[asterisk-users] RTP over TCP

[David Backeberg]

On Fri, Apr 23, 2010 at 3:21 PM,  <adamk at 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.