How to enable RTP timeout and SIP keep alive in asterisk

Dear All ,
I hope everyone is doing fine. I just want to know something two specific parameters…Where can i enable RTP timeout and SIP keep alive in asterisk server. Do i have to configure this in sip.conf or somewhere else?

My 2nd question is how can i see whether those parameters are enabled or not from wireshark. Will be glad to have some kind of support.

Best Regards

Nowadays you should be using PJSIP, so you shouldn’t be using sip.conf, however the way to check is to see what is present in the sample configuration files. Parameters for both do appear in sip.conf.sample.

You cannot tell whether RTP timeout is being used from wireshark; you have to create an interruption in the RTP and see whether the call is dropped.

I think SIP keep alive is new, but if it similar to other implementations, you will see packets containing just CRLF on the signalling channel.

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