[Kamailio-Users] Loose Route / Re-Invite (brandon at cryy.com)

[Iñaki Baz Castillo]

El Miércoles, 3 de Febrero de 2010, Brandon Armstead escribió:
> Hello,
> 
>    Please refer to sip trace:  http://pastebin.org/86010.
> 
> On LINE 294, PSTN sends the Re-INVITE

And there is the error (in PSTN_PROXY server):


U +0.000104 MY_SBC_PROXY:5060 -> PSTN_PROXY_IP:5060
SIP/2.0 200 OK.
To: "" <sip:7143642300 at sip-dev01.myproxy.com>;tag=437a03f314fce56fo0.
From: <sip:8778242288 at sip-
dev01.myproxy.com>;tag=a9d5ed0-13c4-4b69c983-4c452b6f-2d9ebe5b.
Call-ID: 4cf8f943-c7a1f05f at 10.160.100.32.
CSeq: 1 INVITE.
Via: SIP/2.0/UDP 
PSTN_PROXY_IP:5060;rport=5060;received=208.94.157.10;branch=z9hG4bK-48f8c-4b69cd08-4c52ec66-35ca5cf4.
Record-Route: 
<sip:MY_REGISTRAR_PROXY;lr=on;ftag=a9d5ed0-13c4-4b69c983-4c452b6f-2d9ebe5b>.
Record-Route: 
<sip:MY_SBC_PROXY;lr=on;ftag=a9d5ed0-13c4-4b69c983-4c452b6f-2d9ebe5b>.
Contact: "" <sip:7143642300 at 10.160.100.32:5060>.
Server: Linksys/PAP2T-5.1.6(LS).
Content-Length: 255.
Content-Type: application/sdp.
 
v=0.
o=- 130969 130969 IN IP4 10.160.100.32.
s=-.
c=IN IP4 10.160.100.32.
t=0 0.
m=audio 16476 RTP/AVP 0 100 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:100 NSE/8000.
a=fmtp:100 192-193.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
 
U +0.031146 PSTN_PROXY_IP:5060 -> MY_SBC_PROXY:5060
ACK sip:7143642300 at 10.160.100.32:5060 SIP/2.0.
From: <sip:8778242288 at sip-
dev01.myproxy.com>;tag=a9d5ed0-13c4-4b69c983-4c452b6f-2d9ebe5b.
To: "" <sip:7143642300 at sip-dev01.myproxy.com>;tag=437a03f314fce56fo0.
Call-ID: 4cf8f943-c7a1f05f at 10.160.100.32.
CSeq: 1 ACK.
Via: SIP/2.0/UDP 
PSTN_PROXY_IP:5060;branch=z9hG4bK-48f97-4b69cd08-4c52ed08-32120aec.
Max-Forwards: 69.
Contact: <sip:18778242288 at PSTN_PROXY_IP:5060;transport=udp>.
Route: <sip:MY_SBC_PROXY;lr;ftag=437a03f314fce56fo0>.
Route: <sip:MY_REGISTRAR_PROXY;lr;ftag=437a03f314fce56fo0>.
Content-Length: 0.



This is a re-INVITE and the UA (MY_SBC_PROXY) replies a 200 with the natted 
Contact (as usual).
Then the PSTN_PROXY server creates the ACK by setting such private SIP URI as 
RURI. This is incorrect, this RURI *MUST* be the remote target set in the 
initial INVITE/200 (the Contact the UA received when the dialog was 
established). This remote target cannot change within a dialog (as Alex and me 
have explained in this thread).

PSTN_PROXY is buggy.



-- 
Iñaki Baz Castillo <ibc at aliax.net>