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[Linphone-developers] Linphone: BYE problem

From: Petr Kuba
Subject: [Linphone-developers] Linphone: BYE problem
Date: Wed, 21 Apr 2010 12:22:46 +0200
User-agent: Mozilla/5.0 (Windows; U; Windows NT 5.2; en-US; rv:1.9.1.9) Gecko/20100317 Thunderbird/3.0.4 
Hi,
We've met a problem in Linphone/3.2.0. We use command line version with auto-answer mode and Asterisk/1.6.1.11 as PBX.
After a few calls (Linphone is a callee) where caller terminates the calls, the following problem occurs: Linphone sends OK for BYE, but Linphone call does not terminate. Therefore the following call is not accepted. 

Complete log of SIP communication with included comments is below.

Thanks for help,
Petr Kuba

=============================================================================================

REGISTER sip:192.168.10.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.135:5060;rport;branch=z9hG4bK29616
From: <sip:address@hidden>;tag=18366
To: <sip:address@hidden>
Call-ID: 16278
CSeq: 5 REGISTER
Contact: <sip:address@hidden:5060;line=33a73e881a69e9b>
Authorization: Digest username="832", realm="asterisk", nonce="532b68bf", uri="sip:192.168.10.50", response="18734798b0b509cd4683ade8ce3d38ec", algorithm=MD5 
Max-Forwards: 70
User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
Expires: 900
Content-Length: 0

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.135:5060;branch=z9hG4bK29616;received=192.168.10.135;rport=5060 
From: <sip:address@hidden>;tag=18366
To: <sip:address@hidden>;tag=as0784eb4a
Call-ID: 16278
CSeq: 5 REGISTER
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5dea0df5"
Content-Length: 0

REGISTER sip:192.168.10.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.135:5060;rport;branch=z9hG4bK16800
From: <sip:address@hidden>;tag=18366
To: <sip:address@hidden>
Call-ID: 16278
CSeq: 6 REGISTER
Contact: <sip:address@hidden:5060;line=33a73e881a69e9b>
Authorization: Digest username="832", realm="asterisk", nonce="5dea0df5", uri="sip:192.168.10.50", response="1d8463cc6a9f1c030b3022181feef7fa", algorithm=MD5 
Max-Forwards: 70
User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
Expires: 900
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.135:5060;branch=z9hG4bK16800;received=192.168.10.135;rport=5060 
From: <sip:address@hidden>;tag=18366
To: <sip:address@hidden>;tag=as0784eb4a
Call-ID: 16278
CSeq: 6 REGISTER
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 900
Contact: sip:address@hidden:5060;line=33a73e881a69e9b;expires=900
Date: Tue, 20 Apr 2010 08:02:57 GMT
Content-Length: 0

=============================================================================================
Incoming call is automatically aanswered by linphone. Remote party disconnects. 
=============================================================================================
INVITE sip:address@hidden:5060;line=33a73e881a69e9b SIP/2.0
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK45b0ac87;rport
Max-Forwards: 70
From: "GTS" <sip:address@hidden>;tag=as1703441e
To: <sip:address@hidden:5060;line=33a73e881a69e9b>
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.11
Date: Tue, 20 Apr 2010 08:09:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 1326207081 1326207081 IN IP4 192.168.10.50
s=Asterisk PBX 1.6.1.11
c=IN IP4 192.168.10.50
t=0 0
m=audio 18004 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK45b0ac87;rport=5060
From: "GTS" <sip:address@hidden>;tag=as1703441e
To: <sip:address@hidden:5060;line=33a73e881a69e9b>
Call-ID: address@hidden
CSeq: 102 INVITE
User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
Content-Length: 0

SIP/2.0 101 Dialog Establishement
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK45b0ac87;rport=5060
From: "GTS" <sip:address@hidden>;tag=as1703441e
To: <sip:address@hidden:5060;line=33a73e881a69e9b>;tag=41
Call-ID: address@hidden
CSeq: 102 INVITE
Contact: <sip:address@hidden:5060>
User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
Content-Length: 0

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK45b0ac87;rport=5060
From: "GTS" <sip:address@hidden>;tag=as1703441e
To: <sip:address@hidden:5060;line=33a73e881a69e9b>;tag=41
Call-ID: address@hidden
CSeq: 102 INVITE
Contact: <sip:address@hidden:5060>
User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK45b0ac87;rport=5060
From: "GTS" <sip:address@hidden>;tag=as1703441e
To: <sip:address@hidden:5060;line=33a73e881a69e9b>;tag=41
Call-ID: address@hidden
CSeq: 102 INVITE
Contact: <sip:address@hidden:5060>
Content-Type: application/sdp
User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
Content-Length:   183

v=0
o=832 123456 654321 IN IP4 192.168.10.135
s=A conversation
c=IN IP4 192.168.10.135
t=0 0
m=audio 7078 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
ACK sip:address@hidden:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK023625f4;rport
Max-Forwards: 70
From: "GTS" <sip:address@hidden>;tag=as1703441e
To: <sip:address@hidden:5060;line=33a73e881a69e9b>;tag=41
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.11
Content-Length: 0

BYE sip:address@hidden:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK61bf61a5;rport
Max-Forwards: 70
From: "GTS" <sip:address@hidden>;tag=as1703441e
To: <sip:address@hidden:5060;line=33a73e881a69e9b>;tag=41
Call-ID: address@hidden
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.1.11
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK61bf61a5;rport=5060
From: "GTS" <sip:address@hidden>;tag=as1703441e
To: <sip:address@hidden:5060;line=33a73e881a69e9b>;tag=41
Call-ID: address@hidden
CSeq: 103 BYE
User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
Content-Length: 0

=============================================================================================
Linphone confirms BYE but it looks like it is still connected.
In the following call Linphone doesn't send 180 Ringing.
The call was interrupted by Linphone user (see CANCEL below) after more than 20s from INVITE. 
=============================================================================================
INVITE sip:address@hidden:5060;line=33a73e881a69e9b SIP/2.0
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK6a8182d0;rport
Max-Forwards: 70
From: "GTS" <sip:address@hidden>;tag=as1a191a92
To: <sip:address@hidden:5060;line=33a73e881a69e9b>
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.11
Date: Tue, 20 Apr 2010 08:15:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 729825232 729825232 IN IP4 192.168.10.50
s=Asterisk PBX 1.6.1.11
c=IN IP4 192.168.10.50
t=0 0
m=audio 19580 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK6a8182d0;rport=5060
From: "GTS" <sip:address@hidden>;tag=as1a191a92
To: <sip:address@hidden:5060;line=33a73e881a69e9b>
Call-ID: address@hidden
CSeq: 102 INVITE
User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
Content-Length: 0

SIP/2.0 101 Dialog Establishement
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK6a8182d0;rport=5060
From: "GTS" <sip:address@hidden>;tag=as1a191a92
To: <sip:address@hidden:5060;line=33a73e881a69e9b>;tag=18467
Call-ID: address@hidden
CSeq: 102 INVITE
Contact: <sip:address@hidden:5060>
User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
Content-Length: 0

CANCEL sip:address@hidden:5060;line=33a73e881a69e9b SIP/2.0
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK6a8182d0;rport
Max-Forwards: 70
From: "GTS" <sip:address@hidden>;tag=as1a191a92
To: <sip:address@hidden:5060;line=33a73e881a69e9b>
Call-ID: address@hidden
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.1.11
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK6a8182d0;rport=5060
From: "GTS" <sip:address@hidden>;tag=as1a191a92
To: <sip:address@hidden:5060;line=33a73e881a69e9b>;tag=18467
Call-ID: address@hidden
CSeq: 102 CANCEL
User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
Content-Length: 0

SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK6a8182d0;rport=5060
From: "GTS" <sip:address@hidden>;tag=as1a191a92
To: <sip:address@hidden:5060;line=33a73e881a69e9b>;tag=18467
Call-ID: address@hidden
CSeq: 102 INVITE
User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
Content-Length: 0

ACK sip:address@hidden:5060;line=33a73e881a69e9b SIP/2.0
Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK6a8182d0;rport
Max-Forwards: 70
From: "GTS" <sip:address@hidden>;tag=as1a191a92
To: <sip:address@hidden:5060;line=33a73e881a69e9b>;tag=18467
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.11
Content-Length: 0

=============================================================================================
REGISTER sip:192.168.10.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.135:5060;rport;branch=z9hG4bK20051
From: <sip:address@hidden>;tag=30845
To: <sip:address@hidden>
Call-ID: 13720
CSeq: 1 REGISTER
Contact: <sip:address@hidden:5060;line=33a73e881a69e9b>
Max-Forwards: 70
User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
Expires: 900
Content-Length: 0

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.135:5060;branch=z9hG4bK20051;received=192.168.10.135;rport=5060 
From: <sip:address@hidden>;tag=30845
To: <sip:address@hidden>;tag=as64b65fec
Call-ID: 13720
CSeq: 1 REGISTER
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39dc1532"
Content-Length: 0

REGISTER sip:192.168.10.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.135:5060;rport;branch=z9hG4bK18733
From: <sip:address@hidden>;tag=30845
To: <sip:address@hidden>
Call-ID: 13720
CSeq: 2 REGISTER
Contact: <sip:address@hidden:5060;line=33a73e881a69e9b>
Authorization: Digest username="832", realm="asterisk", nonce="39dc1532", uri="sip:192.168.10.50", response="79771bbc0f50febb9ee095909ffe00aa", algorithm=MD5 
Max-Forwards: 70
User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
Expires: 900
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.135:5060;branch=z9hG4bK18733;received=192.168.10.135;rport=5060 
From: <sip:address@hidden>;tag=30845
To: <sip:address@hidden>;tag=as64b65fec
Call-ID: 13720
CSeq: 2 REGISTER
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 900
Contact: sip:address@hidden:5060;line=33a73e881a69e9b;expires=900
Date: Tue, 20 Apr 2010 09:04:21 GMT
Content-Length: 0

=============================================================================================

--
   Petr Kuba, Project Manager
   OptimSys, s.r.o
   address@hidden
   Tel: +420 541 143 065
   Fax: +420 541 143 066
   http://www.optimsys.cz
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