VoIP Mailing List Archives

[ak at hejdu.dk]

We had some problems with "hanging channels" for our webrtc clients (via kamailio).To solve the problem I tried to use "media_timeout" setting but it didn't really work. So I tried the deprecated "rtp-timeout-sec" and this actually works fine?


Not working:
<param name="media_timeout" value="300"/>



Working:
<param name="rtp-timeout-sec" value="300"/>


How to reproduce: Make a call using webrtc and just close browser window, after some time freeswitch should close the channel because of missing rtp.

Instead it just keeps writing: "switch_rtp.c:853 No audio stun for a long time!" forever...


Anyway, it works now....just curious why...a typo or bug?


/Allan


Using:
FreeSWITCH Version 1.10.5-release+git~20200818T185121Z~25569c1631~64bit