SIP Timeouts

Asterisk Asterisk Support
1

I was experiencing call drops at the 2 minute and 30 second mark. Actually, the call is still on but just the audio drops. The firewall shows at this time mark that communication with SIP proxy goes from two way to one way.

After some troubleshooting, I changed the default TCP session timeout on my firewall (for SIP traffic) from the default 120 seconds to 600 seconds. After this, I was able to test with a 30 minute call and there was no audio drop.

My question is, during the UDP steam of the SIP call I assume there was originally no TCP traffic for at least 120 seconds causing the NAT session to close. But, the SIP device was registering with proxy every 30 seconds. Is this only UDP traffic?

So, how often does some TCP traffic traverse the firewall? What variables can change this?

Can someone please illustrate this process for me? Thanks!

2

Unless you explicitly request TCP, there is only UDP.

The options you want are the session timers ones. I’m not sure if they are used over TCP.

Most firewalls only need out of dialogue qualify traffic, but it sounds like yours does need session timers.

Both sides need to support session timers.