On Sat, 2010-04-24 at 10:56 -0500, Michael Graves wrote: > On Fri, 23 Apr 2010 23:11:06 +0200, ad...@3a.hu wrote: > > >Hi Guys, > > > >On 04-23-2010 21:40, Nathan Clemons wrote: > >> SIP is just the control protocol, and can be negotiated over TCP or UDP. > >> The > >> actual payload is done over RTP, which is a UDP-based protocol. > >> > > > >thanks, for both of you for pointing this out. i was obviously on the > >wrong track here. since i see the rtp traffic via tcpdump, i'm going to > >ask the other gw's sysadmin to see into this, maybe there is some > >logging on the other side. > > > >> If you had to add firewall exceptions/PAT config for the TCP SIP traffic, > >> you'll also need to add the same for RTP traffic as well. > >> > > > >this is a private pilot network for testing purposes, no internet > >connection, no nat, no firewall. it's like the 90s :) > > Actually, it is more common for RTP to be over UDP, but RTP over TCP is > possible. In fact, this is the default RTP mode for M$ Office > Communications Server. > > I beleive that it may be possible to use RTP over TCP in Asterisk as > there was someone work on this inorder to have interop with M$. >
No, that was just sip over tcp (instead of udp) I friend of mine had an * talking to M$, so that was one of the reasons for early deployment of an 1.6.x asterisk... -- _____________________________________________________________________ -- 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
