[Asterisk-Users] Peer to Peer calls

[Michael J. Tubby G8TIC]

----- Original Message ----- 
From: "Michiel van Baak" <michiel at vanbaak.info>
To: <asterisk-users at lists.digium.com>
Sent: Sunday, May 29, 2005 10:41 PM
Subject: Re: [Asterisk-Users] Peer to Peer calls


> On 00:32, Mon 30 May 05, Cenk Yabas wrote:
>> Can anybody please answer this.
>> Both clients are behind different NAT's. 
>> One of them starts a SIP call to the other through Asterisk. 
>> Asterisk sets up the call. 
>> Issues reinvite and connects them together.
>> After this point does the media stream flow through Asterisk or Peer to
>> Peer?
>> Does such a call use any system resources of Asterisk server after
>> connection?
>> Thank you in advance.
> 
> Did you test this ?
> My experience is the 'reinvite' does not work in the setup
> you descripted. I always have to set 'canreinvite=no' in
> asterisk config or the audio will not come through.
> If you have only one phone on both NAT's and you can do
> port-forwording on both firewalls, it can work, but that
> scenario is highly uncommon.
> The audio stream is setup on some random port, so your
> firewall will block this by default.
> 

*But* If your firewall is SIP-aware - for example a Cisco 837 ADSL
router with IOS 12.3 - then it should be able to fix up the firewall rules
dynamically so that when the phones in the inside (behind the firewall)
re-invite it should inspect the SIP on udp/5060 and see the invitation and
open the appropriate UDP port(s) for the RTP stream.

Mike