Re: [Linphone-developers] Linphone: BYE problem

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Re: [Linphone-developers] Linphone: BYE problem

From:	Joerg Bergmann
Subject:	Re: [Linphone-developers] Linphone: BYE problem
Date:	Fri, 30 Apr 2010 13:21:11 +0200
User-agent:	Mozilla/5.0 (X11; U; Linux i686; en-US; rv:1.9.1.9) Gecko/20100330 Fedora/3.0.4-1.fc11 Thunderbird/3.0.4

Dear Simon,

tried your new version 3.2.99.7. I want to disable video.
So I called
./configure --disable-video
and got the following error:
...
checking linux/videodev.h usability... yes
checking linux/videodev.h presence... yes
checking for linux/videodev.h... yes
checking linux/videodev2.h usability... yes
checking linux/videodev2.h presence... yes
checking for linux/videodev2.h... yes
checking for LIBV4L2... no
No libv4l2 found.
checking for LIBV4L1... no
No libv4l1 found.
configure: error:
Missing libv4l2. It is highly recommended to build with
libv4l2 headers and library. Many camera will won't work or will crash
your application if libv4l2 is not installed.
If you know what you are doing, you can use --disable-libv4l to disable

Why do I need that video-stuff when I want to diable video?
Any hints?

Joerg Bergmann

Am 30.04.2010 12:48, schrieb Simon Morlat:

The bug you are facing here is relative to this 3.2.99.3. I discovered
it after advicing you to use it. Sorry.
Please try lastest :
sources:
http://download.savannah.gnu.org/releases-noredirect/linphone/unstable/source/linphone-3.2.99.7.tar.gz
windows binary:
http://download.savannah.gnu.org/releases-noredirect/linphone/unstable/win32/linphone-3.2.99.7-setup.exe

Should work fine.

Note: the upload of the windows file is in progress and should be
finished in half an our max.

Simon

e jeudi 29 avril 2010 à 18:02 +0200, Petr Kuba a écrit :

Hi,

Please find the log attached. File complete.log contains complete log,
files 01.log to 05.log contain logs for individual calls.

Calls 1 and 2 were connected, calls 3, 4, and 5 were not. It means that
something wrong probably happened at the end of call 2.

We used version 3.2.99.3 on windows.

Thanks,
Petr

On 22.4.2010 17:22, Simon Morlat wrote:

Hi,

This is very strange.
COuld you please try with the new version here:
http://download.savannah.gnu.org/releases-noredirect/linphone/unstable/source/linphone-3.2.99.4.tar.gz
and if the problem still happens, please send a log collected with
linphonec -d 6
or linphone-3 --verbose

Simon

Le mercredi 21 avril 2010 à 12:22 +0200, Petr Kuba a écrit :

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

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




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[Linphone-developers] Linphone: BYE problem, Petr Kuba, 2010/04/21
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