[SR-Users] Caller Not Receiving RTP Feed

[Benjamin Trent]

Hugh,

Thank you for the response, however, now neither the callee or the caller
are getting a feed. I am not sure if the nat requires symmetric RTP. If
there is a way to determine that, please let me know.

Summery, even with Hugh's possible solution, this still does not work.

Thank you again,

ben


On Fri, Dec 20, 2013 at 10:26 AM, Hugh Waite
<hugh.waite at crocodile-rcs.com>wrote:

>  Hi Ben,
>
> Some NATs require symmetric RTP, so that source and destination ports are
> the same for incoming and outgoing media streams. This can be configured
> using the 'w' flag to rtpproxy_manage in the NATDETECT route.
>
> rtpproxy_manage("cow");
>
> Regards,
> Hugh
>
>
> On 20/12/2013 16:13, Benjamin Trent wrote:
>
>  Hey all,
>
>  **I apologize if this is a duplicate, I received a bounce back on my
> first email.
>
>  I have kamailio set up behind a nat(port restricted, with firewall rules
> to allow sip transactions and allowing rtpproxy packet handling if needed)
> on Amazon EC2. I can register and calls complete, however, the Caller(the
> one initiating the transaction) receives no rtp media feed. I am running
> with NAT enabled on kamailio and have rtpproxy installed listening on the
> public IP. Kamailio and the rtpproxy are communicating(I have verified via
> the kamailio debug logs). If I make a call between the exact same voip
> machines directly via local IP on the same Nat(skipping kamailio), the
> calls complete and they both receive feeds.
>
>  Both the Caller(party making the call) and the Callee(party receiving
> the call) are behind a Port Restricted Nat.
> This is a folder containing the debug output for one of these calls and
> the kamailio.cfg settings
>
>
> https://drive.google.com/folderview?id=0B9Foq0jDF8gLRlVNc001bTUtbFE&usp=sharing
>
>  Quick FYI, the Caller Display Name and the Callee SIP UserName are the
> same string. However, in my understanding about sip, the display name means
> pretty much nothing and is just a human readable string that does not
> effect packet transport. If I am wrong and should test with a different
> display name, let me know.
>
>
>  Thank you for the assistance,
>
>  ben
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
> Hugh Waite
> Principal Design Engineer
> Crocodile RCS Ltd.
>
>
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>
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