Hi All, I have my two extensions configured with rtp_timeout=60, and I connect them both to a conf_bridge conference, audio, video and messaging all works fine. After 60 seconds they get disconnected.
NOTICE: res_pjsip_sdp_rtp.c:145 rtp_check_timeout: Disconnecting channel 'PJSIP/drdZc30SSe-00000003' for lack of video RTP activity in 60 seconds
If its an audio-only conference call, they can go on for ever, or if one is audio and the other is video the video participant will get disconnected after 60 seconds.
If I call
StreamEcho() It also ends after 60 seconds a of perfectly working call.
Are you actually receiving video from the channel referenced by the mesage?
Ah, I see yes, and note your comment on this. I’m trying to work out how this become an issue for me, because it didn’t do this before, and I’m pretty sure have kept to the same Asterisk version and config.
It could be that chrome has changed their default bundle behaviour - I was under the impression that the current default for bundle from Chrome is “balanced” (instructs the browser to pick two tracks to send — one audio and one video).
What does Asterisk do when bundle=yes?
With this option enabled, Asterisk will attempt to negotiate the use of bundle. If negotiated this will result in multiple RTP streams being carried over the same underlying transport…”
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