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:maxim@123.123.123.123;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: mediagw.0@172.21.0.1: Call established: sip:maxim@123.123.123.123;transport=TCP Event: {'event': True, 'type': 'CALL_ESTABLISHED', 'class': 'call', 'accountaor': 'sip:mediagw.0@172.21.0.1', 'direction': 'incoming', 'peeruri': 'sip:maxim@123.123.123.123;transport=TCP', 'peerdisplayname': '7-test321', 'id': 'IL3XKiaw6vH4-Vmwr8SGpg..', 'remoteaudiodir': 'sendrecv', 'remotevideodir': 'inactive', 'audiodir': 'sendrecv', 'videodir': 'inactive', 'param': 'sip:maxim@123.123.123.123;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:mediagw.0@172.21.0.1', 'direction': 'incoming', 'peeruri': 'sip:maxim@123.123.123.123;transport=TCP', 'peerdisplayname': '7-test321', 'id': 'IL3XKiaw6vH4-Vmwr8SGpg..', 'remoteaudiodir': 'sendrecv', 'remotevideodir': 'inactive', 'audiodir': 'sendrecv', 'videodir': 'inactive', 'param': 'rtp stream error'} Baresip: sip:test321@123.123.123.123: Call with sip:maxim@123.123.123.123;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 :-)