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InnovateAsterisk / Browser-Phone

A fully featured browser based WebRTC SIP phone for Asterisk
https://www.innovateasterisk.com
GNU Affero General Public License v3.0
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video call issue drops #321

Open adrehmanforunity opened 1 year ago

adrehmanforunity commented 1 year ago

Dear ALL,

I have a perfectly running scenario with the audio call, but when making video calls it drops after some seconds and the server console says "[2022-09-16 16:12:10] NOTICE[19883]: res_pjsip_sdp_rtp.c:146 rtp_check_timeout: Disconnecting channel 'PJSIP/1083-00000004' for lack of video RTP activity in 30 seconds".

i am clueless.

Regards,

InnovateAsterisk commented 1 year ago

Does the video display on both sides before it ends?

adrehmanforunity commented 1 year ago

yes its 2 way...

prior to your reply, I didn't call from one extension to another. I was calling over *43 (each back test) but even calling from one extension to another result are the same. and voice-only call is always going well.

image

InnovateAsterisk commented 1 year ago

That Debug isn't going to show you what you need. You need to enable the RTP debug. > rtp set debug on but be aware that every packet gets a debug row, so you generate a lot. Look out for the correct destinations.

Typically if you set webrtc = yes on your endpoint, it nicely sets all the things, like encryption etc. The only thing I can think of here is that somehow video is not enabled, or you have left the video codec out, or its not its not 'bundled'.

Between, the RTP debug, and a local wireshark trace, you should be able to visually see the packet flow for the video.

InnovateAsterisk commented 1 year ago

Did you ever get to the bottom of this?

NOTICE[21656]: res_pjsip_sdp_rtp.c:145 rtp_check_timeout: Disconnecting channel 'PJSIP/yAS43lAg8L-00000001' for lack of video RTP activity in 60 seconds

I got this today, from Echo script. Video was flowing correctly.

lachesis commented 4 months ago

This is a very old post, but still the first result for searching that error message. If you set rtp_timeout=0 on the pjsip endpoint, you'll avoid this drop.