: (Connection refused)
Baresip: module pulse.so: Operation not permitted [1]

So when I am connecting from SIP client I see this:
Baresip: call: media-nat 'turn' established/gathered
Baresip: call: answering call on line 1 from sip:[email protected];transport=TCP with 200
Baresip: stream: update 'audio'
Baresip: stream: update 'video'
Baresip: stream: update 'video'
Baresip: audio: Set audio decoder: PCMA 8000Hz 1ch
Baresip: audio: player started with sample format S16LE
Baresip: audio: Set audio encoder: PCMA 8000Hz 1ch
Baresip: audio: source started with sample format S16LE
Baresip: audio tx pipeline: alsa ---> aubuf ---> PCMA
Baresip: audio rx pipeline: alsa <--- aubuf <--- PCMA
Baresip: bfcp channel is disabled
Baresip: stream: Enable RTP timeout (60000 milliseconds)
Baresip: [email protected]: Call established: sip:[email protected];transport=TCP
Event: {'event': True, 'type': 'CALL_ESTABLISHED', 'class': 'call', 'accountaor': 'sip:[email protected]', 'direction': 'incoming', 'peeruri': 'sip:[email protected];transport=TCP', 'peerdisplayname': '7-test321', 'id': 'IL3XKiaw6vH4-Vmwr8SGpg..', 'remoteaudiodir': 'sendrecv', 'remotevideodir': 'inactive', 'audiodir': 'sendrecv', 'videodir': 'inactive', 'param': 'sip:[email protected];transport=TCP'}
Event: MOTD: Element selection: Message: : Blocked a frame with origin "file://" from accessing a cross-origin frame.
Event: (Session info: chrome=110.0.5481.77)
Event: Stacktrace:
Event: #0 0x560f4a4bbd93
Event: #1 0x560f4a28a2d7
Event: #2 0x560f4a28d8d3
Event: #3 0x560f4a28d642
Event: #4 0x560f4a28e2bc
Event: #5 0x560f4a30315e
Event: #6 0x560f4a2ea5f2
Event: #7 0x560f4a302619
Event: #8 0x560f4a2ea353
Event: #9 0x560f4a2b9e40
Event: #10 0x560f4a2bb038
Event: #11 0x560f4a50f8be
Event: #12 0x560f4a5138f0
Event: #13 0x560f4a4f3f90
Event: #14 0x560f4a514b7d
Event: #15 0x560f4a4e5578
Event: #16 0x560f4a539348
Event: #17 0x560f4a5394d6
Event: #18 0x560f4a553341
Event: #19 0x7f2b6ee41ea7 start_thread
Baresip: Baresip: [0:00:42] audio=63573/0 video=0/0 video=0/0 (bit/s) efps=0.0/0.0 efps=0.0/0.0
Baresip: Baresip: [0:00:42] audio=63573/0 video=0/0 video=0/0 (bit/s) efps=0.0/0.0 efps=0.0/0.0

And then the call ends (and jitsi kicks the participant) and I can see this in log:
Event: {'event': True, 'type': 'CALL_CLOSED', 'class': 'call', 'accountaor': 'sip:[email protected]', 'direction': 'incoming', 'peeruri': 'sip:[email protected];transport=TCP', 'peerdisplayname': '7-test321', 'id': 'IL3XKiaw6vH4-Vmwr8SGpg..', 'remoteaudiodir': 'sendrecv', 'remotevideodir': 'inactive', 'audiodir': 'sendrecv', 'videodir': 'inactive', 'param': 'rtp stream error'}
Baresip: sip:[email protected]: Call with sip:[email protected];transport=TCP terminated (duration: 1 min )
Baresip: audio Transmit: Receive:
Baresip: packets: 2997 0
Baresip: avg. bitrate: 63.7 0.0 (kbit/s)
Baresip: errors: 0 0
Baresip: ua: stop all (forced=0)
Baresip: Baresip: [0:01:00] audio=63573/0 video=0/0 video=0/0 (bit/s) efps=0.0/0.0 efps=0.0/0.0

I suppose, that it's another issue and its related with UI update (there are no such lines in the log of the SIPMediaDeploy repo's gw):
Event: MOTD: Element selection: Message: : Blocked a frame with origin "file://" from accessing a cross-origin frame.
Event: (Session info: chrome=110.0.5481.77)

[nicotyze]

Hello, Thanks for your feedback !
I've fixed the selenium browsing errors (+ fixes related to test->deploy folder renaming) and updated sipmediagw:1.4.5 image accordingly.
I also made a call test with linphone after having deployed, locally, the entire environment (sipmediagw+kamailio+coturn) thanks to the vagrant file. The call flow seems to be ok:

[nicolas-semaphor]

Bump.

I'm having issues too with the 1.4.7 Docker image. I can dial a gateway instance and the headless browser will join the meeting on our Jitsi Meet server, but no audio and video is transmitted between softphone, gateway and Jitsi.

Output from "docker logs --follow gw0"

I should mention that it works with the vagrant setup, I just wish to run it dockerized on the host machine, and not in a VirtualBox machine :-)