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Hey Paul,<br>
<br>
Thanks for your feedback! <br>
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.<br>
<br>
Cheers,<br>
DanB<br>
<br>
<div class="moz-cite-prefix">On 14.11.2014 12:00,
<a class="moz-txt-link-abbreviated" href="mailto:sr-users-request@lists.sip-router.org">sr-users-request@lists.sip-router.org</a> wrote:<br>
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cite="mid:mailman.3.1415962801.1169.sr-users@lists.sip-router.org"
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<pre wrap="">Message: 3
Date: Thu, 13 Nov 2014 16:55:18 +0530
From: Varghese Paul <a moz-do-not-send="true" class="moz-txt-link-rfc2396E" href="mailto:varghesepaul87@gmail.com"><varghesepaul87@gmail.com></a>
To: "Kamailio (SER) - Users Mailing List"
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Subject: Re: [SR-Users] UDP-TCP bridging in-dialog SUBSCRIBE
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Hi Dan,
You can check the transport value in the RURI and relay the SIP message as
tcp.
if ($<b class="moz-txt-star"><span class="moz-txt-tag">*</span>rP<span class="moz-txt-tag">*</span></b> == "tcp"){
t_relay_to_tcp()
}
Regards
Varghese Paul</pre>
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