22 May
2007
22 May
'07
4:20 p.m.
Dear Sir,
This is the Packet in initial Proxy:
UA ----> Proxy1 ---> proxy2 ...... -----> called UA
Voice is ok, Invite, ACK, are fine, only BYE has trouble when called
hang up first.
The call signal packets close to Proxy 1
thanks
-----Invite from UA
sip:0908900000@sip.edu;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:2051;branch=z9hG4bK-kblxgjx7vsj6;rport
From: "0709802020 kl" <sip:0709802020@sip.edu>;tag=ooqjvbi7ql
To: <sip:0908900000@sip.edu;user=phone>
Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:0709802020@192.168.0.12:2051;line=5bj76qg6>;flow-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/6.5.1
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 424
v=0
o=root 1112263304 1112263304 IN IP4 192.168.0.12
s=call
c=IN IP4 192.168.0.12
t=0 0
m=audio 50530 RTP/AVP 18 4 0 8 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:
3GR6bicZ4axFL3M0iBo9rx84mGt9TNWJIfgyMprL
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.0.12:2051;branch=z9hG4bK-
kblxgjx7vsj6;rport=2051;received=59.120.208.208
From: "0709802020 kl" <sip:0709802020@sip.edu>;tag=ooqjvbi7ql
To: <sip:0908900000@sip.edu;user=phone>
Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
CSeq: 1 INVITE
Content-Length: 0
------ Invite from 1st Proxy to 2nd Proxy
INVITE sip:0908900000@sip.org:5060;user=phone SIP/2.0
Record-Route: <sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on>
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK2e5a.835198e5.0
Via: SIP/2.0/UDP
192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK-
kblxgjx7vsj6;rport=2051
From: "0709802020 kl" <sip:0709802020@sip.edu>;tag=ooqjvbi7ql
To: <sip:0908900000@sip.edu;user=phone>
Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
CSeq: 1 INVITE
Max-Forwards: 69
Contact: <sip:0709802020@59.120.208.208:2051;line=5bj76qg6>;flow-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/6.5.1
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 426
v=0
o=root 1112263304 1112263304 IN IP4 192.168.0.12
s=call
c=IN IP4 1.2.3.4
t=0 0
m=audio 35054 RTP/AVP 18 4 0 8 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:
3GR6bicZ4axFL3M0iBo9rx84mGt9TNWJIfgyMprL
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK2e5a.835198e5.0
Via: SIP/2.0/UDP
192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK-
kblxgjx7vsj6;rport=2051
From: "0709802020 kl" <sip:0709802020@sip.edu>;tag=ooqjvbi7ql
To: <sip:0908900000@sip.edu;user=phone>
Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
CSeq: 1 INVITE
Content-Length: 0
------- Proxy receive Callee Answer packet
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK2e5a.835198e5.0
Via: SIP/2.0/UDP
192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK-
kblxgjx7vsj6;rport=2051
Record-Route: <sip:16.3.0.100;lr=on;ftag=ooqjvbi7ql>
Record-Route: <sip:5.6.7.8;lr=on;ftag=ooqjvbi7ql>
Record-Route: <sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on>
From: "0709802020 kl" <sip:0709802020@sip.edu>;tag=ooqjvbi7ql
To: <sip:0908900000@sip.edu;user=phone>;tag=as1b41fbeb
Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
CSeq: 1 INVITE
User-Agent: Gentrice_IPPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:7777@163.30.0.199>
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 23878 23878 IN IP4 163.30.0.199
s=session
c=IN IP4 16.3.0.100
t=0 0
m=audio 57124 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
----------OK relay to UA
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK-
kblxgjx7vsj6;rport=2051
Record-Route: <sip:16.3.0.100;lr=on;ftag=ooqjvbi7ql>
Record-Route: <sip:5.6.7.8;lr=on;ftag=ooqjvbi7ql>
Record-Route: <sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on>
From: "0709802020 kl" <sip:0709802020@sip.edu>;tag=ooqjvbi7ql
To: <sip:0908900000@sip.edu;user=phone>;tag=as1b41fbeb
Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
CSeq: 1 INVITE
User-Agent: Gentrice_IPPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:7777@163.30.0.199>
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 23878 23878 IN IP4 163.30.0.199
s=session
c=IN IP4 1.2.3.4
t=0 0
m=audio 35054 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
------ACK received from UA
ACK sip:7777@163.30.0.199 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:2051;branch=z9hG4bK-us1wrcp6otau;rport
Route: <sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on>
Route: <sip:5.6.7.8;lr=on;ftag=ooqjvbi7ql>
Route: <sip:16.3.0.100;lr=on;ftag=ooqjvbi7ql>
From: "0709802020 kl" <sip:0709802020@sip.edu>;tag=ooqjvbi7ql
To: <sip:0908900000@sip.edu;user=phone>;tag=as1b41fbeb
Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:0709802020@192.168.0.12:2051;line=5bj76qg6>;flow-id=1
Content-Length: 0
------ACK relay to next proxy
ACK sip:7777@163.30.0.199 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK2e5a.835198e5.2
Via: SIP/2.0/UDP
192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK-
us1wrcp6otau;rport=2051
Route: <sip:5.6.7.8;lr=on;ftag=ooqjvbi7ql>
Route: <sip:16.3.0.100;lr=on;ftag=ooqjvbi7ql>
From: "0709802020 kl" <sip:0709802020@sip.edu>;tag=ooqjvbi7ql
To: <sip:0908900000@sip.edu;user=phone>;tag=as1b41fbeb
Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
CSeq: 1 ACK
Max-Forwards: 69
Contact: <sip:0709802020@192.168.0.12:2051;line=5bj76qg6>;flow-id=1
Content-Length: 0
----Called party hangup
BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0
Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb>
Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb>
Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0
Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0
Via: SIP/2.0/UDP 16.3.0.100;branch=z9hG4bKc74c.321ef5e3.0
Via: SIP/2.0/UDP 163.30.0.199:5060;branch=z9hG4bK75e8d6f3;rport=5060
From: <sip:0908900000@sip.edu;user=phone>;tag=as1b41fbeb
To: "0709802020 kl" <sip:0709802020@sip.edu>;tag=ooqjvbi7ql
Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
CSeq: 102 BYE
User-Agent: Gentrice_IPPBX
Max-Forwards: 67
Content-Length: 0
-----Looped Bye ...
BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0
Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb>
Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb>
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.3714a4e7.0
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.2714a4e7.0
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.1714a4e7.0
Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0
Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0
Via: SIP/2.0/UDP 16.3.0.100;branch=z9hG4bKc74c.321ef5e3.0
Via: SIP/2.0/UDP 163.30.0.199:5060;branch=z9hG4bK75e8d6f3;rport=5060
From: <sip:0908900000@sip.edu;user=phone>;tag=as1b41fbeb
To: "0709802020 kl" <sip:0709802020@sip.edu>;tag=ooqjvbi7ql
Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
CSeq: 102 BYE
User-Agent: Gentrice_IPPBX
Max-Forwards: 64
Content-Length: 0
在 2007/5/22 下午 8:29 時，Klaus Darilion 寫到：
Further, describe your setup:
IP addresses of the proxies
IP addresses of the clients
kaiser wrote:
> Hi,
>
> Thanks for your help, this is the invite from initial proxy:
>
>
> sip:0908900000@sip.ipox.org.tw:5060;user=phone SIP/2.0
> Record-Route: <sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on>
> Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK2e5a.835198e5.0
> Via: SIP/2.0/UDP
> 192.168.0.12:2051;received=59.120.208.208;branch=z9hG4bK-
> kblxgjx7vsj6;rport=2051
>
> From: "0909802020 kl" <sip:0709802020@sip.edu.tw>;tag=ooqjvbi7ql
> To: <sip:0908900000@sip.edu.tw;user=phone>
> Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
> CSeq: 1 INVITE
> Max-Forwards: 69
> Contact: <sip:0909802020@59.120.208.208:2051;line=5bj76qg6>;flow-id=1
>
>
> do you mean we have to add username in record_route?
>
> best regards
> kaiser
>
>
>
>
> 在 2007/5/22 上午 1:21 時，Klaus Darilion 寫到：
>
>> Hi!
>>
>> AS this is probably a loose-route problem you have to provide a
>> log of
>> the initial INVITE too.
>>
>> regards
>> klaus
>>
>> kaiser wrote:
>>> Dear sir,
>>> We install 3 openser 1.1.0 in public internet, and we find a BYE
>>> relay issue to caller SIP UA.
>>> UA(a)-------Proxy1 --------Proxy2 ----Proxy 3 ---------UA(b)
>>> We make call from UA(a) to UA (b), if UA(b) send Bye first, UA
>>> (a) can
>>> not get Bye.
>>> The packet is from Proxy2 to Proxy1 as following
>>> BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0
>>> Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb>
>>> Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb>
>>> Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0
>>> Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0
>>> Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0
>>> Via: SIP/2.0/UDP
>>> 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060
>>> From: <sip:0908900000@sip.edu.tw;user=phone>;tag=as1b41fbeb
>>> To: "0909802020 kl"
<sip:0909802020@sip.edu.tw>;tag=ooqjvbi7ql
>>> Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
>>> CSeq: 102 BYE
>>> User-Agent: Gentrice_IPPBX
>>> Max-Forwards: 67
>>> Content-Length: 0
>>> Then bye is looping in proxy1.
>>> BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0
>>> Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb>
>>> Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb>
>>> Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.1714a4e7.0
>>> Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0
>>> Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0
>>> Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0
>>> Via: SIP/2.0/UDP
>>> 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060
>>> From: <sip:0908900000@sip.edu.tw;user=phone>;tag=as1b41fbeb
>>> To: "0909802020 kl"
<sip:0909802020@sip.edu.tw>;tag=ooqjvbi7ql
>>> Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B
>>> CSeq: 102 BYE
>>> User-Agent: Gentrice_IPPBX
>>> Max-Forwards: 66
>>> Content-Length: 0
>>> We used to use default script for proxy1, but the same...
>>> And we do record_route, loose_route relay as well.
>>> Anyone know what happen?
>>> best regards
>>> Thrli
>>> _______________________________________________
>>> Users mailing list
>>> Users(a)openser.org
>>> http://openser.org/cgi-bin/mailman/listinfo/users
>