[Asterisk-Users] P2P RTP without SIP re-invites

[Low, Adam]

Apologies for the belated reply but I've spent the weekend fighting DDoS attacks against Superbowl sites ... )c;

Ok, well I am not sure what went wrong with previous testing but I have tried this again with Cisco 7940's and Cisco AS5300's and indeed the RTP stream flows directly between end-points retaining SIP signalling via Asterisk. This is exactly the operation I had hoped for. I had previously tested with my home 7940 which it behind NAT without success and so will re-test this this evening.

Thanks for all the responses and related discussion on clustering Asterisk, thanks to those I now have a running cluster of 3  Asterisk servers each with mirrored sip.conf and extensions.conf built dynamically from a MySQL backend database.

Rgds, Adam

-----Original Message-----
From: Brancaleoni Matteo [mailto:mbrancaleoni at espia.it]
Sent: 31 January 2004 13:20
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] P2P RTP without SIP re-invites


hi
> 
> I guess this would work if both Alice and Bob were NAT'ed on the inside of the same
> NAT box. The problem is that if Alice and Bob both have NAT=yes and CANREINVITE=yes
> and they're on separate NAT'ed networks, the call is broken. So it's a dangerous
> configuration.

nope. I have a public * server (beta server for a free VoIP service),
on a public IP. and some sip phones around , like one in my home,
behind nat, one in my office (another nat) and some others
at my coworkers home... all behind nat. and are different nat
box, do you agree? that works ok, I have RTP passing
directly from one endpoint to the other... no RTP
on the public * server.
No stun is used. The phones are budgetones in this case.
All are configured with nat=yes on asterisk side.
or I missing something?
-- 
Brancaleoni Matteo <mbrancaleoni at espia.it>
Espia - Emmegi Srl

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