[Freeswitch-dev] RTP session timer

[Thiya Ramalingam]

All - we are trying to debug an issue with a point to point call between two linphones going through FS (version 1.5.2). Occasionally, one of the parties in the call hears a brief drop in the audio - we traced the issue to rtp flushing in FS. we don't want to disable flushing as it causes latency issues. 

We looked at the wireshark traces to make sure that the flushing is not caused by talk spurt (marker bit being set by linphone which will cause the FS to flush). That's not the case as well.

One other possibility is the RTP session timer expiry. can someone point us to where the timer expiry is handled in the code ? we see where the timer is started but would like to know what happens when this timer expires ? 


2014-02-21 14:14:02.930515 [DEBUG] switch_rtp.c:2016 Starting timer [soft] 960 bytes per 20ms
2014-02-21 14:14:02.930515 [DEBUG] switch_rtp.c:2106 RTP started .


Thanks in advance,
Thiyag.