Hello,
The configuration is:
Jitsi #1 registered with sip2sip.info
Jitsi #2 registered with my server(s) (Kamailio edge proxy and
Kamailio registrar) on Amazon AWS running Kamailio from git master
I make a call from Jitsi #1 to Jitsi #2 (there are SRV records for
my domain in Amazon Route 53 so that sip2sip.info can find my
servers). The flow of the call is:
Jitsi #1 -> sip2sip.info -> Kamailio edge proxy
-> Kamailio registrar -> Kamailio edge proxy -> Jitsi
#2
The calls do not get established properly because the ACK is not
correctly routed. Calls between two Jitsi instances registered on
my servers works. Calls from Jitsi #2 to Jitsi #1 (my servers to
sip2sip.info) also work. This problem occurs whatever the value
of the "enable_double_rr" modparam.
The 200 OK (INVITE) sent from my servers back to sip2sip.info
looks OK and has a valid looking route-set:
SIP/2.0 200 OK
To: <sip:...callee...@...callee domain...>;tag=e5116094
Via: SIP/2.0/UDP 81.23.228.129:5060;rport=5060;branch=z9hG4bKdb93.87290485.0;i=0efbc,SIP/2.0/TLS 192.168.0.135:52379;rport=52379;received=a.b.c.d;branch=z9hG4bK-363631-23ca624fe9c119f229080992ed4fc143
Record-Route: <sip:Rlwj4KMCy9UUUAMK9JufE8VamABismk=@e.f.g.h:5061;transport=tls;r2=on;lr=on>,<sip:Rlwj4KMCy9UUUAMK9JufE8VamABismk=@10.244.155.159:5080;transport=tcp;r2=on;lr=on>,<sip:10.101.62.154;transport=tcp;lr=on>,<sip:10.244.155.159:5080;transport=tcp;r2=on;lr=on>,<sip:e.f.g.h;r2=on;lr=on>,<sip:81.23.228.129;r2=
on;lr;ftag=2d246de3;did=989.4a20eb73>,<sip:81.23.228.129:5061;transport=tls;r2=on;lr;ftag=2d246de3;did=989.4a20eb73>
CSeq: 2 INVITE
Call-ID: 38135c0b82a8a7d571352350d207b943@0:0:0:0:0:0:0:0
From: "Caller" <sip:...caller...@sip2sip.info>;tag=2d246de3
Contact: "callee" <sip:...callee...@10.6.201.224:60847;transport=tls;registering_acc=...callee domain...>
User-Agent: Jitsi2.3.4622.9647Linux
Content-Type: application/sdp
Content-Length: 838
...lots of SDP...
The ACK from sip2sip.info to my servers looks OK too:
ACK sip:...callee...@174.129.184.181:5060;transport=tls;registering_acc=...callee domain... SIP/2.0
Call-ID: 38135c0b82a8a7d571352350d207b943@0:0:0:0:0:0:0:0
CSeq: 2 ACK
Via: SIP/2.0/UDP 81.23.228.129:5060;branch=z9hG4bKdb93.87290485.2;i=0efbc
Via: SIP/2.0/TLS 192.168.0.135:52379;rport=52379;received=a.b.c.d;branch=z9hG4bK-363631-212696616e66a14286e3165485387895
From: "Caller" <sip:...caller...@sip2sip.info>;tag=2d246de3
To: "callee" <sip:...callee...@...callee domain...>;tag=e5116094
Max-Forwards: 69
Route: <sip:e.f.g.h;r2=on;lr=on>,<sip:10.244.155.159:5080;transport=tcp;r2=on;lr=on>,<sip:10.101.62.154;transport=tcp;lr=on>,<sip:Rlwj4KMCy9UUUAMK9JufE8VamABismk=@10.244.155.159:5080;transport=tcp;r2=on;lr=on>,<sip:Rlwj4KMCy9UUUAMK9JufE8VamABismk=@e.f.g.h:5061;transport=tls;r2=on;lr=on>
Contact: "Caller" <sip:...caller...@90.152.0.102:52379;transport=tls;registering_acc=sip2sip_info>
User-Agent: Jitsi2.2.4603.9615Windows 7
Content-Length: 0
But the ACK from from my edge proxy to my registrar looks very
broken:
ACK sip:Rlwj4KMCy9UUUAMK9JufE8VamABismk=@e.f.g.h:5061;transport=tls;r2=on;lr=on SIP/2.0
Call-ID: 38135c0b82a8a7d571352350d207b943@0:0:0:0:0:0:0:0
CSeq: 2 ACK
Via: SIP/2.0/TCP e.f.g.h:5060;branch=z9hG4bKdb93.444f42f24139f6dc512d70f61ca64be4.0
Via: SIP/2.0/UDP 81.23.228.129:5060;rport=5060;branch=z9hG4bKdb93.87290485.2;i=0efbc
Via: SIP/2.0/TLS 192.168.0.135:52379;rport=52379;received=a.b.c.d;branch=z9hG4bK-363631-212696616e66a14286e3165485387895
From: "Caller" <sip:...caller...@sip2sip.info>;tag=2d246de3
To: "callee" <sip:...callee...@...callee domain...>;tag=e5116094
Max-Forwards: 16
Route: <sip:10.244.155.159:5080;transport=tcp;r2=on;lr=on>,<sip:10.101.62.154;transport=tcp;lr=on>,<sip:Rlwj4KMCy9UUUAMK9JufE8VamABismk=@10.244.155.159:5080;transport=tcp;r2=on;lr=on>
Contact: "Caller" <sip:...caller...@a.b.c.d:52379;transport=tls;registering_acc=sip2sip_info>
User-Agent: Jitsi2.2.4603.9615Windows 7
Content-Length: 0
With debug turned up on the edge proxy I see the output from lines
673 and 723 indicating that the loose.c:after_strict() is run.
My edge proxy configuration is:
request_route {
route(REQINIT);
route(OPTIONS);
sip_trace();
setflag(F_SIPTRACE);
if (is_method("CANCEL")) {
if (t_check_trans()) {
route(RELAY);
}
exit;
}
route(WITHINDLG);
...
route[WITHINDLG] {
if (has_totag()) {
if (!loose_route()) {
switch($rc) {
case -2:
send_reply("403", "Forbidden");
exit;
default:
if (is_method("ACK")) {
if ( t_check_trans() ) {
route(RELAY);
exit;
} else {
exit;
}
}
send_reply("404","Not Found");
}
} else {
if ($rc == 2) {
setflag(F_OB_DRIVEN);
}
if (is_method("NOTIFY")) {
record_route();
}
route(RELAY);
}
exit;
}
}
Have I done something fundamentally stupid here, or is there some
obscure bug in the loose/strict routing code?
Regards,
Peter
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