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.<br><br>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.<br>
<br clear="all">-- Nathan Clemons<br>
<br><br><div class="gmail_quote">On Fri, Apr 23, 2010 at 12:21 PM, <span dir="ltr"><<a href="mailto:adamk@3a.hu">adamk@3a.hu</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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>
</blockquote></div><br>