[OpenSIPS-Users] 477 Send failed from Opensips via TCP

[Bogdan-Andrei Iancu]

You need to use a media relay (rtpproxy or mediaproxy) to make RTP go 
through the nat.

Regards,
Bogdan

Premalatha Kuppan wrote:
> What about TLS, i can establish the call using TLS, but audio not 
> heard. TLS is also behind NAT. I can RTP packets from TLS client to 
> Opensips. I believe RTP is Peer to peer.
> But, in my setup, since asterisk doesn't support TLS/TCP, i forward 
> the call to opensips for protocol switching  to connect to TLS client.
>
> I dono why audio is not heard. Is it also related to NAT. Do i need to 
> change anything in opensips.cfg ?
>
> On Thu, Jul 29, 2010 at 9:02 PM, Andrew Pogrebennyk 
> <andrew.pogrebennyk at portaone.com 
> <mailto:andrew.pogrebennyk at portaone.com>> wrote:
>
>     On 29.07.2010 18:12, Bogdan-Andrei Iancu wrote:
>     > oh yes, you need to use the nathelper module to fix the
>     signalling - to
>     > replace the private ips with public Ips in contact - during
>     registration
>     > and during calls..
>
>     IIRC UA which uses TCP behind NAT shall create connection to proxy
>     and keep it forever. And proxy shall reuse this connection, instead
>     of attempt to create new one. This means proxy shall maintain
>     map of translation of registered AORs to current TCP (or SCTP)
>     connections and pass requests to the connection which corresponds to
>     this AOR. This is the implication of running the connection-aware
>     protocols. As for the first part, many UAs do offer TCP connection
>     keeping, as for second - I'd be surprised if OpenSIPS keeps the
>     connections map. I didn't have a chance to try it though.
>
>     --
>     Sincerely,
>     Andrew Pogrebennyk
>
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-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
20 - 24 September 2010, Frankfurt, Germany
www.voice-system.ro