<p>I don't think RTP can be sent over TCP at all, it would defeat the whole purpose of RTP. Even if you somehow manage to do so, voice quality will go down the drain.<br></p>
<p>Zeeshan A Zakaria</p>
<p>--<br>
Sent from my Android phone with K-9 Mail.</p>
<p><blockquote type="cite">On 2010-04-23 3:27 PM, <<a href="mailto:adamk@3a.hu">adamk@3a.hu</a>> wrote:<br><br>Hi List,<br>
<br>
i have to put an * between two other SIP gateways and due to some<br>
circumstances, i have to use sip over tcp. With 1.6.2.6 this is working<br>
fine: sip gw A (deverto4) sends the call, i hand it over to sip gw B<br>
(ocs) and that's about it. In the other direction however (ocs -> me -><br>
deverto4) the call setup is complete but there is no audio.<br>
<br>
I can see the audio in the form of tcpdump, but neither party hears the<br>
other side. Tcpdump also shows that while the call setup is via tcp,<br>
the audio is transmitted via udp. I'm guessing this is the reason of<br>
silence. The first setup is working because one of the gateways are<br>
supporting sip over tcp only and * accepts both.<br>
<br>
my setup is pretty simple as * is only handing calls over to the<br>
gateways. Relevant parts are below.<br>
<br>
could anyone please confirm that it is an error, that asterisk sends the<br>
RTP stream via udp and this is the cause of the silence? Is there any<br>
way to tell asterisk to use tcp only? I'm aware of the drawbacks, but i<br>
still need to get this working.<br>
<br>
I'd appreciate any help.<br>
<br>
thanks<br>
adam<br>
<br>
<br>
sip.conf:<br>
<br>
tcpenable=yes<br>
tcpbindaddr=0.0.0.0<br>
<br>
[ocs]<br>
type=friend<br>
host=192.168.1.1<br>
context=ocs<br>
qualify=yes<br>
transport=tcp<br>
nat=no<br>
canreinvite=no<br>
disallow=all<br>
allow=alaw<br>
allow=ulaw<br>
<br>
[deverto4]<br>
type=friend<br>
host=172.18.200.4<br>
context=deverto<br>
qualify=yes<br>
nat=no<br>
canreinvite=yes<br>
transport=tcp<br>
disallow=all<br>
allow=alaw<br>
allow=ulaw<br>
<br>
and the extensions.conf:<br>
<br>
[deverto]<br>
exten => _X.,1,Dial(SIP/${EXTEN}@ocs)<br>
exten => _+X.,1,Dial(SIP/${EXTEN}@ocs)<br>
<br>
[ocs]<br>
exten => _X.,1,Dial(SIP/${EXTEN}@deverto4)<br>
exten => _+X.,1,Dial(SIP/${EXTEN}@deverto4)<br>
<br>
<br>
<br>
<br>
<br>
<br>
--<br>
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</blockquote></p>