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

Wilkins, Steve swwilkins at mitre.org
Sun May 13 16:59:34 CEST 2018

Hi Alex,

On a WebRTC to WebRTC call (both UAC's registered in Kamailio) =>

- When Asterisk sends the INVITE back to Kamailio, it has the id of the call and the IP Address of the laptop the call was made from 
INVITE sip:vn768thu at 3stum5dgqf9j.invalid;transport=ws;alias=;alias=;ob SIP/2.0

- I have been using IP Addresses instead of Domains, as I have had much more luck doing it that way.

Should I switch to FQDN and Domains?  If so, may I ask what all would need to be changed?

Thank you for your time. 


-----Original Message-----
From: sr-users [mailto:sr-users-bounces at lists.kamailio.org] On Behalf Of Alex Balashov
Sent: Sunday, May 13, 2018 10:32 AM
To: Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
Subject: Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

That seems correct and sounds like it should work. What is the request URI of the INVITE that comes back into Kamailio on the B leg? Does its domain match the AOR domain? Do you have your registrar/usrloc set to use domains on lookup()?

-- Alex

Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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