BREKEKE SIP SERVER FAILURE TO MATCH "BYE".

[Akinea78]

1. Brekeke Product Name and version:

Brekeke SIP Server version 2.4.8.6/286.3

2. Java version:

Java 6

3. OS type and the version:

Microsoft Windows XP SP3

4. UA (phone), gateway or other hardware/software involved:

Asterisk, Grandstream.

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html :

Pattern 4

6. Your problem:

Hello,

Brekeke Sip Server (version 2.4.8.6/286.3) fails to close SIP sessions in our infrastructure. In fact Brekeke fails to
match BYE, when we send BYE, it can't find the SIP dialog indded it sends a SIP code 481.
With an anterior version, Brekeke Sip Server (version 2.0.7.2/217), there's no such problem!!!



- This is the topology:


http://imageshack.us/photo/my-images/819/topologysip.png/

- this is a call flow sample that shows the error:

www.xxxx.yyy.zzz=IPBX
aaa.bbb.ccc.ddd=BREKEKE SIP SERVER
jjj.kkk.mmm.lll= Provider SIP (SIP Signalisation)
rrr.sss.ttt.uuu= Provider SIP (Voice Relay)

-------------------------------------------------------------



|Time | www.xxx.yyy.zzz | jjj.kkk.mmm.lll |
| | | aaa.bbb.ccc.ddd | | rrr.sss.ttt.uuu |
|512.284 | INVITE SDP (g711U g711A GSM) | | |SIP From: "0033130303030"

<sip:0033130303030@www.xxx.yyy.zzz To:<sip:0612121212@aaa.bbb.ccc.ddd
| |(5060) ------------------> (5060) | | |
|512.285 | 100 Trying| | | |SIP Status
| |(5060) <------------------ (5060) | | |
|512.359 | | INVITE SDP (g711U g711A GSM) | |SIP Request
| | |(5060) ------------------> (5060) | |
|512.389 | | 183 Session progress SDP (g711U) | |SIP Status
| | |(5060) <------------------ (5060) | |
|512.406 | 183 Session progress SDP (g711U) | | |SIP Status
| |(5060) <------------------ (5060) | | |
|521.160 | | 200 Ok SDP (g711U) | |SIP Status
| | |(5060) <------------------ (5060) | |
|521.161 | 200 Ok SDP (g711U) | | |SIP Status
| |(5060) <------------------ (5060) | | |
|521.162 | ACK | | | |SIP Request
| |(5060) ------------------> (5060) | | |
|521.163 | | ACK | | |SIP Request
| | |(5060) ------------------> (5060) | |
|521.257 | RTP (g711U) | | |RTP Num packets:683 Duration:13.637s SSRC:0x6148CA7F
| |(18928) ------------------> (10002) | | |
|521.257 | | RTP (g711U) | |RTP Num packets:682 Duration:13.617s SSRC:0x6148CA7F
| | |(10000) --------------------------------------> (25426) |
|521.290 | | RTP (g711U) | |RTP Num packets:468 Duration:9.339s SSRC:0xFF0000
| | |(10000) <-------------------------------------- (25426) |
|521.290 | RTP (g711U) | | |RTP Num packets:468 Duration:9.339s SSRC:0xFF0000
| |(18928) <------------------ (10002) | | |
|530.641 | | BYE | | |SIP Request
| | |(5060) <------------------ (5060) | |
|530.642 | | 481 Call Leg/Transaction Does Not Exist | |SIP Status
| | |(5060) ------------------> (5060) | |


-----------------------------------------------------------------


We ca note that Brekeke (aaa.bbb.ccc.ddd) sends SIP code 481 in response to BYE. Why?


- this is the corresponding SIP trace:

www.xxxx.yyy.zzz=IPBX
aaa.bbb.ccc.ddd=BREKEKE SIP SERVER
jjj.kkk.mmm.lll= Provider SIP (SIP Signalisation)
rrr.sss.ttt.uuu= Provider SIP (Voice Relay)

-------------------------------------------------------------

- Message from www.xxx.yyy.zzz to aaa.bbb.ccc.ddd:

INVITE sip:0612121212@aaa.bbb.ccc.ddd SIP/2.0
Via: SIP/2.0/UDP www.xxx.yyy.zzz:5060;branch=z9hG4bK7a837fab;rport
Max-Forwards: 70
From: "0033130303030" <sip:0033130303030@www.xxx.yyy.zzz>;tag=as610ddcb0
To: <sip:0612121212@aaa.bbb.ccc.ddd>
Contact: <sip:0033130303030@www.xxx.yyy.zzz>
Call-ID: 03ad8cfb7582091743253d84382780d4@www.xxx.yyy.zzz
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.17.2
Date: Mon, 13 Jun 2011 09:54:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 496706583 496706583 IN IP4 www.xxx.yyy.zzz
s=Asterisk PBX 1.6.2.17.2
c=IN IP4 www.xxx.yyy.zzz
t=0 0
m=audio 18928 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=sendrecv



- Message from aaa.bbb.ccc.ddd to www.xxx.yyy.zzz:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP www.xxx.yyy.zzz:5060;branch=z9hG4bK7a837fab;rport=5060
From: "0033130303030" <sip:0033130303030@www.xxx.yyy.zzz>;tag=as610ddcb0
To: <sip:0612121212@aaa.bbb.ccc.ddd>
Call-ID: 03ad8cfb7582091743253d84382780d4@www.xxx.yyy.zzz
CSeq: 102 INVITE
Server: Brekeke SIP Server rev.286.3
Content-Length: 0



- Message from aaa.bbb.ccc.ddd to jjj.kkk.mmm.lll:

INVITE sip:0000033612121212@jjj.kkk.mmm.lll SIP/2.0
Via: SIP/2.0/UDP aaa.bbb.ccc.ddd:5060;rport;branch=z9hG4bK7b46eb550f313a1431ba6-f0098ab3-afd9a06c
Via: SIP/2.0/UDP www.xxx.yyy.zzz:5060;branch=z9hG4bK7a837fab;rport=5060
Max-Forwards: 69
From: "0033130303030" <sip:0033033130303030@83.167.146.230>;tag=as610ddcb0
To: <sip:0000033612121212@jjj.kkk.mmm.lll>
Contact: <sip:0033130303030@aaa.bbb.ccc.ddd:5060>
Call-ID: 03ad8cfb7582091743253d84382780d4@www.xxx.yyy.zzz
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.17.2
Date: Mon, 13 Jun 2011 09:54:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Record-Route: <sip:aaa.bbb.ccc.ddd:5060;lr>
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 496706583 496706583 IN IP4 aaa.bbb.ccc.ddd
s=Asterisk PBX 1.6.2.17.2
c=IN IP4 aaa.bbb.ccc.ddd
t=0 0
m=audio 10000 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=sendrecv


- Message from 194.221.62..198 to aaa.bbb.ccc.ddd:

SIP/2.0 183 Session progress
Via: SIP/2.0/UDP aaa.bbb.ccc.ddd:5060;rport;branch=z9hG4bK7b46eb550f313a1431ba6-f0098ab3-afd9a06c
Via: SIP/2.0/UDP www.xxx.yyy.zzz:5060;branch=z9hG4bK7a837fab;rport=5060
From: "0033130303030" <sip:0033033130303030@83.167.146.230>;tag=as610ddcb0
To: <sip:0000033612121212@jjj.kkk.mmm.lll>;tag=4e0313ac670313ac4dd266b3a61eea
Contact: sip:0000033612121212@jjj.kkk.mmm.lll:5060
Call-ID: 03ad8cfb7582091743253d84382780d4@www.xxx.yyy.zzz
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Record-route: <sip:aaa.bbb.ccc.ddd:5060;lr>
Content-Type: application/sdp
Content-Length: 157

v=0
o=CARRIER 1307958511 1307958511 IN IP4 rrr.sss.ttt.uuu
s=SIP Call
c=IN IP4 rrr.sss.ttt.uuu
t=0 0
m=audio 25426 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20



- Message from aaa.bbb.ccc.ddd to www.xxx.yyy.zzz:


SIP/2.0 183 Session progress
Via: SIP/2.0/UDP www.xxx.yyy.zzz:5060;branch=z9hG4bK7a837fab;rport=5060
From: "0033130303030" <sip:0033130303030@www.xxx.yyy.zzz>;tag=as610ddcb0
To: <sip:0612121212@aaa.bbb.ccc.ddd>;tag=4e0313ac670313ac4dd266b3a61eea
Contact: sip:0000033612121212@aaa.bbb.ccc.ddd:5060
Call-ID: 03ad8cfb7582091743253d84382780d4@www.xxx.yyy.zzz
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Record-route: <sip:aaa.bbb.ccc.ddd:5060;lr>
Content-Type: application/sdp
Content-Length: 163

v=0
o=CARRIER 1307958511 1307958511 IN IP4 aaa.bbb.ccc.ddd
s=SIP Call
c=IN IP4 aaa.bbb.ccc.ddd
t=0 0
m=audio 10002 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20



- Message from jjj.kkk.mmm.lll to aaa.bbb.ccc.ddd:


SIP/2.0 200 Ok
Via: SIP/2.0/UDP aaa.bbb.ccc.ddd:5060;rport;branch=z9hG4bK7b46eb550f313a1431ba6-f0098ab3-afd9a06c
Via: SIP/2.0/UDP www.xxx.yyy.zzz:5060;branch=z9hG4bK7a837fab;rport=5060
From: "0033130303030" <sip:0033033130303030@83.167.146.230>;tag=as610ddcb0
To: <sip:0000033612121212@jjj.kkk.mmm.lll>;tag=4e0313ac670313ac4dd266b3a61eea
Contact: sip:0000033612121212@jjj.kkk.mmm.lll:5060
Call-ID: 03ad8cfb7582091743253d84382780d4@www.xxx.yyy.zzz
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Record-route: <sip:aaa.bbb.ccc.ddd:5060;lr>
Content-Type: application/sdp
Content-Length: 157

v=0
o=CARRIER 1307958520 1307958520 IN IP4 rrr.sss.ttt.uuu
s=SIP Call
c=IN IP4 rrr.sss.ttt.uuu
t=0 0
m=audio 25426 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20



- Message from aaa.bbb.ccc.ddd to www.xxx.yyy.zzz:


SIP/2.0 200 Ok
Via: SIP/2.0/UDP www.xxx.yyy.zzz:5060;branch=z9hG4bK7a837fab;rport=5060
From: "0033130303030" <sip:0033130303030@www.xxx.yyy.zzz>;tag=as610ddcb0
To: <sip:0612121212@aaa.bbb.ccc.ddd>;tag=4e0313ac670313ac4dd266b3a61eea
Contact: sip:0000033612121212@aaa.bbb.ccc.ddd:5060
Call-ID: 03ad8cfb7582091743253d84382780d4@www.xxx.yyy.zzz
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Record-route: <sip:aaa.bbb.ccc.ddd:5060;lr>
Content-Type: application/sdp
Content-Length: 163

v=0
o=CARRIER 1307958520 1307958520 IN IP4 aaa.bbb.ccc.ddd
s=SIP Call
c=IN IP4 aaa.bbb.ccc.ddd
t=0 0
m=audio 10002 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20


- Message from www.xxx.yyy.zzz to aaa.bbb.ccc.ddd:

ACK sip:0000033612121212@aaa.bbb.ccc.ddd:5060 SIP/2.0
Via: SIP/2.0/UDP www.xxx.yyy.zzz:5060;branch=z9hG4bK7e1e912a;rport
Route: <sip:aaa.bbb.ccc.ddd:5060;lr>
Max-Forwards: 70
From: "0033130303030" <sip:0033130303030@www.xxx.yyy.zzz>;tag=as610ddcb0
To: <sip:0612121212@aaa.bbb.ccc.ddd>;tag=4e0313ac670313ac4dd266b3a61eea
Contact: <sip:0033130303030@www.xxx.yyy.zzz>
Call-ID: 03ad8cfb7582091743253d84382780d4@www.xxx.yyy.zzz
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.17.2
Content-Length: 0


- Message from aaa.bbb.ccc.ddd to jjj.kkk.mmm.lll:

ACK sip:0000033612121212@jjj.kkk.mmm.lll:5060 SIP/2.0
Via: SIP/2.0/UDP aaa.bbb.ccc.ddd:5060;rport;branch=z9hG4bKfa697b502c0f2a1431ba6-f0098ab3-1faf735e
Via: SIP/2.0/UDP www.xxx.yyy.zzz:5060;branch=z9hG4bK7e1e912a;rport=5060
Max-Forwards: 69
From: "0033130303030" <sip:0033033130303030@83.167.146.230>;tag=as610ddcb0
To: <sip:0000033612121212@jjj.kkk.mmm.lll>;tag=4e0313ac670313ac4dd266b3a61eea
Contact: <sip:0033130303030@aaa.bbb.ccc.ddd:5060>
Call-ID: 03ad8cfb7582091743253d84382780d4@www.xxx.yyy.zzz
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.17.2
Record-Route: <sip:aaa.bbb.ccc.ddd:5060;lr>
Content-Length: 0


- Message from jjj.kkk.mmm.lll to aaa.bbb.ccc.ddd:

BYE sip:0033130303030@aaa.bbb.ccc.ddd:5060 SIP/2.0
Via: SIP/2.0/UDP jjj.kkk.mmm.lll:5060;branch=z9hG4bK7b46eb550f313a1431ba6-f0098ab3-afd9a06c
From: <sip:0000033612121212@jjj.kkk.mmm.lll>;tag=4e0313ac670313ac4dd266b3a61eea
To: "0033130303030" <sip:0033033130303030@83.167.146.230>;tag=as610ddcb0
Contact: sip:0000033612121212@jjj.kkk.mmm.lll:5060
Call-ID: 03ad8cfb7582091743253d84382780d4@www.xxx.yyy.zzz
CSeq: 1 BYE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Route: <sip:aaa.bbb.ccc.ddd:5060;lr>
Content-Length: 0


- Message from aaa.bbb.ccc.ddd to jjj.kkk.mmm.lll :

SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP jjj.kkk.mmm.lll:5060;branch=z9hG4bK7b46eb550f313a1431ba6-f0098ab3-afd9a06c
From: <sip:0000033612121212@jjj.kkk.mmm.lll>;tag=4e0313ac670313ac4dd266b3a61eea
To: "0033130303030" <sip:0033033130303030@83.167.146.230>;tag=as610ddcb0
Call-ID: 03ad8cfb7582091743253d84382780d4@www.xxx.yyy.zzz
CSeq: 1 BYE
Server: Brekeke SIP Server rev.286.3
Content-Length: 0

-------------------------------------------------------------




Best regards.

[taitan]

What kind of SIP entity is running at jjj.kkk.mmm.lll?
Is "(Very nice Sip Registrar/Proxy Server)" product name??

Why does it send the same "branch" parameter which Brekeke SIP server added?

> Via: SIP/2.0/UDP jjj.kkk.mmm.lll:5060;branch=z9hG4bK7b46eb550f313a1431ba6-f0098ab3-afd9a06c

Stop using "(Very nice Sip Registrar/Proxy Server)".

[Akinea78]

Hi taitan,

Firstly, thanks for your quick answer.

The "Very Nice Sip Registrar/Proxy Server" is our SIP provider so we can't deal without it.

The fact is that i don't understand why using the same branch is causing the problem while previous versions of BREKEKE SIP Server deals with the scenario without any problem.

Best regards

[taitan]

It seems that the current version of Brekeke SIP Server checks SIP parameter tight.


Set the following variable at the [Configuration] > [Advanced] page and restart the server.
-----------------------------
net.sip.spiral.use = false
-----------------------------