[Asterisk-Users] Peer to Peer calls

Michiel van Baak michiel at vanbaak.info
Sun May 29 14:41:05 MST 2005


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.

We need more info before we can tell you the answer.
-- 
Michiel van Baak
http://michiel.vanbaak.info
michiel at vanbaak.info
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence."




More information about the asterisk-users mailing list