[OpenSIPS-Users] Testing opensips with a public client

[Ignacio Gonzalez]

Hello everybody, I'm trying to connect two sip clients, an xlite softphone
behind a NAT and a Sip User Agent build with JAIN-SIP, this UA is in a host
with a public IP. The JAIN Agent REGISTER successfully but when I use the
xlite to call the agent, opensips does not redirect the OK response from
the User Agent. Here are the SIP messages that the opensips proxy receive:

INVITE sip:2427002 at domain.org SIP/2.0
Via: SIP/2.0/UDP 189.181.48.143:1737
;branch=z9hG4bK-d87543-db6cdc02d2345b76-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:2427001 at 189.181.48.143:1737>
To: "2427002"<sip:2427002 at domain.org>
From: "2427001"<sip:2427001 at domain.org>;tag=1d740661
Call-ID: 406db678fb29d214MzFjY2NjYTIwNWFjZDQwNWFmNzQ0OGFjYzI0YzY4N2U.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1002tx stamp 29712
Content-Length: 847

v=0
o=- 3 2 IN IP4 189.181.48.143
s=<CounterPath eyeBeam 1.5>
c=IN IP4 189.181.48.143
t=0 0
m=audio 1742 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 3 : DyqmwKwg iKJbLF1I 10.188.123.26 61510
a=alt:2 2 : 9s6ct+vW fiwpiJbN 192.188.200.1 61510
a=alt:3 1 : JBvmcu7C aeAEMO9J 192.188.129.1 61510
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:4E160A53513C4828AD841903785A5B73
m=video 1740 RTP/AVP 115 34
a=alt:1 3 : 7KF3a7ry trG+qu1k 10.188.123.26 48170
a=alt:2 2 : 7Or3DIwR SUvlz0xo 192.188.200.1 48170
a=alt:3 1 : Pe/okXax QdIQD2uP 192.188.129.1 48170
a=fmtp:115 QCIF=1 MAXBR=1960
a=fmtp:34 QCIF=1 MAXBR=1960
a=rtpmap:115 H263-1998/90000
a=rtpmap:34 H263/90000
a=sendrecv
a=x-rtp-session-id:8EA1C094BF1E43718B27CF80B76C38EA


SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 189.181.48.143:1737
;received=189.181.48.143;branch=z9hG4bK-d87543-6b13cd3fa93abf2e-1--d87543-;rport=1737
To: "2427002"<sip:2427002 at domain.org>
From: "2427001"<sip:2427001 at domain.org>;tag=1d740661
Call-ID: 406db678fb29d214MzFjY2NjYTIwNWFjZDQwNWFmNzQ0OGFjYzI0YzY4N2U.
CSeq: 1 INVITE
Server: OpenSIPS (1.8.0-notls (i386/linux))
Content-Length: 0


SIP/2.0 200 OK
To: "2427002" <sip:2427002 at domain.org>;tag=42e61ae4
CSeq: 2 INVITE
Call-ID: 406db678fb29d214MzFjY2NjYTIwNWFjZDQwNWFmNzQ0OGFjYzI0YzY4N2U.
Via: SIP/2.0/UDP 189.181.48.143:1737
;received=189.181.48.143;branch=z9hG4bK-d87543-6b13cd3fa93abf2e-1--d87543-;rport=1737
From: "2427001" <sip:2427001 at domain.org>;tag=1d740661
Record-Route: <sip:domain.org;lr;did=bcb.7d8c5155>
Contact: <sip:2427002 at 70.70.70.70:8000;lr>
Content-Type: application/sdp
Content-Length: 196



v=0
o=- 0 0 IN IP4 70.70.70.70
s=-
c=IN IP4 70.70.70.70
t=0 0
m=audio 50000 RTP/AVP 0
c=IN IP4 70.70.70.70
a=sendrecv
m=video 50010 RTP/AVP 34
a=sendonly
a=rtpmap:34 H263/90000


...  (retries OK messages)

Can any body tell me why xlite do not receive the OK response? or why
opensips do not send the OK response?
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