VoIP Phones - Asterisk: Difference between revisions

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==PJSIP: NAT Issues: Keep-Alive / Anti-Idle==

If you are using a firewall or NAT router with short timeouts on UDP sessions ~~you~~ ~~can~~ ~~force~~ packets to be sent over the connection to keep it alive.

If you are using a firewall or NAT router with short timeouts on UDP sessions the example registration configuration above enables OPTIONS packets to be sent over the connection to keep it alive.

A symptom of NAT issues can be inbound call delivery working, then not working ~~for a bit~~ (with the code "183" in the CDR in the control panel) then calls working again without you doing anything (or possibly reloading or restarting asterisk will solve it for a bit).

A symptom of NAT issues can be inbound call delivery working for a bit after startup, then not working (with the code "183" in the CDR in the control panel) then calls working again a number of minutes later without you doing anything (or possibly reloading or restarting asterisk will solve it for a bit).

As Andrews & Arnold have multiple call servers it is possible ~~calls~~ ~~can~~ come from multiple IP addresses. Your NAT session or SIP helper/ALG may only be allowing SIP traffic from the ~~server~~ your Asterisk registered to.

As Andrews & Arnold have multiple call servers it is possible for call signalling to come from multiple IP addresses. Your NAT session or SIP helper/ALG may only be allowing SIP traffic from the IP address your Asterisk registered to.

In the PJSIP registration example ~~above~~ this is ~~mitigated~~ by having three AOR (Address of Record) sections. All three have qualify_frequency set (this makes asterisk send keepalive packets). The endpoint section then references the 3 aor entries.<br />

In the PJSIP registration example on this page the problem is solved by having three AOR (Address of Record) sections. All three have qualify_frequency set (this makes asterisk send keepalive packets). The endpoint section then references the 3 aor entries.<br />

Your Asterisk server will send outbound packets to the shared hostname, and the two call servers specifically. This has the effect of keeping the NAT session on your router alive for all the call servers meaning it doesn't matter which call server delivers the INVITE (call), it should always reach your Asterisk server.