test with ‘guestuser’ works fine but when i create user with freepbx with all the same parameters from guestuser but with name 602 (phone number) - in 30seconds after starting video-conference i see error: res_pjsip_sdp_rtp.c:146 rtp_check_timeout: Disconnecting channel ‘PJSIP/602-000000d8’ for lack of video RTP activity in 30 seconds.
can video-conference in asterisk work without webrtc? if so - can you point me to client softphone which support it (preferably non-paid)?
i’ve tried 3cx, microsip, linphone, phonerlite, blink and others which i don’t remember already
Asterisk doesn’t touch video, decode it, make a single “video conference” video stream, none of that. Any support for “video conferencing” would have to be done outside of Asterisk, such as by something establishing multiple SIP calls itself.
i was not asking for single conference video-stream. i need a softphone sip-client which can accept multiple video-stream from asterisk like “cyber mega phone 2000” from guide that i mentioned in first post here. and second question was: can it be done without webrtc?
I have seen a Desktop Softphone (UDP-RTP) work with a single video stream - but failed when put into a SFU conference - or just showed a single video stream in any case. It was a long time ago - i think I used Zoiper or Linphone.
Mostly the inspiration for Siperb, among other things, was to accommodate for this kind of thing. UI wise, it can be tricky to accommodate N-many video streams. Not many of them are capable of doing it.