Here is my setup
7960(A)--Firewall/PAT--dsl---------Internet--------dsl--Firewall/NAT---7960(B) | | | | 7960(C)--NAT--cable----------------- -----dsl -- Asterisk
(A) can communicate with (C) only when C is configured with canreinvite=no. The call gets dropped in few seconds if canreinvite is set to yes for C.
(A) and (B) can communicate fine when both sides have canreinvite=yes.
Since (C) is not working with canreinvite, traffic goes thru Asterisk server. This brings the Dsl connection to asterisk to a crawl. It is so bad that even a idle ssh connection gets disconnected.
Is it possible to configure C so that reinvite works. If not what kind of a bandwidth should I have for Asterisk server. Currently it has a upload of 128K.
The codec currently getting used is ULAW. Even if I configure 7960's to use g729, show sip channel reports as using ULAW.
Thanks, ==ratnakar
If you are moving your traffic from behind a NAT, your Asterisk server must have a G.729 license to terminate the traffic, since Asterisk must be the media proxy for the stream. As you are connecting endpoints together that are behind NAT, you would need multiple G.729 licenses - one for every device that would be concurrently talking to the Asterisk server. I do not believe that it is possible to configure C so that reinvite works, though I would be interested in how you do it if you are able to make that function without Asterisk being a media channel proxy (quasi-border session controller.)
You should have at least 56kbps for G.729, in my experience, unless you have no other traffic on the end legs of the diagram. G.729 uses less than 32kbps during normal circumstances, but other TCP traffic needs to squeeze in (as you have discovered.) Your Asterisk server will of course need to have N*(leg bandwidth) capacity, where N is the number of legs active at any one time.
JT _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
