[SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

Wilkins, Steve swwilkins at mitre.org
Wed May 9 19:15:02 CEST 2018

Hello All,

I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using Kamailio and Asterisk.

Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 from WebRTC Client2 what I see is ->
The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the Invite back to Kamailio.  Also depending on
The version of Asterisk, the INVITE will then get forwarded to the AOR that is registered in Kamailio for the called number.
Does this seem correct?  It seems like there is an extra hop in there.

The reason I am now very curious now is because everything works fine if using Kamailio 5.0 and Asterisk 14.6, but I switch to Asterisk 15.3
I get the extra hop and call is dropped after 30 seconds.

I would appreciate any thoughts on this.

Thank you in advance.

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