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/ ... terns.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/81 ... gysip.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.
BREKEKE SIP SERVER FAILURE TO MATCH "BYE".
Moderator: Brekeke Support Team
BREKEKE SIP SERVER FAILURE TO MATCH "BYE".
Akinea Internet
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)".
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)".
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
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
Akinea Internet