On Fri, Apr 23, 2010 at 3:21 PM, <ad...@3a.hu> wrote: > i have to put an * between two other SIP gateways and due to some > circumstances, i have to use sip over tcp. With 1.6.2.6 this is working > fine: sip gw A (deverto4) sends the call, i hand it over to sip gw B > (ocs) and that's about it. In the other direction however (ocs -> me -> > deverto4) the call setup is complete but there is no audio.
Don't do it. There have been any number of posts to asterisk-users begging asterisk to bend over backwards to accommodate Microsoft's foray into the world of VoIP. Nobody seems to be asking Microsoft to build a stack compatible with the rest of the world of VoIP. I disagree that sending SIP over TCP is superior to sending it over UDP. Think about it for a second. UDP is 'unreliable' in that lost packets are not rebroadcast. Now let's say you have an 'unreliable' connection where it's just barely good enough that the SIP call setup goes through, but the RTP stream immediately fails. Why would that be superior to having the SIP call setup getting dropped? The end result of no reliable voice is the same, but now you have a funkier debug condition that's going to be more complex to track down. I personally think it would be superior to see the bad connection as early in call setup as possible. And of course, SIP over UDP is the way the rest of the world works. If anybody wants to speak up about a framework that supports BOTH SIP over UDP AND SIP over TCP, maybe somebody already built a middleware layer that will let Microsoft join the world of voip. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users