[Users] "BYE" messages change path - sr-users

6 Sep 2006

Hi users,

         This is ngrep report regarding my problem;

-----------------<when a sip proxy behind NAT and the user registerd to that
sip-proxy calls>-------------------------
U 82.102.69.105:39871 -> 81.21.33.35:5060
INVITE sip:99106883@81.21.33.35:5060 SIP/2.0.
To: "99106883"<sip:99106883@81.21.33.35:5060>.
From: "12345"<sip:12345@81.21.33.35:5060>;tag=c86b66ad8b9187c8.
Via: SIP/2.0/UDP 192.168.1.100:5060
;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport.
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKf638e18b56022ea3.
Call-ID: a78d5c993a9dd6b4(a)192.168.1.102.
CSeq: 47344 INVITE.
Record-Route: <sip:192.168.1.100:5060>.
Contact: <sip:192.168.1.100:5060>.
Max-Forwards: 69.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.
Content-Type: application/sdp.
Supported: replaces.
User-Agent: Grandstream BT110 1.0.8.23.
Content-Length: 361.
.
v=0.
o=line2 8000 8000 IN IP4 192.168.1.102.
s=SIP Call.
c=IN IP4 82.102.69.105.---------------------------------------------> this
is the NAT for the sip-proxy ,
t=0 0.
m=audio 5004 RTP/AVP 18 4 2 97 9 0 101.
a=fmtp:97 mode=20.
a=fmtp:101 0-11.
a=ptime:20.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:97 iLBC/8000.
a=rtpmap:9 G722/16000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=sendrecv.

-----------------------------<when my SIP-SERVER standing on public ip
recieved the "INV" message from the
above"-----------------------------------

U 81.21.33.35:5060 -> pstngw:5060
INVITE sip:99106883@pstngw:5060 SIP/2.0.
Record-Route: <sip:99106883@81.21.33.35
:5060;nat=yes;ftag=c86b66ad8b9187c8;lr=on>.
To: "99106883"<sip:99106883@81.21.33.35:5060>.
From: "12345"<sip:12345@81.21.33.35:5060>;tag=c86b66ad8b9187c8.
Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK0ab.9522bc25.0.
Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105
;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport=39871.
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKf638e18b56022ea3.
Call-ID: a78d5c993a9dd6b4(a)192.168.1.102.
CSeq: 47344 INVITE.
Record-Route: <sip:192.168.1.100:5060>.
Contact: <sip:82.102.69.105:39871>.
Max-Forwards: 16.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.
Content-Type: application/sdp.
Supported: replaces.
User-Agent: Grandstream BT110 1.0.8.23.
Content-Length: 360.
.
v=0.
o=line2 8000 8000 IN IP4 192.168.1.102.
s=SIP Call.
c=IN IP4 81.21.33.35.--------------------------------------------->This is
my SIP-SERVER standing on public  ip (and is sending to pstngw)
t=0 0.
m=audio 60516 RTP/AVP 18 4 2 97 9 0 101.
a=fmtp:97 mode=20.
a=fmtp:101 0-11.
a=ptime:20.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:97 iLBC/8000.
a=rtpmap:9 G722/16000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=sendrecv.

---------------------------------------------<and the pstngw is making
call>-----------------------------------------------

U 81.21.33.35:5060 -> pstngw:39871
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP
192.168.1.100:5060;received=82.102.69.105;branch=z9hG4bK-d87543-ac3b034487ba16641188e4c9e5ad0664-1--d87543-;rport=39871,SIP/2.0/UDP
192.168.1.102;branch=z9hG4bK01c64e769ba6c176.
From: "12345"<sip:12345@81.21.33.35:5060>;tag=9ca5c1fe3b43aad6.
To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E6010EC-1764.
Date: Wed, 06 Sep 2006 10:27:40 GMT.
Call-ID: 72d0e4e8adca2ab2(a)192.168.1.102.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 32352 INVITE.
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Disposition: session;handling=required.
Content-Length: 237.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 54 9902 IN IP4 pstngw
s=SIP Call.
c=IN IP4 81.21.33.35.-------------------------------------<in pstngw it
noticed the contact header of SIP-SERVER>---------------------
t=0 0.
m=audio 60518 RTP/AVP 18 100.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
a=rtpmap:100 X-NSE/8000.
a=fmtp:100 192-194.
a=ptime:20.

---------------------------------------------<after talking some time if any
of  UA hung the phone (here i am showing from pstn hung up) the result goes
like this----------------

U pstngw:52991 -> 81.21.33.35:5060
BYE sip:99106883@81.21.33.35:5060;nat=yes;ftag=1bab08cdaad1d6e8;lr=on
SIP/2.0.
Via: SIP/2.0/UDP  81.21.38.15:5060.
From: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E66E3D8-1292.
To: "12345"<sip:12345@81.21.33.35:5060>;tag=1bab08cdaad1d6e8.
Date: Wed, 06 Sep 2006 10:35:07 GMT.
Call-ID: c2dd3fb9554ef6e4(a)192.168.1.102.
User-Agent: Cisco-SIPGateway/IOS-12.x.
Max-Forwards: 6.
Route: <sip:192.168.1.100:5060>,
<sip:82.102.69.105:39871>.------------------------>
What is happening here ???????
Timestamp: 1157538919.
CSeq: 101 BYE.
Content-Length: 0.
.

#
U 81.21.33.35:5060 ->
192.168.1.100:5060--------------------------------------------------->here
unexpected problem arises for me it have to use 82.102.69.105
BYE sip:192.168.1.100:5060 SIP/2.0.
Record-Route: <sip:81.21.33.35;ftag=7E66E3D8-1292;lr=on>.
Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bKb0f1.3fe32691.0.
Via: SIP/2.0/UDP  pstngw:5060.
From: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E66E3D8-1292.
To: "12345"<sip:12345@81.21.33.35:5060>;tag=1bab08cdaad1d6e8.
Date: Wed, 06 Sep 2006 10:35:07 GMT.
Call-ID: c2dd3fb9554ef6e4(a)192.168.1.102.
User-Agent: Cisco-SIPGateway/IOS-12.x.
Max-Forwards: 5.
Route: <sip:82.102.69.105:39871>.
Timestamp: 1157538919.
CSeq: 101 BYE.
Content-Length: 0.
P-hint: rr-enforced.
So the call never ends up from SIP SERVER to other party

 Make some comments on above and assist me where am I going wrong:

and i am using www.openser.org  pstn default script as ser.cfg

                     Hope to get some help
                         Thanks,
Ravi.