Hey Paul,
Thanks for your feedback!
In this case the call will come in via UDP (so it has 2 record route
headers). I can still play tricks like reading the record-route headers
but, I was in some way hoping that is the responsibility of
loose_route() function to properly use the right protocol out of route
headers since we are a proxy.
Cheers,
DanB
On 14.11.2014 12:00, sr-users-request(a)lists.sip-router.org wrote:
Message: 3
Date: Thu, 13 Nov 2014 16:55:18 +0530
From: Varghese Paul<varghesepaul87(a)gmail.com>
To: "Kamailio (SER) - Users Mailing List"
<sr-users(a)lists.sip-router.org>
Subject: Re: [SR-Users] UDP-TCP bridging in-dialog SUBSCRIBE
Message-ID:
<CAM33Ch4furD1FM-ZNKfbQ0nXNQd88BmzW73gjjb=cVyuC0K=8w(a)mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
Hi Dan,
You can check the transport value in the RURI and relay the SIP message as
tcp.
if ($*rP* == "tcp"){
t_relay_to_tcp()
}
Regards
Varghese Paul