30 Oct
2009
30 Oct
'09
2:11 a.m.
Hi Klaus.
I use the rtpproxy_offer + answer functions without any flags. I've listed
parts of my code below.
route[2]
{
if (is_method("BYE|CANCEL"))
{
unforce_rtp_proxy();
} else if (is_method("INVITE")) {
if(has_body("application/sdp")){
if(rtpproxy_offer())
t_on_reply("1");
}else{
t_on_reply("2"); #this will handle the initial INVITE that has no SDP
}
}
if(is_method("ACK") && has_body("application/sdp")){
rtpproxy_answer();
}
}
onreply_route[1]
{
if (has_body("application/sdp"))
rtpproxy_answer();
if (isbflagset(6))
{
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nty=yes');
search_append('m:.<sip:[^>[:cntrl:]]*', ';nty=yes');
if(cmp_str("MY_IP","$si"))
else {
fix_nated_contact();
}
}
exit;
}
onreply_route[2]
{
if (has_body("application/sdp"))
rtpproxy_offer();
if (isbflagset(6))
{
search_append('Contact:.*sip:[:cntrl:]]*', ';nty=yes');
search_append('m:.<sip:[^>[:cntrl:]]*', ';nty=yes');
if(cmp_str("MY_IP","$si"))
else {
fix_nated_contact();
}
}
exit;
}
Klaus Darilion-2 wrote:
Do you use force_rtpproxy() or the ...offer() and ...answer() functions?
Do you use any flags when call the functions?
klaus
General Lee schrieb:
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Hi,
I am currently integrating an H323 to SIP gateway with Kamailio and
trying
to route all calls through RTP Proxy.
I have a problem where RTP Proxy treats both the incoming H323 gateway
call
(Caller) and outgoing SIP call (Callee) as the 'caller' in the RTP Proxy
syslog. RTP Proxy doesn't assign a 'callee' therefore not able to setup a
call (See syslog below). I have configured Kamailio to accept the 'ACK'
with
SDP and this is working correctly.
When I enable H323 Fast connect and the SDP is included in the INVITE,
the
call connects correctly and is routed through RTP Proxy. I feel the
problem
is related to RTP Proxy receiving an INVITE from the H323-SIP gateway
without SDP.
Can anyone explain why RTP Proxy treats both the incoming H323 Gateway
call
and outgoing SIP call as the 'caller' in the RTP Proxy syslog. How can I
make RTP Proxy treat the incoming H323 call as the 'callee'?
Thanks,
More information below
*******************************************************************************
My H323 endpoints use H323 Slow Connect, so when the H323-SIP Gateway
delivers the 'INVITE' to Kamailio there is no SDP included in the INVITE.
I
added a 'onreply_route' to the Kamailio configuration file which handles
the
'ACK' with SDP which is working correctly.
All of my SIP calls (Signalling + Media) are forced though RTP Proxy and
I
would like to force all H323-SIP Gateway calls through RTP Proxy.
When placing a call from my H323 endpoint to my SIP UA, the RTP Proxy
syslog
records the incoming and outgoing call, however RTP Proxy states that the
‘callee’ is actually the 'caller'. The RTP Proxy syslog also states that
the
caller is the caller so there is no 'callee' (see below). In the syslogs
both the ‘callee’ and ‘caller’ are recognised as the ‘caller’ so RTP
Proxy
has no callee to send the traffic back to.
When the INVITE is received the ‘callee’ is populated in the syslogs as
the
‘caller’. The H323 Gateway call is not recorded until after the ‘ACK’
with
SDP is received from the gateway.
Oct 27 17:33:05 rtpproxy[24086]: INFO:handle_command: pre-filling
caller's
address with 72.19.211.106:49620 (Should be callee)
Oct 27 17:33:05 rtpproxy[24086]: INFO:handle_command: pre-filling
caller's
address with 72.19.211.106:49622 (Should be callee)
Then after the ‘ACK’ is received from the Gateway, the H323 call is
mentioned in the syslog as well as the other caller who is supposed to be
the callee.
Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling
caller's
address with 69.72.11.51:10204
Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling
caller's
address with 69.72.11.51:10214
Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling
caller's
address with 72.19.211.106:49620
Oct 27 17:33:06 rtpproxy[24086]: INFO:handle_command: pre-filling
caller's
address with 72.19.211.106:49622
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