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

[Premalatha Kuppan]

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> 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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>
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