100k question - does asterisk correctly handle following situations:
There are plenty of good documents on Asterisk, SIP and NAT on the voip-info.org wiki. Please look them up. There are also information within the configs/sip.conf.sample file within Asterisk.
If the media stream SHOULD NOT go through Asterisk, then it's up to the phones1. Asterisk is on a public IP Two SIP clients on separate networks, each of them are behind dynamic NAT gateway. Nat gateway does not have ALG. Media stream SHOULD NOT go thought asterisk.
to support NAT traversal properly and handle this, it's not an Asterisk problem. From Asterisk's point of view, we should not see that they are in fact behind NAT. Modern phones in combination with STUN and a decent NAT device supports this.
2. Even worst case - three clients, two of them on one site, second is on another site. For example extensions 500 and 600 are on the same site and in the same subnet and extension 1000 is on another site/network. There are PAT FW/gateways with dynamic public IP in front of clients and those are symmetric NAT/FW.
The task - clients registering on Asterisk server, calling each other and RTP should not go via asterisk. So, media stream should go directly from one client to another.
If Asterisk is on a public IP, again: it's up to the phones. It's still not an Asterisk problem.
Yes, but you need to pick the right phone, the right NAT/FW and have a lot of patience :-)I want to know:
1. Is it possible? - yes/no. Implementation should involve asterisk and SIP clients and not involving third party hardware products - ALG, session border controllers or so on.
2. If it is possible, what are requirements for SIP clients.
Good NAT traversal support.
From Asterisk's point of view, all of these phones are on a public IP and we3. What configuration changes should be done on Asterisk server and on a sip clients.
do not give them any NAT traversal support. If you want detailed configurations, there are several consultants available that can help you with that (including my company).
And final question - if it is NOT possible with Asterisk, do you know anIt is possible with Asterisk and every other SIP server. With your requirements, it's completely a client-side problem.
open source product which works in above stated scenarios and you've
actually tested it.
Best regards, /Olle _______________________________________________ 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
