----- Original Message -----
From: "Michiel van Baak" <[EMAIL PROTECTED]>
To: <[email protected]>
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
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