I am no Daniel, but here is a stab at it ...
The pseudo variable “oP” maybe something you can do a quick if/else on?
Taken from http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:1.5.x :
Transport protocol of SIP request original URI
$oP - reference to transport protocol of original R-URI
There is also dP and rP as well.
Hope this helps,
-graham
On 3/16/12 12:11 PM, "Andres Collazos" <anfecora@gmail.com> wrote:
thanks for all the support for all this years.
Can you please help me to know if there is any way to route sip calls based on transport protocol, for example a call incoming on tcp i will assign a route and if a call comes in udp i will assign a different route.
my scenario is calls coming from different devices registering to kamailio and then kamailio send those calls to asterisk.
unfortunately i have to create a different peer set for each device. for this scenario i have two types of UAs and they need completely different configuration on the asterisk switch.
i am planing on segregate the traffic and build a media server for each type of device means 2 asterisk, teh only way that i can identify those incoming registrations is by the use of the port one client connects trough udp the other trough tcp.
I appreciated any input in this matter.
and again thank a lot to all for the great support.
Andres Collazos.
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