[SR-Users] RTPPROXY issue and sip to sip calling

SamyGo govoiper at gmail.com
Sun Jan 31 20:39:19 CET 2016

Hi Rehan,

No matter which mode you are running rtpproxy in that IP will always be the
IP of the machine it is running on.
That means that SDP will take that IP once routed to locally subnet A2B
As far as the A2B detecting SIP user as online or offline based on DB,  I
am not too sure about it. If it is realtime then I think it should work out
of box. You may need to try it out to know it accurately.

On Jan 30, 2016 02:05, "Ahmed Rehan" <ahmed.rehan at gmail.com> wrote:

> Dear All
> I m trying to setup kamailio and asterisk in load balancing with a2billing
> . Currently all of my VMs, one Kamailio and two asterisks are on same
> subnet . I have started the RTPproxy like below
> ./rtpproxy -s udp: -l X.X.X.153 -m 10000 -M 50000 -u root
> root -F -d INFO LOG_LOCAL0
> My question is if all the VMs are on same subnet with same gateway what
> should be written in the private IP X.X.X.153/<private - ip>
> Secondly i m authenticating and registering the SIP on kamailio using the
> A2B DB . all the dialplan for a2b is being run on asterisk . Now if i want
> to call SIP peer to Peer like in case of followme case ,
> How should i route the calls in Kamailio ? will it be using usr loc
> module? if so any help will be appreciated
> --
> Regards
> Ahmed Rehan
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
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