Sip.conf

This is main Asterisk configuration file for SIP protocol.

It resides in /etc/asterisk/sip.conf

Registry (Providers with registrations)

• registry_diff_reload - modifies how Asterisk handles Registry reloads. When set to 0 (the default), Asterisk will utilize the standard reloading method, which entails deleting all Registry entries, rebuilding them from scratch, and sending SIP Register requests to all Registry entries (Providers). Setting the option to 1 activates a "diff reload" mode, which only reloads entries that have changed since the previous reload. This approach is advantageous for systems with a large number of Registry entries (Providers with registrations) as reloading all entries simultaneously can be time-consuming. Allowed values are either 0 (the default) - disabled, or 1 - enabled. This is a custom MOR setting and only works with the latest Asterisk 15 (starting with GIT revision af5d137).

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)