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 [EMAIL PROTECTED] 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." _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
