[asterisk-users] RTP over TCP

[adamk at 3a.hu]

Hi List,

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.

I can see the audio in the form of tcpdump, but neither party hears the 
other side.  Tcpdump also shows that while the call setup is via tcp, 
the audio is transmitted via udp.  I'm guessing this is the reason of 
silence.  The first setup is working because one of the gateways are 
supporting sip over tcp only and * accepts both.

my setup is pretty simple as * is only handing calls over to the 
gateways.  Relevant parts are below.

could anyone please confirm that it is an error, that asterisk sends the 
RTP stream via udp and this is the cause of the silence?  Is there any 
way to tell asterisk to use tcp only?  I'm aware of the drawbacks, but i 
still need to get this working.

I'd appreciate any help.

thanks
adam


sip.conf:

tcpenable=yes
tcpbindaddr=0.0.0.0

[ocs]
type=friend
host=192.168.1.1
context=ocs
qualify=yes
transport=tcp
nat=no
canreinvite=no
disallow=all
allow=alaw
allow=ulaw

[deverto4]
type=friend
host=172.18.200.4
context=deverto
qualify=yes
nat=no
canreinvite=yes
transport=tcp
disallow=all
allow=alaw
allow=ulaw

and the extensions.conf:

[deverto]
exten => _X.,1,Dial(SIP/${EXTEN}@ocs)
exten => _+X.,1,Dial(SIP/${EXTEN}@ocs)

[ocs]
exten => _X.,1,Dial(SIP/${EXTEN}@deverto4)
exten => _+X.,1,Dial(SIP/${EXTEN}@deverto4)