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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050529/e35e07d3/attachment.htm
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@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."
Be aware to the codec compatibility between peers. Direct calls to the peer has to be under the same codec and initiation protocol. And, yes, if you have (eg.) SIP and GSM, and careinvite=yes, the media path dont pass through Asterisk. Denis Galvao. On 29/05/2005, at 18:32, 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._______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users