[SR-Users] Caller Not Receiving RTP Feed

[Hugh Waite]

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
>
>
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-- 
Hugh Waite
Principal Design Engineer
Crocodile RCS Ltd.

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