Sip.conf

This is main Asterisk configuration file for SIP protocol.

It resides in /etc/asterisk/sip.conf

RTP timers

These timers are currently used for both audio and video streams. The RTP timeouts are only applied to the audio channel. The settings are settable in the global section as well as per device

rtptimeout = 60

Terminate call if 60 seconds of no RTP or RTCP activity on the audio channel when we're not on hold. This is to be able to hangup a call in the case of a phone disappearing from the net like a powerloss or grandma tripping over a cable.

rtpholdtimeout = 300

Terminate call if 300 seconds of no RTP or RTCP activity on the audio channel when we're on hold (must be > rtptimeout).

rtpkeepalive=<secs>

Send keepalives in the RTP stream to keep NAT open (default is off - zero)