[sip-comm-dev] any volunteers to fix multichatting?


#1

Hey all,

We've been having multichatting about 80% implemented for a while now.
The problem is that despite the significant amount of work that we've
already spent on it, the remaining 20% consist in a number of bugs and
misfeatures that make it feel quite shaky. We are probably going to
remove it from our 1.0 branch for that reason.

I was wondering if anyone on this list would like to work on completing
it. This would include resolving bugs (most trivial), writing unit
tests, and if necessary adding essential features. According to my
personal estimations the effort would amount to approximately 2 or 3
person months.

Don't be shy and speak up if you are interested!

Cheers
Emil

···

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#2

hey,
   I am interested in it and has read partial source code before, is there
somebody can guide me to join it ?
Really appreciated.

···

2009/6/24 Emil Ivov <emcho@sip-communicator.org>

Hey all,

We've been having multichatting about 80% implemented for a while now.
The problem is that despite the significant amount of work that we've
already spent on it, the remaining 20% consist in a number of bugs and
misfeatures that make it feel quite shaky. We are probably going to
remove it from our 1.0 branch for that reason.

I was wondering if anyone on this list would like to work on completing
it. This would include resolving bugs (most trivial), writing unit
tests, and if necessary adding essential features. According to my
personal estimations the effort would amount to approximately 2 or 3
person months.

Don't be shy and speak up if you are interested!

Cheers
Emil

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#3

Hi, all

Just for saying to those who are working on multichatting, that I'm on too.

Regards,

Valentin

···

Le 24 juin 09 à 12:37, Emil Ivov a écrit :

Hey all,

We've been having multichatting about 80% implemented for a while now.
The problem is that despite the significant amount of work that we've
already spent on it, the remaining 20% consist in a number of bugs and
misfeatures that make it feel quite shaky. We are probably going to
remove it from our 1.0 branch for that reason.

I was wondering if anyone on this list would like to work on completing
it. This would include resolving bugs (most trivial), writing unit
tests, and if necessary adding essential features. According to my
personal estimations the effort would amount to approximately 2 or 3
person months.

Don't be shy and speak up if you are interested!

Cheers
Emil

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#4

Hey Martin,

Martin Han wrote:

hey,
   I am interested in it and has read partial source code before, is
there somebody can guide me to join it ?

I've already mentioned this offlist but still - glad to hear it. If you
(and/or anyone else for that matter) is interested in working on this
then you could start by using the existing code and trying to uncover
problems with it. You could do this on a protocol by protocol basis
starting with XMPP, then moving through IRC, MSN, ICQ/AIM, and Yahoo!
Messenger.

In addition to fixing the issues you could also have a look at the unit
tests (slicks). We already have some running for XMPP (courtesy of
Sympho) but they may need further work, and we'd also need to make them
run for the rest of the projects.

Good luck, and let us know if you have any questions
Cheers
Emil

···

Really appreciated.

2009/6/24 Emil Ivov <emcho@sip-communicator.org
<mailto:emcho@sip-communicator.org>>

    Hey all,

    We've been having multichatting about 80% implemented for a while now.
    The problem is that despite the significant amount of work that we've
    already spent on it, the remaining 20% consist in a number of bugs and
    misfeatures that make it feel quite shaky. We are probably going to
    remove it from our 1.0 branch for that reason.

    I was wondering if anyone on this list would like to work on completing
    it. This would include resolving bugs (most trivial), writing unit
    tests, and if necessary adding essential features. According to my
    personal estimations the effort would amount to approximately 2 or 3
    person months.

    Don't be shy and speak up if you are interested!

    Cheers
    Emil

    ---------------------------------------------------------------------
    To unsubscribe, e-mail:
    dev-unsubscribe@sip-communicator.dev.java.net
    <mailto:dev-unsubscribe@sip-communicator.dev.java.net>
    For additional commands, e-mail:
    dev-help@sip-communicator.dev.java.net
    <mailto:dev-help@sip-communicator.dev.java.net>

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#5

Hello,

I have been able to make and receive calls with good quality for some time but when I updated to new sources, I have bad outgoing connections. The audio from Sip-Communicator is choppy and gets more and more laggey. Looking at the debug output, it says that it is able to transmit in:

debug: We will be able to transmit in:
debug: unknown encoding format mpegaudio/rtp

Sorry, I am confused by this. Because of copyright issues, mpegaudio has to be separately downloaded doesn't it? Would appreciate any help in figuring this out. I include the debug output below.

BTW, is Stun disabled by default? How does SC do the NAT traversal without it?

thanks, Kim

deploy-os-specific-bundles:
run:

Welcome to Felix.

···

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

debug: Service Impl: net.java.sip.communicator.impl.configuration.ConfigurationActivator [ STARTED ]
debug: Using config file in $HOME/.sip-communicator: C:\trunk.pure\sip-communicator.xml
debug: Service Impl: net.java.sip.communicator.impl.configuration.ConfigurationActivator [REGISTERED]
Info: Resource manager ... [REGISTERED]
Info: UI Service...[ STARTED ]
Info: UI Service ...[REGISTERED]
Info: Resource registered net.java.sip.communicator.plugin.defaultresourcepack.DefaultColorPackImpl@1995d80
Info: Resource registered net.java.sip.communicator.plugin.defaultresourcepack.DefaultImagePackImpl@6e293a
Info: Resource registered net.java.sip.communicator.plugin.defaultresourcepack.DefaultLanguagePackImpl@1319c
Info: Resource registered net.java.sip.communicator.plugin.defaultresourcepack.DefaultSettingsPackImpl@1479feb
Info: Resource registered net.java.sip.communicator.plugin.defaultresourcepack.DefaultSoundPackImpl@1592174
Info: Default resources ... [REGISTERED]
debug: Started.
debug: SIP Protocol Provider Factory ... [REGISTERED]
debug: SIP Communicator Version: sip-communicator-1.2.1
debug: Started.
created a deviceConfiguration
MediaControl created
debug: Media Service ... [REGISTERED]
Starting mediaServiceimpl
deviceConfiguration.initialize
debug: Started.
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.icq accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.icq accounts
IOException in readRegistry: java.io.EOFException
debug: SIP Protocol Provider Factory ... [REGISTERED]
debug: Discovered 2 stored net.java.sip.communicator.impl.protocol.sip accounts
debug: Loading account net.java.sip.communicator.impl.protocol.sip.acc1244603766502
Info: Gibberish protocol implementation [STARTED].
Info: SSH protocol implementation [STARTED].
Info: Stun server address(null)/port(null) not set (or invalid). Disabling STUN.
Info: Network Address Manager ...[ STARTED ]
Info: Network Address Manager Service ...[REGISTERED]
Info: Zeroconf protocol implementation [STARTED].
Info: IRC protocol implementation [STARTED].
Info: DICT protocol implementation [STARTED].
debug: Service Impl: net.java.sip.communicator.impl.contactlist.ContactlistActivator [ STARTED ]
debug: Starting the meta contact list implementation.
debug: Service Impl: net.java.sip.communicator.impl.contactlist.ContactlistActivator [REGISTERED]
Info: RSS protocol implementation [STARTED].
debug: User-Agent set to Info-Tell Operator/1.2.1
debug: New history service registered.
debug: Starting the msg history implementation.
Info: Message History Service ...[REGISTERED]
debug: New history service registered.
debug: Starting the call history implementation.
Info: Call History Service ...[REGISTERED]
Info: Audio Notifier Service...[ STARTED ]
Info: Audio Notifier Service ...[REGISTERED]
debug: Service Impl: net.java.sip.communicator.impl.keybindings.KeybindingsActivator [ STARTED ]
Info: Notification Service...[ STARTED ]
Info: Notification Service ...[REGISTERED]
debug: Registering default event IncomingMessage/PopupMessageAction/null/null
debug: Registering default event IncomingMessage/SoundAction/resources/sounds/incomingMessage.wav/null
Info: Looking for Audio capturer
Info: DirectSound Capture Supported = true
debug: Registering default event IncomingCall/PopupMessageAction/null/null
debug: presence initialized with :true, true, 30, 3600 for null
debug: Registering default event ProactiveNotification/PopupMessageAction/null/null
debug: Registering default event SecurityMessage/PopupMessageAction/null/null
debug: Registering default event CallSecurityOn/SoundAction/resources/sounds/zrtpSecure.wav/null
debug: Registering default event CallSecurityError/SoundAction/resources/sounds/zrtpAlert.wav/null
Info: UI Service...[ STARTED ]
Info: UI Service ...[REGISTERED]
debug: Stored account for id SIP:kim_sip@phone.wenco.com for package net.java.sip.communicator.impl.protocol.sip
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP:kim_sip@phone.wenco.com
debug: All contacts loaded for account SIP:kim_sip@phone.wenco.com
debug: Loading account net.java.sip.communicator.impl.protocol.sip.acc1244609754970
debug: presence initialized with :true, true, 30, 3600 for null
debug: Stored account for id SIP:doha_amber@phone.wenco.com for package net.java.sip.communicator.impl.protocol.sip
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP:doha_amber@phone.wenco.com
debug: All contacts loaded for account SIP:doha_amber@phone.wenco.com
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.jabber accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.msn accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.yahoo accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.gibberish accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.ssh accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.zeroconf accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.irc accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.dict accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.rss accounts
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event: net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.branding.AboutWindowPluginComponent@a6e0a9]
Info: ABOUT WINDOW ... [REGISTERED]
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event: net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.branding.AboutWindowPluginComponent@f81402]
Info: CHAT ABOUT WINDOW ... [REGISTERED]
Info: Swing Notification ...[ STARTING ]
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event: net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.updatechecker.UpdateCheckActivator$UpdateMenuButtonComponent@154ea79]
Info: No config file specified for update checker. Disabling update checks
debug: Updates are disabled. Faking latest version.
Info: Swing Notification ...[REGISTERED]
Info: added the following popup handler : SIP Communicator popups
21:28:14.218 INFO: doha.impl.phone.PhoneActivator.start().59 Phone Service ...[Starting]
21:28:14.218 FINEST: doha.impl.phone.PhoneActivator.start().62 [entry] start
Info: Systray Service...[ STARTED ]
Info: Systray Service ...[REGISTERED]
PhoneActivator Start!
Info: Phone Service ...[Starting]
OperatorServiceImpl started!
Session created
startListener
Session.startListener turned on!
users.getCallerIDDocument
loadDom: c:\db\admin\callerids.xml
Opening connection to: jdbc:mysql://ljade.com:3306/mydatabase
1 Database.java:412 open()
2 Database.java:763 getXML()
3 Database.java:750 getXMLFile()
4 DomServiceImpl.java:546 loadDom()
5 Users.java:241 getCallerIDDocument()
6 Users.java:37 <init>()
7 OperatorServiceImpl.java:108 <init>()
8 PhoneActivator.java:64 start()
9 SecureAction.java:589 startActivator()
Opening SQL Database connection...
debug: The provider changed state from: RegistrationState=Unregistered to: RegistrationState=Registering
debug: The provider changed state from: RegistrationState=Unregistered to: RegistrationState=Registering
debug: generated max forwards: Max-Forwards: 70

debug: generated via headers:Via: SIP/2.0/UDP 192.168.2.95:5060

debug: generated contactHeader:Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>

Info: DirectSoundAuto: Committed ok
Info: JavaSound Capture Supported = true
Info: JavaSoundAuto: Committed ok
debug: No FMJ javasound detected: net.sf.fmj.media.cdp.javasound.CaptureDevicePlugger
Info: Looking for video capture devices
Info: Scanning for configured Audio Devices.
debug: Found 2 capture devices: [Ljavax.media.CaptureDeviceInfo;@8d41f2
Info: Found DirectSoundCapture as an audio capture device.
Info: Scanning for configured Video Devices.
Info: No Video Device was found.
defaultMediaControl.initialize
MediaControl initialized: net.java.sip.communicator.impl.media.codec.EncodingConfiguration@1e492d8
debug: Adding package : net.java.sip.communicator.impl
debug: Adding package : net.sf.fmj
debug: Registering new protocol prefix list : [javax, com.sun, com.ibm, net.java.sip.communicator.impl, net.sf.fmj]
Info: Creating datasource for:dsound://
MediaContol initiProcessor: net.java.sip.communicator.impl.media.codec.EncodingConfiguration@1e492d8
debug: Codec net.java.sip.communicator.impl.media.codec.audio.alaw.JavaEncoder is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.audio.alaw.DePacketizer is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.audio.alaw.Packetizer is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.audio.ulaw.Packetizer is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.video.h264.JNIEncoder is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.video.h264.Packetizer is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.video.h264.JNIDecoder is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.video.ImageScaler is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.audio.speex.JavaEncoder is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.audio.speex.JavaDecoder is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaEncoder is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaDecoder is successfully registered
connection: com.mysql.jdbc.Connection@149249e
... SQL Database connected in 0.863 seconds
query: select * from Callerids where ider = 'admin_callerids_0'
Local file is up-to-date localMark: 232 serverMark: 232
Cache.initializing...
Initial Interaction created - used for plugin
Info: trying database in phoneactivator
Info: Phone Service ...[REGISTERED]
Phone Opened Connection
21:28:15.184 INFO: doha.impl.phone.PhoneActivator.start().95 trying database in phoneactivator
21:28:15.185 INFO: doha.impl.phone.PhoneActivator.start().104 Phone Service ...[REGISTERED]
21:28:15.185 FINEST: doha.impl.phone.PhoneActivator.start().109 [exit] start
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event: net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.whiteboard.WhiteboardMenuItem@58d7c2]
Info: WHITEBOARD... [REGISTERED]
Info: Handling registration of a new Account Wizard.
Info: Loading gibberish account wizard.
Info: Handling registration of a new Account Wizard.
Info: Gibberish account registration wizard [STARTED].
Info: Loading ssh account wizard.
Info: Handling registration of a new Account Wizard.
Info: SSH account registration wizard [STARTED].
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event: net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.callhistoryform.ExtendedCallHistorySearchItem@959fa1]
Info: EXTENDED CALL HISTORY SEARCH... [REGISTERED]
Info: Loading rss account wizard.
Info: Handling registration of a new Account Wizard.
Info: RSS account registration wizard [STARTED].
Info: Loading zeroconf account wizard.
Info: Handling registration of a new Account Wizard.
Info: Zeroconf account registration wizard [STARTED].
Info: Loading irc account wizard.
Info: Handling registration of a new Account Wizard.
Info: IRC account registration wizard [STARTED].
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event: net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.contactinfo.ContactInfoMenuItem@423606]
Info: CONTACT INFO... [REGISTERED]
debug: Found 2 already installed providers.
debug: Adding protocol provider SIP
debug: Service Impl: net.java.sip.communicator.plugin.keybindingchooser.KeybindingChooserActivator [ STARTED ]
Info: PREFERENCES PLUGIN... [REGISTERED]
Info: Handling registration of a new Account Wizard.
Started the CMSActivator
Info: EXTENDED CALL HISTORY SEARCH... [REGISTERED]
Info: SIMPLE ACCOUNT REGISTRATION ...[STARTED]
debug: Starting the ShutdownTimeout service.
Info: Mailbox plug-in...[STARTING]
21:28:15.370 INFO: org.osgi.framework.BundleActivator.start().75 Mailbox plug-in...[STARTING]
debug: Starting the mailbox implementation.
debug: Found 2 already installed providers.
debug: Adding protocol provider SIP
warn: Missing resource for key: outgoing
21:28:15.374 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: outgoing
21:28:15.795 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: browse
21:28:15.796 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: incoming
warn: Missing resource for key: browse
warn: Missing resource for key: incoming
21:28:15.942 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: browse
warn: Missing resource for key: browse
warn: Missing resource for key: waitTime
21:28:15.944 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: waitTime
warn: Missing resource for key: maxMessageTime
21:28:15.953 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: maxMessageTime
warn: Missing resource for key: confirm
21:28:15.955 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: confirm
warn: Missing resource for key: default
21:28:15.955 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: default
fileLocation: null
21:28:15.959 INFO: org.osgi.framework.BundleActivator.start().89 Mailbox plug-in...[STARTED]
Info: Mailbox plug-in...[STARTED]
debug: sent request=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER

From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc

To: <sip:kim_sip@phone.wenco.com>
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Content-Length: 0

debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95;received=206.81.217.93

From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc

To: <sip:kim_sip@phone.wenco.com>
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95;received=206.81.217.93

From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc

To: <sip:kim_sip@phone.wenco.com>;tag=as61098680
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5,realm="phone.wenco.com",nonce="2c5e294e"
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: Authenticating a Register request.
debug: trying to authenticate using : MD5, kim_sip, phone.wenco.com, true, 2c5e294e, 00000001, xyz, REGISTER, sip:phone.wenco.com, , null
debug: Created authorization header: Authorization: Digest username="kim_sip",realm="phone.wenco.com",nonce="2c5e294e",uri="sip:phone.wenco.com",response="5e9d462a45fd6f9abbe705e4b09a7d0d",algorithm=MD5

debug: Returning authorization transaction.
debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKd01785901b2b5e8e6974fd2708407ef9;received=206.81.217.93

From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc

To: <sip:kim_sip@phone.wenco.com>
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKd01785901b2b5e8e6974fd2708407ef9;received=206.81.217.93

From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc

To: <sip:kim_sip@phone.wenco.com>;tag=as61098680
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Date: Fri, 26 Jun 2009 03:28:25 GMT
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: enter registered state
debug: The provider changed state from: RegistrationState=Registering to: RegistrationState=Registered
debug: The provider changed state from: RegistrationState=Registering to: RegistrationState=Registered
debug: Dispatching Provider Status Change. Listeners=2 evt=ProviderPresenceStatusChangeEvent-[OldStatus=PresenceStatus:Offline, NewStatus=PresenceStatus:Online]
debug: status dispatching done.
debug: Dispatching stat. msg change. Listeners=2 evt=java.beans.PropertyChangeEvent[source=net.java.sip.communicator.impl.protocol.sip.ProtocolProviderServiceSipImpl@cb2185]
debug: status dispatching done.
debug: generated via headers:Via: SIP/2.0/UDP 192.168.2.95:5060

debug: generated contactHeader:Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>

debug: Dispatching a CallParticipant event to 0 listeners. event is: CallParticipantEvent: ID=1 source participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Unknown source call=Call: id=124598690248221078831 participants=1
debug: Dispatching a CallParticipantChangeEvent event to 1 listeners. event is: CallParticipantChangeEvent: type=CallParticipantStatusChange oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Unknown newV=net.java.sip.communicator.service.protocol.CallParticipantState:Initiating Call for participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Initiating Call
debug: Dispatching a CallEvent to 5 listeners. event is: CallEvent:[ id=1 Call=Call: id=124598690248221078831 participants=1]
outgoingCallCreated: null
Session.CallTracker: Call: id=124598690248221078831 participants=1 ready: false
created OutgoingCallTracker: [c:\db\sounds\phoneReminder.wav]
Session insureAudioFilesExists is not implemented
session. Call is connected, playing: [c:\db\sounds\phoneReminder.wav]
Session mediaService.playAudio not implemented
debug: registering format ilbc/rtp, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed with RTP manager
debug: registering format ALAW/rtp, 8000.0 Hz, 8-bit, Mono, Signed with RTP manager
debug: registering format speex/rtp, 8000.0 Hz, 8-bit, Mono, Signed with RTP manager
debug: registering format H264/RTP with RTP manager
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182, skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182, skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182, skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182, skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Will create media descs with: audio public address=/192.168.2.95:5000 and video public address=/192.168.2.95:5002
debug: We will be able to transmit in:
debug: unknown encoding format mpegaudio/rtp
debug: received response=
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bK0df49cdfbd46c87bf8ff5c79f2eb5ec6;received=206.81.217.93

From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74

To: <sip:14063661383@phone.wenco.com>;tag=as6567575c
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5,realm="phone.wenco.com",nonce="47aca73a"
Content-Length: 0

debug: Found 1 processor(s) for method INVITE
debug: Authenticating an INVITE request.
debug: trying to authenticate using : MD5, kim_sip, phone.wenco.com, true, 47aca73a, 00000001, xyz, INVITE, sip:14063661383@phone.wenco.com, v=0
o=kim_sip 0 0 IN IP4 192.168.2.95
s=-
c=IN IP4 192.168.2.95
t=0 0
m=audio 5000 RTP/AVP 0 8 97 3 5 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=zrtp-hash:1.10 1d486d9ff6653a351616ab66d6649d10a82f8adc55c82d12ec05674a7e36bc39
m=video 5002 RTP/AVP 99 34 26 31
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1
a=recvonly
a=zrtp-hash:1.10 873daca7f9d570514de22e9000f568944ab28b22eba583a6fb855cb382eb2ac3
, null
debug: Created authorization header: Proxy-Authorization: Digest username="kim_sip",realm="phone.wenco.com",nonce="47aca73a",uri="sip:14063661383@phone.wenco.com",response="60a31fd99e629e6415bd6fdd27f4a01d",algorithm=MD5

debug: Returning authorization transaction.
debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff;received=206.81.217.93

From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74

To: <sip:14063661383@phone.wenco.com>
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:14063661383@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method INVITE
debug: Dispatching a CallParticipantChangeEvent event to 3 listeners. event is: CallParticipantChangeEvent: type=CallParticipantStatusChange oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Initiating Call newV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting for participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Connecting
debug: received response=
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff;received=206.81.217.93

From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74

To: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:14063661383@209.237.251.182>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 2603 2603 IN IP4 209.237.251.182
s=session
c=IN IP4 209.237.251.182
t=0 0
m=audio 10066 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

debug: Found 1 processor(s) for method INVITE
debug: received the following JMF SendStreamEvent - javax.media.rtp.event.NewSendStreamEvent=javax.media.rtp.event.NewSendStreamEvent[source = RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
debug: Dispatching a CallParticipantChangeEvent event to 3 listeners. event is: CallParticipantChangeEvent: type=CallParticipantStatusChange oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting newV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting* for participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Connecting*
debug: Dispatching a CallChange event to 7 listeners. event is: CallChangeEvent: type=CallState oldV=net.java.sip.communicator.service.protocol.CallState:Initializing newV=net.java.sip.communicator.service.protocol.CallState:In Progress
Session.CallStateChange: net.java.sip.communicator.service.protocol.CallState:In Progress
Session. newvalue
debug: call connected. starting streaming
debug: Transaction terminated for req=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER

From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc

To: <sip:kim_sip@phone.wenco.com>
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: Transaction terminated for req=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 2 REGISTER

From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc

To: <sip:kim_sip@phone.wenco.com>
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKd01785901b2b5e8e6974fd2708407ef9
Authorization: Digest username="kim_sip",realm="phone.wenco.com",nonce="2c5e294e",uri="sip:phone.wenco.com",response="5e9d462a45fd6f9abbe705e4b09a7d0d",algorithm=MD5
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received a new incoming stream. javax.media.rtp.event.NewReceiveStreamEvent[source = RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
debug: Received a ControllerEvent: javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Unrealized,current=Realizing,target=Realized]
debug: Received a ControllerEvent: javax.media.RealizeCompleteEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Realizing,current=Realized,target=Realized]
debug: A player was realized and will be started.
debug: Received a ControllerEvent: javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Realized,current=Prefetching,target=Started]
debug: Received a ControllerEvent: javax.media.PrefetchCompleteEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Prefetching,current=Prefetched,target=Started]
debug: Received a ControllerEvent: javax.media.StartEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Prefetched,current=Started,target=Started,mediaTime=javax.media.Time@38c8c5,timeBaseTime=javax.media.Time@53b2c]
debug: Received a StartEvent
warn: container is null
debug: received request=
OPTIONS sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com SIP/2.0
Via: SIP/2.0/UDP 209.237.251.182:5060;branch=z9hG4bK1f60dfbe;rport=5060;received=209.237.251.182

From: "asterisk" <sip:asterisk@209.237.251.182>;tag=as5228afbf

To: <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>
Contact: <sip:asterisk@209.237.251.182>
Call-ID: 7e794f3d5fea2d1b5340ed4c43518400@209.237.251.182
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 26 Jun 2009 03:28:34 GMT
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Content-Length: 0

debug: Found 1 processor(s) for method OPTIONS
debug: Dialog terminated for req=gov.nist.javax.sip.stack.SIPDialog@e9493a
debug: received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff;received=206.81.217.93

From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74

To: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:14063661383@209.237.251.182>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 2603 2604 IN IP4 209.237.251.182
s=session
c=IN IP4 209.237.251.182
t=0 0
m=audio 10066 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

debug: Found 1 processor(s) for method INVITE
debug: Dispatching a CallParticipantChangeEvent event to 3 listeners. event is: CallParticipantChangeEvent: type=CallParticipantStatusChange oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting* newV=net.java.sip.communicator.service.protocol.CallParticipantState:Connected for participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Connected
debug: received a new incoming stream. javax.media.rtp.event.NewReceiveStreamEvent[source = RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
debug: Received a ControllerEvent: javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Unrealized,current=Realizing,target=Realized]
debug: Received a ControllerEvent: javax.media.RealizeCompleteEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Realizing,current=Realized,target=Realized]
debug: A player was realized and will be started.
debug: Received a ControllerEvent: javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Realized,current=Prefetching,target=Started]
debug: Received a ControllerEvent: javax.media.PrefetchCompleteEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Prefetching,current=Prefetched,target=Started]
debug: Received a ControllerEvent: javax.media.StartEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Prefetched,current=Started,target=Started,mediaTime=javax.media.Time@1540a77,timeBaseTime=javax.media.Time@7b2d29]
debug: Received a StartEvent
debug: Transaction terminated for req=
INVITE sip:14063661383@phone.wenco.com SIP/2.0
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE

From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74

To: <sip:14063661383@phone.wenco.com>
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>
Content-Type: application/sdp
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff
Proxy-Authorization: Digest username="kim_sip",realm="phone.wenco.com",nonce="47aca73a",uri="sip:14063661383@phone.wenco.com",response="60a31fd99e629e6415bd6fdd27f4a01d",algorithm=MD5
Content-Length: 432

v=0
o=kim_sip 0 0 IN IP4 192.168.2.95
s=-
c=IN IP4 192.168.2.95
t=0 0
m=audio 5000 RTP/AVP 0 8 97 3 5 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=zrtp-hash:1.10 1d486d9ff6653a351616ab66d6649d10a82f8adc55c82d12ec05674a7e36bc39
m=video 5002 RTP/AVP 99 34 26 31
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1
a=recvonly
a=zrtp-hash:1.10 873daca7f9d570514de22e9000f568944ab28b22eba583a6fb855cb382eb2ac3

debug: Found 1 processor(s) for method INVITE
debug: received request=
BYE sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com SIP/2.0
Via: SIP/2.0/UDP 209.237.251.182:5060;branch=z9hG4bK7c10da18;rport=5060;received=209.237.251.182

From: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe

To: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

debug: Found 1 processor(s) for method BYE
debug: We seem to already have a tag in this dialog. Returning
debug: sent response SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.237.251.182:5060;branch=z9hG4bK7c10da18;rport=5060;received=209.237.251.182

From: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe

To: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 102 BYE
User-Agent: Info-Tell Operator1.2.1Windows Vista
Content-Length: 0

debug: Dispatching a CallParticipantChangeEvent event to 3 listeners. event is: CallParticipantChangeEvent: type=CallParticipantStatusChange oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Connected newV=net.java.sip.communicator.service.protocol.CallParticipantState:Disconnected for participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Disconnected
debug: Dispatching a CallParticipant event to 7 listeners. event is: CallParticipantEvent: ID=2 source participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Disconnected source call=Call: id=124598690248221078831 participants=0
Session.callParticipantRemoved: CallParticipantEvent: ID=2 source participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Disconnected source call=Call: id=124598690248221078831 participants=0
debug: Dispatching a CallChange event to 7 listeners. event is: CallChangeEvent: type=CallState oldV=net.java.sip.communicator.service.protocol.CallState:In Progress newV=net.java.sip.communicator.service.protocol.CallState:Ended
debug: Dispatching a CallEvent to 5 listeners. event is: CallEvent:[ id=3 Call=Call: id=124598690248221078831 participants=0]
21:28:40.722 WARNING: impl.media.CallSessionImpl.callStateChanged().2534 Stopping streaming.
Session.callEnded
Users.setCallEnded
Session without an interaction
Mailbox callEnded
Session.CallStateChange: net.java.sip.communicator.service.protocol.CallState:Ended
warn: Stopping streaming.
debug: received the following JMF SendStreamEvent - javax.media.rtp.event.StreamClosedEvent=javax.media.rtp.event.StreamClosedEvent[source = RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
Info: Creating datasource for:dsound://
MediaContol initiProcessor: net.java.sip.communicator.impl.media.codec.EncodingConfiguration@1e492d8
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.alaw.JavaEncoder is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.alaw.DePacketizer is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.alaw.Packetizer is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.ulaw.Packetizer is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.video.h264.JNIEncoder is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.video.h264.Packetizer is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.video.h264.JNIDecoder is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.video.ImageScaler is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.speex.JavaEncoder is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.speex.JavaDecoder is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaEncoder is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaDecoder is already registered
debug: Received a ControllerEvent: javax.media.StopByRequestEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Started,current=Prefetched,target=Prefetched,mediaTime=javax.media.Time@10a4a32]
debug: Received a ControllerEvent: javax.media.DeallocateEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Prefetched,current=Realized,target=Realized,mediaTime=javax.media.Time@11f91ac]
debug: Received a ControllerEvent: javax.media.ControllerClosedEvent[source=com.sun.media.content.unknown.Handler@cf9bf0]
debug: Received a ControllerClosedEvent
debug: Received a ControllerEvent: javax.media.StopByRequestEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Started,current=Prefetched,target=Prefetched,mediaTime=javax.media.Time@101f287]
debug: Received a ControllerEvent: javax.media.DeallocateEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Prefetched,current=Realized,target=Realized,mediaTime=javax.media.Time@1578426]
debug: Received a ControllerEvent: javax.media.ControllerClosedEvent[source=com.sun.media.content.unknown.Handler@126456a]
debug: Received a ControllerClosedEvent
debug: generated via headers:Via: SIP/2.0/UDP 192.168.2.95:5060

debug: generated contactHeader:Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>

debug: sent request=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 3 REGISTER

From: <sip:kim_sip@phone.wenco.com>;tag=3887af51

To: <sip:kim_sip@phone.wenco.com>
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKde0facaff8dcb78368e5794fb057eee4
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Content-Length: 0

debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKde0facaff8dcb78368e5794fb057eee4;received=206.81.217.93

From: <sip:kim_sip@phone.wenco.com>;tag=3887af51

To: <sip:kim_sip@phone.wenco.com>
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKde0facaff8dcb78368e5794fb057eee4;received=206.81.217.93

From: <sip:kim_sip@phone.wenco.com>;tag=3887af51

To: <sip:kim_sip@phone.wenco.com>;tag=as15958e4b
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5,realm="phone.wenco.com",nonce="2997f0aa"
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: Authenticating a Register request.
debug: trying to authenticate using : MD5, kim_sip, phone.wenco.com, true, 2997f0aa, 00000001, xyz, REGISTER, sip:phone.wenco.com, , null
debug: Created authorization header: Authorization: Digest username="kim_sip",realm="phone.wenco.com",nonce="2997f0aa",uri="sip:phone.wenco.com",response="a2031b9798b755065bd1f42a8717a844",algorithm=MD5

debug: Returning authorization transaction.
debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bK02bab5577b988cb0ad4ee57808efbb74;received=206.81.217.93

From: <sip:kim_sip@phone.wenco.com>;tag=3887af51

To: <sip:kim_sip@phone.wenco.com>
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bK02bab5577b988cb0ad4ee57808efbb74;received=206.81.217.93

From: <sip:kim_sip@phone.wenco.com>;tag=3887af51

To: <sip:kim_sip@phone.wenco.com>;tag=as15958e4b
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Date: Fri, 26 Jun 2009 03:28:51 GMT
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER

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#6

Hello Emil
I am interested. Hope this is the time for me to get involved into
SIP-Communicator if needed.

Thanks
Rossi Kamal

···

On Thu, Jun 25, 2009 at 11:34 PM, Emil Ivov <emcho@sip-communicator.org>wrote:

Hey Martin,

Martin Han wrote:
> hey,
> I am interested in it and has read partial source code before, is
> there somebody can guide me to join it ?

I've already mentioned this offlist but still - glad to hear it. If you
(and/or anyone else for that matter) is interested in working on this
then you could start by using the existing code and trying to uncover
problems with it. You could do this on a protocol by protocol basis
starting with XMPP, then moving through IRC, MSN, ICQ/AIM, and Yahoo!
Messenger.

In addition to fixing the issues you could also have a look at the unit
tests (slicks). We already have some running for XMPP (courtesy of
Sympho) but they may need further work, and we'd also need to make them
run for the rest of the projects.

Good luck, and let us know if you have any questions
Cheers
Emil

> Really appreciated.
>
> 2009/6/24 Emil Ivov <emcho@sip-communicator.org
> <mailto:emcho@sip-communicator.org>>
>
> Hey all,
>
> We've been having multichatting about 80% implemented for a while
now.
> The problem is that despite the significant amount of work that we've
> already spent on it, the remaining 20% consist in a number of bugs
and
> misfeatures that make it feel quite shaky. We are probably going to
> remove it from our 1.0 branch for that reason.
>
> I was wondering if anyone on this list would like to work on
completing
> it. This would include resolving bugs (most trivial), writing unit
> tests, and if necessary adding essential features. According to my
> personal estimations the effort would amount to approximately 2 or 3
> person months.
>
> Don't be shy and speak up if you are interested!
>
> Cheers
> Emil
>
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> To unsubscribe, e-mail:
> dev-unsubscribe@sip-communicator.dev.java.net
> <mailto:dev-unsubscribe@sip-communicator.dev.java.net>
> For additional commands, e-mail:
> dev-help@sip-communicator.dev.java.net
> <mailto:dev-help@sip-communicator.dev.java.net>
>
>

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#7

Hey Kim,

Which packet are you using to run SIP Communicator? Could you please
send your complete log file as well as a wireshark capture from a
session with bad audio?

Thanks
Emil

kim Fairchild wrote:

···

Hello,

I have been able to make and receive calls with good quality for some
time but when I updated to new sources, I have bad outgoing connections.
The audio from Sip-Communicator is choppy and gets more and more laggey.
Looking at the debug output, it says that it is able to transmit in:

debug: We will be able to transmit in:
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp

Sorry, I am confused by this. Because of copyright issues, mpegaudio has
to be separately downloaded doesn't it? Would appreciate any help in
figuring this out. I include the debug output below.

BTW, is Stun disabled by default? How does SC do the NAT traversal
without it?

thanks, Kim

deploy-os-specific-bundles:
run:

Welcome to Felix.

debug: Service Impl:
net.java.sip.communicator.impl.configuration.ConfigurationActivator [
STARTED ]
debug: Using config file in $HOME/.sip-communicator:
C:\trunk.pure\sip-communicator.xml
debug: Service Impl:
net.java.sip.communicator.impl.configuration.ConfigurationActivator
[REGISTERED]
Info: Resource manager ... [REGISTERED]
Info: UI Service...[ STARTED ]
Info: UI Service ...[REGISTERED]
Info: Resource registered
net.java.sip.communicator.plugin.defaultresourcepack.DefaultColorPackImpl@1995d80
Info: Resource registered
net.java.sip.communicator.plugin.defaultresourcepack.DefaultImagePackImpl@6e293a
Info: Resource registered
net.java.sip.communicator.plugin.defaultresourcepack.DefaultLanguagePackImpl@1319c
Info: Resource registered
net.java.sip.communicator.plugin.defaultresourcepack.DefaultSettingsPackImpl@1479feb
Info: Resource registered
net.java.sip.communicator.plugin.defaultresourcepack.DefaultSoundPackImpl@1592174
Info: Default resources ... [REGISTERED]
debug: Started.
debug: SIP Protocol Provider Factory ... [REGISTERED]
debug: SIP Communicator Version: sip-communicator-1.2.1
debug: Started.
created a deviceConfiguration
MediaControl created
debug: Media Service ... [REGISTERED]
Starting mediaServiceimpl
deviceConfiguration.initialize
debug: Started.
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.icq
accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.icq
accounts
IOException in readRegistry: java.io.EOFException
debug: SIP Protocol Provider Factory ... [REGISTERED]
debug: Discovered 2 stored net.java.sip.communicator.impl.protocol.sip
accounts
debug: Loading account
net.java.sip.communicator.impl.protocol.sip.acc1244603766502
Info: Gibberish protocol implementation [STARTED].
Info: SSH protocol implementation [STARTED].
Info: Stun server address(null)/port(null) not set (or invalid).
Disabling STUN.
Info: Network Address Manager ...[ STARTED ]
Info: Network Address Manager Service ...[REGISTERED]
Info: Zeroconf protocol implementation [STARTED].
Info: IRC protocol implementation [STARTED].
Info: DICT protocol implementation [STARTED].
debug: Service Impl:
net.java.sip.communicator.impl.contactlist.ContactlistActivator [ STARTED ]
debug: Starting the meta contact list implementation.
debug: Service Impl:
net.java.sip.communicator.impl.contactlist.ContactlistActivator [REGISTERED]
Info: RSS protocol implementation [STARTED].
debug: User-Agent set to Info-Tell Operator/1.2.1
debug: New history service registered.
debug: Starting the msg history implementation.
Info: Message History Service ...[REGISTERED]
debug: New history service registered.
debug: Starting the call history implementation.
Info: Call History Service ...[REGISTERED]
Info: Audio Notifier Service...[ STARTED ]
Info: Audio Notifier Service ...[REGISTERED]
debug: Service Impl:
net.java.sip.communicator.impl.keybindings.KeybindingsActivator [ STARTED ]
Info: Notification Service...[ STARTED ]
Info: Notification Service ...[REGISTERED]
debug: Registering default event
IncomingMessage/PopupMessageAction/null/null
debug: Registering default event
IncomingMessage/SoundAction/resources/sounds/incomingMessage.wav/null
Info: Looking for Audio capturer
Info: DirectSound Capture Supported = true
debug: Registering default event IncomingCall/PopupMessageAction/null/null
debug: presence initialized with :true, true, 30, 3600 for null
debug: Registering default event
ProactiveNotification/PopupMessageAction/null/null
debug: Registering default event
SecurityMessage/PopupMessageAction/null/null
debug: Registering default event
CallSecurityOn/SoundAction/resources/sounds/zrtpSecure.wav/null
debug: Registering default event
CallSecurityError/SoundAction/resources/sounds/zrtpAlert.wav/null
Info: UI Service...[ STARTED ]
Info: UI Service ...[REGISTERED]
debug: Stored account for id SIP:kim_sip@phone.wenco.com for package
net.java.sip.communicator.impl.protocol.sip
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP:kim_sip@phone.wenco.com
debug: All contacts loaded for account SIP:kim_sip@phone.wenco.com
debug: Loading account
net.java.sip.communicator.impl.protocol.sip.acc1244609754970
debug: presence initialized with :true, true, 30, 3600 for null
debug: Stored account for id SIP:doha_amber@phone.wenco.com for package
net.java.sip.communicator.impl.protocol.sip
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP:doha_amber@phone.wenco.com
debug: All contacts loaded for account SIP:doha_amber@phone.wenco.com
debug: Discovered 0 stored
net.java.sip.communicator.impl.protocol.jabber accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.msn
accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.yahoo
accounts
debug: Discovered 0 stored
net.java.sip.communicator.impl.protocol.gibberish accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.ssh
accounts
debug: Discovered 0 stored
net.java.sip.communicator.impl.protocol.zeroconf accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.irc
accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.dict
accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.rss
accounts
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event:
net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.branding.AboutWindowPluginComponent@a6e0a9]
Info: ABOUT WINDOW ... [REGISTERED]
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event:
net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.branding.AboutWindowPluginComponent@f81402]
Info: CHAT ABOUT WINDOW ... [REGISTERED]
Info: Swing Notification ...[ STARTING ]
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event:
net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.updatechecker.UpdateCheckActivator$UpdateMenuButtonComponent@154ea79]
Info: No config file specified for update checker. Disabling update checks
debug: Updates are disabled. Faking latest version.
Info: Swing Notification ...[REGISTERED]
Info: added the following popup handler : SIP Communicator popups
21:28:14.218 INFO: doha.impl.phone.PhoneActivator.start().59 Phone
Service ...[Starting]
21:28:14.218 FINEST: doha.impl.phone.PhoneActivator.start().62 [entry] start
Info: Systray Service...[ STARTED ]
Info: Systray Service ...[REGISTERED]
PhoneActivator Start!
Info: Phone Service ...[Starting]
OperatorServiceImpl started!
Session created
startListener
Session.startListener turned on!
Session.startListener turned on!
users.getCallerIDDocument
loadDom: c:\db\admin\callerids.xml
Opening connection to: jdbc:mysql://ljade.com:3306/mydatabase
1 Database.java:412 open()
2 Database.java:763 getXML()
3 Database.java:750 getXMLFile()
4 DomServiceImpl.java:546 loadDom()
5 Users.java:241 getCallerIDDocument()
6 Users.java:37 <init>()
7 OperatorServiceImpl.java:108 <init>()
8 PhoneActivator.java:64 start()
9 SecureAction.java:589 startActivator()
Opening SQL Database connection...
debug: The provider changed state from: RegistrationState=Unregistered
to: RegistrationState=Registering
debug: The provider changed state from: RegistrationState=Unregistered
to: RegistrationState=Registering
debug: generated max forwards: Max-Forwards: 70

debug: generated via headers:Via: SIP/2.0/UDP 192.168.2.95:5060

debug: generated contactHeader:Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>

Info: DirectSoundAuto: Committed ok
Info: JavaSound Capture Supported = true
Info: JavaSoundAuto: Committed ok
debug: No FMJ javasound detected:
net.sf.fmj.media.cdp.javasound.CaptureDevicePlugger
Info: Looking for video capture devices
Info: Scanning for configured Audio Devices.
debug: Found 2 capture devices: [Ljavax.media.CaptureDeviceInfo;@8d41f2
Info: Found DirectSoundCapture as an audio capture device.
Info: Scanning for configured Video Devices.
Info: No Video Device was found.
defaultMediaControl.initialize
MediaControl initialized:
net.java.sip.communicator.impl.media.codec.EncodingConfiguration@1e492d8
debug: Adding package : net.java.sip.communicator.impl
debug: Adding package : net.sf.fmj
debug: Registering new protocol prefix list : [javax, com.sun, com.ibm,
net.java.sip.communicator.impl, net.sf.fmj]
Info: Creating datasource for:dsound://
MediaContol initiProcessor:
net.java.sip.communicator.impl.media.codec.EncodingConfiguration@1e492d8
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.alaw.JavaEncoder is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.alaw.DePacketizer is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.alaw.Packetizer is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.ulaw.Packetizer is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.video.h264.JNIEncoder is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.video.h264.Packetizer is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.video.h264.JNIDecoder is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.video.ImageScaler is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.speex.JavaEncoder is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.speex.JavaDecoder is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaEncoder is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaDecoder is
successfully registered
connection: com.mysql.jdbc.Connection@149249e
... SQL Database connected in 0.863 seconds
query: select * from Callerids where ider = 'admin_callerids_0'
Local file is up-to-date localMark: 232 serverMark: 232
Cache.initializing...
Initial Interaction created - used for plugin
Info: trying database in phoneactivator
Info: Phone Service ...[REGISTERED]
Phone Opened Connection
21:28:15.184 INFO: doha.impl.phone.PhoneActivator.start().95 trying
database in phoneactivator
21:28:15.185 INFO: doha.impl.phone.PhoneActivator.start().104 Phone
Service ...[REGISTERED]
21:28:15.185 FINEST: doha.impl.phone.PhoneActivator.start().109 [exit] start
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event:
net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.whiteboard.WhiteboardMenuItem@58d7c2]
Info: WHITEBOARD... [REGISTERED]
Info: Handling registration of a new Account Wizard.
Info: Handling registration of a new Account Wizard.
Info: Handling registration of a new Account Wizard.
Info: Handling registration of a new Account Wizard.
Info: Handling registration of a new Account Wizard.
Info: Handling registration of a new Account Wizard.
Info: Handling registration of a new Account Wizard.
Info: Loading gibberish account wizard.
Info: Handling registration of a new Account Wizard.
Info: Gibberish account registration wizard [STARTED].
Info: Loading ssh account wizard.
Info: Handling registration of a new Account Wizard.
Info: SSH account registration wizard [STARTED].
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event:
net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.callhistoryform.ExtendedCallHistorySearchItem@959fa1]
Info: EXTENDED CALL HISTORY SEARCH... [REGISTERED]
Info: Loading rss account wizard.
Info: Handling registration of a new Account Wizard.
Info: RSS account registration wizard [STARTED].
Info: Loading zeroconf account wizard.
Info: Handling registration of a new Account Wizard.
Info: Zeroconf account registration wizard [STARTED].
Info: Loading irc account wizard.
Info: Handling registration of a new Account Wizard.
Info: IRC account registration wizard [STARTED].
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event:
net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.contactinfo.ContactInfoMenuItem@423606]
Info: CONTACT INFO... [REGISTERED]
debug: Found 2 already installed providers.
debug: Adding protocol provider SIP
debug: Adding protocol provider SIP
debug: Service Impl:
net.java.sip.communicator.plugin.keybindingchooser.KeybindingChooserActivator
[ STARTED ]
Info: PREFERENCES PLUGIN... [REGISTERED]
Info: Handling registration of a new Account Wizard.
Started the CMSActivator
Info: EXTENDED CALL HISTORY SEARCH... [REGISTERED]
Info: SIMPLE ACCOUNT REGISTRATION ...[STARTED]
debug: Starting the ShutdownTimeout service.
Info: Mailbox plug-in...[STARTING]
21:28:15.370 INFO: org.osgi.framework.BundleActivator.start().75 Mailbox
plug-in...[STARTING]
debug: Starting the mailbox implementation.
debug: Found 2 already installed providers.
debug: Adding protocol provider SIP
debug: Adding protocol provider SIP
warn: Missing resource for key: outgoing
21:28:15.374 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: outgoing
21:28:15.795 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: browse
21:28:15.796 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: incoming
warn: Missing resource for key: browse
warn: Missing resource for key: incoming
21:28:15.942 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: browse
warn: Missing resource for key: browse
warn: Missing resource for key: waitTime
21:28:15.944 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: waitTime
warn: Missing resource for key: maxMessageTime
21:28:15.953 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: maxMessageTime
warn: Missing resource for key: confirm
21:28:15.955 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: confirm
warn: Missing resource for key: default
21:28:15.955 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: default
fileLocation: null
21:28:15.959 INFO: org.osgi.framework.BundleActivator.start().89 Mailbox
plug-in...[STARTED]
Info: Mailbox plug-in...[STARTED]
debug: sent request=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Content-Length: 0

debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>;tag=as61098680
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
WWW-Authenticate: Digest
algorithm=MD5,realm="phone.wenco.com",nonce="2c5e294e"
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: Authenticating a Register request.
debug: trying to authenticate using : MD5, kim_sip, phone.wenco.com,
true, 2c5e294e, 00000001, xyz, REGISTER, sip:phone.wenco.com, , null
debug: Created authorization header: Authorization: Digest
username="kim_sip",realm="phone.wenco.com",nonce="2c5e294e",uri="sip:phone.wenco.com",response="5e9d462a45fd6f9abbe705e4b09a7d0d",algorithm=MD5

debug: Returning authorization transaction.
debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKd01785901b2b5e8e6974fd2708407ef9;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKd01785901b2b5e8e6974fd2708407ef9;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>;tag=as61098680
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Expires: 3600
Contact:
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Date: Fri, 26 Jun 2009 03:28:25 GMT
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: enter registered state
debug: The provider changed state from: RegistrationState=Registering
to: RegistrationState=Registered
debug: The provider changed state from: RegistrationState=Registering
to: RegistrationState=Registered
debug: Dispatching Provider Status Change. Listeners=2
evt=ProviderPresenceStatusChangeEvent-[OldStatus=PresenceStatus:Offline,
NewStatus=PresenceStatus:Online]
debug: status dispatching done.
debug: Dispatching stat. msg change. Listeners=2
evt=java.beans.PropertyChangeEvent[source=net.java.sip.communicator.impl.protocol.sip.ProtocolProviderServiceSipImpl@cb2185]
debug: status dispatching done.
debug: generated via headers:Via: SIP/2.0/UDP 192.168.2.95:5060

debug: generated contactHeader:Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>

debug: Dispatching a CallParticipant event to 0 listeners. event is:
CallParticipantEvent: ID=1 source
participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Unknown source call=Call:
id=124598690248221078831 participants=1
debug: Dispatching a CallParticipantChangeEvent event to 1 listeners.
event is: CallParticipantChangeEvent: type=CallParticipantStatusChange
oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Unknown
newV=net.java.sip.communicator.service.protocol.CallParticipantState:Initiating
Call for participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Initiating Call
debug: Dispatching a CallEvent to 5 listeners. event is: CallEvent:[
id=1 Call=Call: id=124598690248221078831 participants=1]
outgoingCallCreated: null
Session.CallTracker: Call: id=124598690248221078831 participants=1
ready: false
created OutgoingCallTracker: [c:\db\sounds\phoneReminder.wav]
Session insureAudioFilesExists is not implemented
session. Call is connected, playing: [c:\db\sounds\phoneReminder.wav]
Session mediaService.playAudio not implemented
debug: registering format ilbc/rtp, 8000.0 Hz, 16-bit, Mono,
LittleEndian, Signed with RTP manager
debug: registering format ALAW/rtp, 8000.0 Hz, 8-bit, Mono, Signed with
RTP manager
debug: registering format speex/rtp, 8000.0 Hz, 8-bit, Mono, Signed with
RTP manager
debug: registering format H264/RTP with RTP manager
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182,
skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182,
skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182,
skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182,
skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Will create media descs with: audio public
address=/192.168.2.95:5000 and video public address=/192.168.2.95:5002
debug: We will be able to transmit in:
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: received response=
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bK0df49cdfbd46c87bf8ff5c79f2eb5ec6;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>;tag=as6567575c
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Proxy-Authenticate: Digest
algorithm=MD5,realm="phone.wenco.com",nonce="47aca73a"
Content-Length: 0

debug: Found 1 processor(s) for method INVITE
debug: Authenticating an INVITE request.
debug: trying to authenticate using : MD5, kim_sip, phone.wenco.com,
true, 47aca73a, 00000001, xyz, INVITE, sip:14063661383@phone.wenco.com, v=0
o=kim_sip 0 0 IN IP4 192.168.2.95
s=-
c=IN IP4 192.168.2.95
t=0 0
m=audio 5000 RTP/AVP 0 8 97 3 5 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=zrtp-hash:1.10
1d486d9ff6653a351616ab66d6649d10a82f8adc55c82d12ec05674a7e36bc39
m=video 5002 RTP/AVP 99 34 26 31
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1
a=recvonly
a=zrtp-hash:1.10
873daca7f9d570514de22e9000f568944ab28b22eba583a6fb855cb382eb2ac3
, null
debug: Created authorization header: Proxy-Authorization: Digest
username="kim_sip",realm="phone.wenco.com",nonce="47aca73a",uri="sip:14063661383@phone.wenco.com",response="60a31fd99e629e6415bd6fdd27f4a01d",algorithm=MD5

debug: Returning authorization transaction.
debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:14063661383@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method INVITE
debug: Dispatching a CallParticipantChangeEvent event to 3 listeners.
event is: CallParticipantChangeEvent: type=CallParticipantStatusChange
oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Initiating
Call
newV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting
for participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Connecting
debug: received response=
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:14063661383@209.237.251.182>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 2603 2603 IN IP4 209.237.251.182
s=session
c=IN IP4 209.237.251.182
t=0 0
m=audio 10066 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

debug: Found 1 processor(s) for method INVITE
debug: received the following JMF SendStreamEvent -
javax.media.rtp.event.NewSendStreamEvent=javax.media.rtp.event.NewSendStreamEvent[source
= RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address
        RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
debug: Dispatching a CallParticipantChangeEvent event to 3 listeners.
event is: CallParticipantChangeEvent: type=CallParticipantStatusChange
oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting
newV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting*
for participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Connecting*
debug: Dispatching a CallChange event to 7 listeners. event is:
CallChangeEvent: type=CallState
oldV=net.java.sip.communicator.service.protocol.CallState:Initializing
newV=net.java.sip.communicator.service.protocol.CallState:In Progress
Session.CallStateChange:
net.java.sip.communicator.service.protocol.CallState:In Progress
Session. newvalue
debug: call connected. starting streaming
debug: Transaction terminated for req=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: Transaction terminated for req=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 2 REGISTER
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKd01785901b2b5e8e6974fd2708407ef9
Authorization: Digest
username="kim_sip",realm="phone.wenco.com",nonce="2c5e294e",uri="sip:phone.wenco.com",response="5e9d462a45fd6f9abbe705e4b09a7d0d",algorithm=MD5
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received a new incoming stream.
javax.media.rtp.event.NewReceiveStreamEvent[source = RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address
        RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
debug: Received a ControllerEvent:
javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Unrealized,current=Realizing,target=Realized]
debug: Received a ControllerEvent:
javax.media.RealizeCompleteEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Realizing,current=Realized,target=Realized]
debug: A player was realized and will be started.
debug: Received a ControllerEvent:
javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Realized,current=Prefetching,target=Started]
debug: Received a ControllerEvent:
javax.media.PrefetchCompleteEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Prefetching,current=Prefetched,target=Started]
debug: Received a ControllerEvent:
javax.media.StartEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Prefetched,current=Started,target=Started,mediaTime=javax.media.Time@38c8c5,timeBaseTime=javax.media.Time@53b2c]
debug: Received a StartEvent
warn: container is null
debug: received request=
OPTIONS
sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com
SIP/2.0
Via: SIP/2.0/UDP
209.237.251.182:5060;branch=z9hG4bK1f60dfbe;rport=5060;received=209.237.251.182
From: "asterisk" <sip:asterisk@209.237.251.182>;tag=as5228afbf
To:
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>
Contact: <sip:asterisk@209.237.251.182>
Call-ID: 7e794f3d5fea2d1b5340ed4c43518400@209.237.251.182
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 26 Jun 2009 03:28:34 GMT
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Content-Length: 0

debug: Found 1 processor(s) for method OPTIONS
debug: Dialog terminated for req=gov.nist.javax.sip.stack.SIPDialog@e9493a
debug: received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:14063661383@209.237.251.182>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 2603 2604 IN IP4 209.237.251.182
s=session
c=IN IP4 209.237.251.182
t=0 0
m=audio 10066 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

debug: Found 1 processor(s) for method INVITE
debug: Dispatching a CallParticipantChangeEvent event to 3 listeners.
event is: CallParticipantChangeEvent: type=CallParticipantStatusChange
oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting*
newV=net.java.sip.communicator.service.protocol.CallParticipantState:Connected
for participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Connected
debug: received a new incoming stream.
javax.media.rtp.event.NewReceiveStreamEvent[source = RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address
        RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
debug: Received a ControllerEvent:
javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Unrealized,current=Realizing,target=Realized]
debug: Received a ControllerEvent:
javax.media.RealizeCompleteEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Realizing,current=Realized,target=Realized]
debug: A player was realized and will be started.
debug: Received a ControllerEvent:
javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Realized,current=Prefetching,target=Started]
debug: Received a ControllerEvent:
javax.media.PrefetchCompleteEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Prefetching,current=Prefetched,target=Started]
debug: Received a ControllerEvent:
javax.media.StartEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Prefetched,current=Started,target=Started,mediaTime=javax.media.Time@1540a77,timeBaseTime=javax.media.Time@7b2d29]
debug: Received a StartEvent
debug: Transaction terminated for req=
INVITE sip:14063661383@phone.wenco.com SIP/2.0
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>
Content-Type: application/sdp
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff
Proxy-Authorization: Digest
username="kim_sip",realm="phone.wenco.com",nonce="47aca73a",uri="sip:14063661383@phone.wenco.com",response="60a31fd99e629e6415bd6fdd27f4a01d",algorithm=MD5
Content-Length: 432

v=0
o=kim_sip 0 0 IN IP4 192.168.2.95
s=-
c=IN IP4 192.168.2.95
t=0 0
m=audio 5000 RTP/AVP 0 8 97 3 5 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=zrtp-hash:1.10
1d486d9ff6653a351616ab66d6649d10a82f8adc55c82d12ec05674a7e36bc39
m=video 5002 RTP/AVP 99 34 26 31
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1
a=recvonly
a=zrtp-hash:1.10
873daca7f9d570514de22e9000f568944ab28b22eba583a6fb855cb382eb2ac3

debug: Found 1 processor(s) for method INVITE
debug: received request=
BYE
sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com
SIP/2.0
Via: SIP/2.0/UDP
209.237.251.182:5060;branch=z9hG4bK7c10da18;rport=5060;received=209.237.251.182
From: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe
To: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

debug: Found 1 processor(s) for method BYE
debug: We seem to already have a tag in this dialog. Returning
debug: sent response SIP/2.0 200 OK
Via: SIP/2.0/UDP
209.237.251.182:5060;branch=z9hG4bK7c10da18;rport=5060;received=209.237.251.182
From: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe
To: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 102 BYE
User-Agent: Info-Tell Operator1.2.1Windows Vista
Content-Length: 0

debug: Dispatching a CallParticipantChangeEvent event to 3 listeners.
event is: CallParticipantChangeEvent: type=CallParticipantStatusChange
oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Connected
newV=net.java.sip.communicator.service.protocol.CallParticipantState:Disconnected
for participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Disconnected
debug: Dispatching a CallParticipant event to 7 listeners. event is:
CallParticipantEvent: ID=2 source
participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Disconnected source call=Call:
id=124598690248221078831 participants=0
Session.callParticipantRemoved: CallParticipantEvent: ID=2 source
participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Disconnected source call=Call:
id=124598690248221078831 participants=0
debug: Dispatching a CallChange event to 7 listeners. event is:
CallChangeEvent: type=CallState
oldV=net.java.sip.communicator.service.protocol.CallState:In Progress
newV=net.java.sip.communicator.service.protocol.CallState:Ended
debug: Dispatching a CallEvent to 5 listeners. event is: CallEvent:[
id=3 Call=Call: id=124598690248221078831 participants=0]
21:28:40.722 WARNING: impl.media.CallSessionImpl.callStateChanged().2534
Stopping streaming.
Session.callEnded
Users.setCallEnded
Session without an interaction
Mailbox callEnded
Session.CallStateChange:
net.java.sip.communicator.service.protocol.CallState:Ended
warn: Stopping streaming.
debug: received the following JMF SendStreamEvent -
javax.media.rtp.event.StreamClosedEvent=javax.media.rtp.event.StreamClosedEvent[source
= RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address
        RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
Info: Creating datasource for:dsound://
MediaContol initiProcessor:
net.java.sip.communicator.impl.media.codec.EncodingConfiguration@1e492d8
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.alaw.JavaEncoder is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.alaw.DePacketizer is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.alaw.Packetizer is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.ulaw.Packetizer is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.video.h264.JNIEncoder is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.video.h264.Packetizer is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.video.h264.JNIDecoder is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.video.ImageScaler is already
registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.speex.JavaEncoder is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.speex.JavaDecoder is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaEncoder is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaDecoder is
already registered
debug: Received a ControllerEvent:
javax.media.StopByRequestEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Started,current=Prefetched,target=Prefetched,mediaTime=javax.media.Time@10a4a32]
debug: Received a ControllerEvent:
javax.media.DeallocateEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Prefetched,current=Realized,target=Realized,mediaTime=javax.media.Time@11f91ac]
debug: Received a ControllerEvent:
javax.media.ControllerClosedEvent[source=com.sun.media.content.unknown.Handler@cf9bf0]
debug: Received a ControllerClosedEvent
debug: Received a ControllerEvent:
javax.media.StopByRequestEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Started,current=Prefetched,target=Prefetched,mediaTime=javax.media.Time@101f287]
debug: Received a ControllerEvent:
javax.media.DeallocateEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Prefetched,current=Realized,target=Realized,mediaTime=javax.media.Time@1578426]
debug: Received a ControllerEvent:
javax.media.ControllerClosedEvent[source=com.sun.media.content.unknown.Handler@126456a]
debug: Received a ControllerClosedEvent
debug: generated via headers:Via: SIP/2.0/UDP 192.168.2.95:5060

debug: generated contactHeader:Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>

debug: sent request=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 3 REGISTER
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKde0facaff8dcb78368e5794fb057eee4
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Content-Length: 0

debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKde0facaff8dcb78368e5794fb057eee4;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKde0facaff8dcb78368e5794fb057eee4;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>;tag=as15958e4b
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
WWW-Authenticate: Digest
algorithm=MD5,realm="phone.wenco.com",nonce="2997f0aa"
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: Authenticating a Register request.
debug: trying to authenticate using : MD5, kim_sip, phone.wenco.com,
true, 2997f0aa, 00000001, xyz, REGISTER, sip:phone.wenco.com, , null
debug: Created authorization header: Authorization: Digest
username="kim_sip",realm="phone.wenco.com",nonce="2997f0aa",uri="sip:phone.wenco.com",response="a2031b9798b755065bd1f42a8717a844",algorithm=MD5

debug: Returning authorization transaction.
debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bK02bab5577b988cb0ad4ee57808efbb74;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bK02bab5577b988cb0ad4ee57808efbb74;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>;tag=as15958e4b
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Expires: 3600
Contact:
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Date: Fri, 26 Jun 2009 03:28:51 GMT
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER

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--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#8

Emil Ivov wrote:

Thanks for responding so quickly.

I am sorry, I don't know what you mean by "which packet are you using..". Package? I am running with current code from about 2 weeks ago.

I am including the wireshark capture and the output from the console of Netbeans (I modified the logger to output there).

I started wireshark and then SC. I placed one outgoing call. And then I quit.

log.txt (50 KB)

wireshark.pcap (425 KB)

···

Hey Kim,

Which packet are you using to run SIP Communicator? Could you please
send your complete log file as well as a wireshark capture from a
session with bad audio?

Thanks
Emil

kim Fairchild wrote:
  

Hello,

I have been able to make and receive calls with good quality for some time but when I updated to new sources, I have bad outgoing connections. The audio from Sip-Communicator is choppy and gets more and more laggey. Looking at the debug output, it says that it is able to transmit in:

debug: We will be able to transmit in:
debug: unknown encoding format mpegaudio/rtp

Sorry, I am confused by this. Because of copyright issues, mpegaudio has to be separately downloaded doesn't it? Would appreciate any help in figuring this out. I include the debug output below.

BTW, is Stun disabled by default? How does SC do the NAT traversal without it?

thanks, Kim

deploy-os-specific-bundles:
run:

Welcome to Felix.

debug: Service Impl: net.java.sip.communicator.impl.configuration.ConfigurationActivator [ STARTED ]
debug: Using config file in $HOME/.sip-communicator: C:\trunk.pure\sip-communicator.xml
debug: Service Impl: net.java.sip.communicator.impl.configuration.ConfigurationActivator [REGISTERED]
Info: Resource manager ... [REGISTERED]
Info: UI Service...[ STARTED ]
Info: UI Service ...[REGISTERED]
Info: Resource registered net.java.sip.communicator.plugin.defaultresourcepack.DefaultColorPackImpl@1995d80
Info: Resource registered net.java.sip.communicator.plugin.defaultresourcepack.DefaultImagePackImpl@6e293a
Info: Resource registered net.java.sip.communicator.plugin.defaultresourcepack.DefaultLanguagePackImpl@1319c
Info: Resource registered net.java.sip.communicator.plugin.defaultresourcepack.DefaultSettingsPackImpl@1479feb
Info: Resource registered net.java.sip.communicator.plugin.defaultresourcepack.DefaultSoundPackImpl@1592174
Info: Default resources ... [REGISTERED]
debug: Started.
debug: SIP Protocol Provider Factory ... [REGISTERED]
debug: SIP Communicator Version: sip-communicator-1.2.1
debug: Started.
created a deviceConfiguration
MediaControl created
debug: Media Service ... [REGISTERED]
Starting mediaServiceimpl
deviceConfiguration.initialize
debug: Started.
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.icq accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.icq accounts
IOException in readRegistry: java.io.EOFException
debug: SIP Protocol Provider Factory ... [REGISTERED]
debug: Discovered 2 stored net.java.sip.communicator.impl.protocol.sip accounts
debug: Loading account net.java.sip.communicator.impl.protocol.sip.acc1244603766502
Info: Gibberish protocol implementation [STARTED].
Info: SSH protocol implementation [STARTED].
Info: Stun server address(null)/port(null) not set (or invalid). Disabling STUN.
Info: Network Address Manager ...[ STARTED ]
Info: Network Address Manager Service ...[REGISTERED]
Info: Zeroconf protocol implementation [STARTED].
Info: IRC protocol implementation [STARTED].
Info: DICT protocol implementation [STARTED].
debug: Service Impl: net.java.sip.communicator.impl.contactlist.ContactlistActivator [ STARTED ]
debug: Starting the meta contact list implementation.
debug: Service Impl: net.java.sip.communicator.impl.contactlist.ContactlistActivator [REGISTERED]
Info: RSS protocol implementation [STARTED].
debug: User-Agent set to Info-Tell Operator/1.2.1
debug: New history service registered.
debug: Starting the msg history implementation.
Info: Message History Service ...[REGISTERED]
debug: New history service registered.
debug: Starting the call history implementation.
Info: Call History Service ...[REGISTERED]
Info: Audio Notifier Service...[ STARTED ]
Info: Audio Notifier Service ...[REGISTERED]
debug: Service Impl: net.java.sip.communicator.impl.keybindings.KeybindingsActivator [ STARTED ]
Info: Notification Service...[ STARTED ]
Info: Notification Service ...[REGISTERED]
debug: Registering default event IncomingMessage/PopupMessageAction/null/null
debug: Registering default event IncomingMessage/SoundAction/resources/sounds/incomingMessage.wav/null
Info: Looking for Audio capturer
Info: DirectSound Capture Supported = true
debug: Registering default event IncomingCall/PopupMessageAction/null/null
debug: presence initialized with :true, true, 30, 3600 for null
debug: Registering default event ProactiveNotification/PopupMessageAction/null/null
debug: Registering default event SecurityMessage/PopupMessageAction/null/null
debug: Registering default event CallSecurityOn/SoundAction/resources/sounds/zrtpSecure.wav/null
debug: Registering default event CallSecurityError/SoundAction/resources/sounds/zrtpAlert.wav/null
Info: UI Service...[ STARTED ]
Info: UI Service ...[REGISTERED]
debug: Stored account for id SIP:kim_sip@phone.wenco.com for package net.java.sip.communicator.impl.protocol.sip
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP:kim_sip@phone.wenco.com
debug: All contacts loaded for account SIP:kim_sip@phone.wenco.com
debug: Loading account net.java.sip.communicator.impl.protocol.sip.acc1244609754970
debug: presence initialized with :true, true, 30, 3600 for null
debug: Stored account for id SIP:doha_amber@phone.wenco.com for package net.java.sip.communicator.impl.protocol.sip
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP:doha_amber@phone.wenco.com
debug: All contacts loaded for account SIP:doha_amber@phone.wenco.com
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.jabber accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.msn accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.yahoo accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.gibberish accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.ssh accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.zeroconf accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.irc accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.dict accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.rss accounts
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event: net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.branding.AboutWindowPluginComponent@a6e0a9]
Info: ABOUT WINDOW ... [REGISTERED]
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event: net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.branding.AboutWindowPluginComponent@f81402]
Info: CHAT ABOUT WINDOW ... [REGISTERED]
Info: Swing Notification ...[ STARTING ]
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event: net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.updatechecker.UpdateCheckActivator$UpdateMenuButtonComponent@154ea79]
Info: No config file specified for update checker. Disabling update checks
debug: Updates are disabled. Faking latest version.
Info: Swing Notification ...[REGISTERED]
Info: added the following popup handler : SIP Communicator popups
21:28:14.218 INFO: doha.impl.phone.PhoneActivator.start().59 Phone Service ...[Starting]
21:28:14.218 FINEST: doha.impl.phone.PhoneActivator.start().62 [entry] start
Info: Systray Service...[ STARTED ]
Info: Systray Service ...[REGISTERED]
PhoneActivator Start!
Info: Phone Service ...[Starting]
OperatorServiceImpl started!
Session created
startListener
Session.startListener turned on!
users.getCallerIDDocument
loadDom: c:\db\admin\callerids.xml
Opening connection to: jdbc:mysql://ljade.com:3306/mydatabase
1 Database.java:412 open()
2 Database.java:763 getXML()
3 Database.java:750 getXMLFile()
4 DomServiceImpl.java:546 loadDom()
5 Users.java:241 getCallerIDDocument()
6 Users.java:37 <init>()
7 OperatorServiceImpl.java:108 <init>()
8 PhoneActivator.java:64 start()
9 SecureAction.java:589 startActivator()
Opening SQL Database connection...
debug: The provider changed state from: RegistrationState=Unregistered to: RegistrationState=Registering
debug: The provider changed state from: RegistrationState=Unregistered to: RegistrationState=Registering
debug: generated max forwards: Max-Forwards: 70

debug: generated via headers:Via: SIP/2.0/UDP 192.168.2.95:5060

debug: generated contactHeader:Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>

Info: DirectSoundAuto: Committed ok
Info: JavaSound Capture Supported = true
Info: JavaSoundAuto: Committed ok
debug: No FMJ javasound detected: net.sf.fmj.media.cdp.javasound.CaptureDevicePlugger
Info: Looking for video capture devices
Info: Scanning for configured Audio Devices.
debug: Found 2 capture devices: [Ljavax.media.CaptureDeviceInfo;@8d41f2
Info: Found DirectSoundCapture as an audio capture device.
Info: Scanning for configured Video Devices.
Info: No Video Device was found.
defaultMediaControl.initialize
MediaControl initialized: net.java.sip.communicator.impl.media.codec.EncodingConfiguration@1e492d8
debug: Adding package : net.java.sip.communicator.impl
debug: Adding package : net.sf.fmj
debug: Registering new protocol prefix list : [javax, com.sun, com.ibm, net.java.sip.communicator.impl, net.sf.fmj]
Info: Creating datasource for:dsound://
MediaContol initiProcessor: net.java.sip.communicator.impl.media.codec.EncodingConfiguration@1e492d8
debug: Codec net.java.sip.communicator.impl.media.codec.audio.alaw.JavaEncoder is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.audio.alaw.DePacketizer is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.audio.alaw.Packetizer is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.audio.ulaw.Packetizer is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.video.h264.JNIEncoder is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.video.h264.Packetizer is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.video.h264.JNIDecoder is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.video.ImageScaler is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.audio.speex.JavaEncoder is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.audio.speex.JavaDecoder is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaEncoder is successfully registered
debug: Codec net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaDecoder is successfully registered
connection: com.mysql.jdbc.Connection@149249e
... SQL Database connected in 0.863 seconds
query: select * from Callerids where ider = 'admin_callerids_0'
Local file is up-to-date localMark: 232 serverMark: 232
Cache.initializing...
Initial Interaction created - used for plugin
Info: trying database in phoneactivator
Info: Phone Service ...[REGISTERED]
Phone Opened Connection
21:28:15.184 INFO: doha.impl.phone.PhoneActivator.start().95 trying database in phoneactivator
21:28:15.185 INFO: doha.impl.phone.PhoneActivator.start().104 Phone Service ...[REGISTERED]
21:28:15.185 FINEST: doha.impl.phone.PhoneActivator.start().109 [exit] start
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event: net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.whiteboard.WhiteboardMenuItem@58d7c2]
Info: WHITEBOARD... [REGISTERED]
Info: Handling registration of a new Account Wizard.
Info: Loading gibberish account wizard.
Info: Handling registration of a new Account Wizard.
Info: Gibberish account registration wizard [STARTED].
Info: Loading ssh account wizard.
Info: Handling registration of a new Account Wizard.
Info: SSH account registration wizard [STARTED].
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event: net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.callhistoryform.ExtendedCallHistorySearchItem@959fa1]
Info: EXTENDED CALL HISTORY SEARCH... [REGISTERED]
Info: Loading rss account wizard.
Info: Handling registration of a new Account Wizard.
Info: RSS account registration wizard [STARTED].
Info: Loading zeroconf account wizard.
Info: Handling registration of a new Account Wizard.
Info: Zeroconf account registration wizard [STARTED].
Info: Loading irc account wizard.
Info: Handling registration of a new Account Wizard.
Info: IRC account registration wizard [STARTED].
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event: net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.contactinfo.ContactInfoMenuItem@423606]
Info: CONTACT INFO... [REGISTERED]
debug: Found 2 already installed providers.
debug: Adding protocol provider SIP
debug: Service Impl: net.java.sip.communicator.plugin.keybindingchooser.KeybindingChooserActivator [ STARTED ]
Info: PREFERENCES PLUGIN... [REGISTERED]
Info: Handling registration of a new Account Wizard.
Started the CMSActivator
Info: EXTENDED CALL HISTORY SEARCH... [REGISTERED]
Info: SIMPLE ACCOUNT REGISTRATION ...[STARTED]
debug: Starting the ShutdownTimeout service.
Info: Mailbox plug-in...[STARTING]
21:28:15.370 INFO: org.osgi.framework.BundleActivator.start().75 Mailbox plug-in...[STARTING]
debug: Starting the mailbox implementation.
debug: Found 2 already installed providers.
debug: Adding protocol provider SIP
warn: Missing resource for key: outgoing
21:28:15.374 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: outgoing
21:28:15.795 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: browse
21:28:15.796 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: incoming
warn: Missing resource for key: browse
warn: Missing resource for key: incoming
21:28:15.942 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: browse
warn: Missing resource for key: browse
warn: Missing resource for key: waitTime
21:28:15.944 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: waitTime
warn: Missing resource for key: maxMessageTime
21:28:15.953 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: maxMessageTime
warn: Missing resource for key: confirm
21:28:15.955 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: confirm
warn: Missing resource for key: default
21:28:15.955 WARNING: impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing resource for key: default
fileLocation: null
21:28:15.959 INFO: org.osgi.framework.BundleActivator.start().89 Mailbox plug-in...[STARTED]
Info: Mailbox plug-in...[STARTED]
debug: sent request=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Content-Length: 0

debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>;tag=as61098680
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5,realm="phone.wenco.com",nonce="2c5e294e"
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: Authenticating a Register request.
debug: trying to authenticate using : MD5, kim_sip, phone.wenco.com, true, 2c5e294e, 00000001, xyz, REGISTER, sip:phone.wenco.com, , null
debug: Created authorization header: Authorization: Digest username="kim_sip",realm="phone.wenco.com",nonce="2c5e294e",uri="sip:phone.wenco.com",response="5e9d462a45fd6f9abbe705e4b09a7d0d",algorithm=MD5

debug: Returning authorization transaction.
debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKd01785901b2b5e8e6974fd2708407ef9;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKd01785901b2b5e8e6974fd2708407ef9;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>;tag=as61098680
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Date: Fri, 26 Jun 2009 03:28:25 GMT
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: enter registered state
debug: The provider changed state from: RegistrationState=Registering to: RegistrationState=Registered
debug: The provider changed state from: RegistrationState=Registering to: RegistrationState=Registered
debug: Dispatching Provider Status Change. Listeners=2 evt=ProviderPresenceStatusChangeEvent-[OldStatus=PresenceStatus:Offline, NewStatus=PresenceStatus:Online]
debug: status dispatching done.
debug: Dispatching stat. msg change. Listeners=2 evt=java.beans.PropertyChangeEvent[source=net.java.sip.communicator.impl.protocol.sip.ProtocolProviderServiceSipImpl@cb2185]
debug: status dispatching done.
debug: generated via headers:Via: SIP/2.0/UDP 192.168.2.95:5060

debug: generated contactHeader:Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>

debug: Dispatching a CallParticipant event to 0 listeners. event is: CallParticipantEvent: ID=1 source participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Unknown source call=Call: id=124598690248221078831 participants=1
debug: Dispatching a CallParticipantChangeEvent event to 1 listeners. event is: CallParticipantChangeEvent: type=CallParticipantStatusChange oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Unknown newV=net.java.sip.communicator.service.protocol.CallParticipantState:Initiating Call for participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Initiating Call
debug: Dispatching a CallEvent to 5 listeners. event is: CallEvent:[ id=1 Call=Call: id=124598690248221078831 participants=1]
outgoingCallCreated: null
Session.CallTracker: Call: id=124598690248221078831 participants=1 ready: false
created OutgoingCallTracker: [c:\db\sounds\phoneReminder.wav]
Session insureAudioFilesExists is not implemented
session. Call is connected, playing: [c:\db\sounds\phoneReminder.wav]
Session mediaService.playAudio not implemented
debug: registering format ilbc/rtp, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed with RTP manager
debug: registering format ALAW/rtp, 8000.0 Hz, 8-bit, Mono, Signed with RTP manager
debug: registering format speex/rtp, 8000.0 Hz, 8-bit, Mono, Signed with RTP manager
debug: registering format H264/RTP with RTP manager
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182, skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182, skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182, skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182, skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Will create media descs with: audio public address=/192.168.2.95:5000 and video public address=/192.168.2.95:5002
debug: We will be able to transmit in:
debug: unknown encoding format mpegaudio/rtp
debug: received response=
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bK0df49cdfbd46c87bf8ff5c79f2eb5ec6;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>;tag=as6567575c
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5,realm="phone.wenco.com",nonce="47aca73a"
Content-Length: 0

debug: Found 1 processor(s) for method INVITE
debug: Authenticating an INVITE request.
debug: trying to authenticate using : MD5, kim_sip, phone.wenco.com, true, 47aca73a, 00000001, xyz, INVITE, sip:14063661383@phone.wenco.com, v=0
o=kim_sip 0 0 IN IP4 192.168.2.95
s=-
c=IN IP4 192.168.2.95
t=0 0
m=audio 5000 RTP/AVP 0 8 97 3 5 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=zrtp-hash:1.10 1d486d9ff6653a351616ab66d6649d10a82f8adc55c82d12ec05674a7e36bc39
m=video 5002 RTP/AVP 99 34 26 31
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1
a=recvonly
a=zrtp-hash:1.10 873daca7f9d570514de22e9000f568944ab28b22eba583a6fb855cb382eb2ac3
, null
debug: Created authorization header: Proxy-Authorization: Digest username="kim_sip",realm="phone.wenco.com",nonce="47aca73a",uri="sip:14063661383@phone.wenco.com",response="60a31fd99e629e6415bd6fdd27f4a01d",algorithm=MD5

debug: Returning authorization transaction.
debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:14063661383@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method INVITE
debug: Dispatching a CallParticipantChangeEvent event to 3 listeners. event is: CallParticipantChangeEvent: type=CallParticipantStatusChange oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Initiating Call newV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting for participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Connecting
debug: received response=
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:14063661383@209.237.251.182>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 2603 2603 IN IP4 209.237.251.182
s=session
c=IN IP4 209.237.251.182
t=0 0
m=audio 10066 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

debug: Found 1 processor(s) for method INVITE
debug: received the following JMF SendStreamEvent - javax.media.rtp.event.NewSendStreamEvent=javax.media.rtp.event.NewSendStreamEvent[source = RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
debug: Dispatching a CallParticipantChangeEvent event to 3 listeners. event is: CallParticipantChangeEvent: type=CallParticipantStatusChange oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting newV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting* for participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Connecting*
debug: Dispatching a CallChange event to 7 listeners. event is: CallChangeEvent: type=CallState oldV=net.java.sip.communicator.service.protocol.CallState:Initializing newV=net.java.sip.communicator.service.protocol.CallState:In Progress
Session.CallStateChange: net.java.sip.communicator.service.protocol.CallState:In Progress
Session. newvalue
debug: call connected. starting streaming
debug: Transaction terminated for req=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: Transaction terminated for req=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 2 REGISTER
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKd01785901b2b5e8e6974fd2708407ef9
Authorization: Digest username="kim_sip",realm="phone.wenco.com",nonce="2c5e294e",uri="sip:phone.wenco.com",response="5e9d462a45fd6f9abbe705e4b09a7d0d",algorithm=MD5
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received a new incoming stream. javax.media.rtp.event.NewReceiveStreamEvent[source = RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
debug: Received a ControllerEvent: javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Unrealized,current=Realizing,target=Realized]
debug: Received a ControllerEvent: javax.media.RealizeCompleteEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Realizing,current=Realized,target=Realized]
debug: A player was realized and will be started.
debug: Received a ControllerEvent: javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Realized,current=Prefetching,target=Started]
debug: Received a ControllerEvent: javax.media.PrefetchCompleteEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Prefetching,current=Prefetched,target=Started]
debug: Received a ControllerEvent: javax.media.StartEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Prefetched,current=Started,target=Started,mediaTime=javax.media.Time@38c8c5,timeBaseTime=javax.media.Time@53b2c]
debug: Received a StartEvent
warn: container is null
debug: received request=
OPTIONS sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com SIP/2.0
Via: SIP/2.0/UDP 209.237.251.182:5060;branch=z9hG4bK1f60dfbe;rport=5060;received=209.237.251.182
From: "asterisk" <sip:asterisk@209.237.251.182>;tag=as5228afbf
To: <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>
Contact: <sip:asterisk@209.237.251.182>
Call-ID: 7e794f3d5fea2d1b5340ed4c43518400@209.237.251.182
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 26 Jun 2009 03:28:34 GMT
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Content-Length: 0

debug: Found 1 processor(s) for method OPTIONS
debug: Dialog terminated for req=gov.nist.javax.sip.stack.SIPDialog@e9493a
debug: received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:14063661383@209.237.251.182>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 2603 2604 IN IP4 209.237.251.182
s=session
c=IN IP4 209.237.251.182
t=0 0
m=audio 10066 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

debug: Found 1 processor(s) for method INVITE
debug: Dispatching a CallParticipantChangeEvent event to 3 listeners. event is: CallParticipantChangeEvent: type=CallParticipantStatusChange oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting* newV=net.java.sip.communicator.service.protocol.CallParticipantState:Connected for participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Connected
debug: received a new incoming stream. javax.media.rtp.event.NewReceiveStreamEvent[source = RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
debug: Received a ControllerEvent: javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Unrealized,current=Realizing,target=Realized]
debug: Received a ControllerEvent: javax.media.RealizeCompleteEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Realizing,current=Realized,target=Realized]
debug: A player was realized and will be started.
debug: Received a ControllerEvent: javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Realized,current=Prefetching,target=Started]
debug: Received a ControllerEvent: javax.media.PrefetchCompleteEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Prefetching,current=Prefetched,target=Started]
debug: Received a ControllerEvent: javax.media.StartEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Prefetched,current=Started,target=Started,mediaTime=javax.media.Time@1540a77,timeBaseTime=javax.media.Time@7b2d29]
debug: Received a StartEvent
debug: Transaction terminated for req=
INVITE sip:14063661383@phone.wenco.com SIP/2.0
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>
Content-Type: application/sdp
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff
Proxy-Authorization: Digest username="kim_sip",realm="phone.wenco.com",nonce="47aca73a",uri="sip:14063661383@phone.wenco.com",response="60a31fd99e629e6415bd6fdd27f4a01d",algorithm=MD5
Content-Length: 432

v=0
o=kim_sip 0 0 IN IP4 192.168.2.95
s=-
c=IN IP4 192.168.2.95
t=0 0
m=audio 5000 RTP/AVP 0 8 97 3 5 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=zrtp-hash:1.10 1d486d9ff6653a351616ab66d6649d10a82f8adc55c82d12ec05674a7e36bc39
m=video 5002 RTP/AVP 99 34 26 31
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1
a=recvonly
a=zrtp-hash:1.10 873daca7f9d570514de22e9000f568944ab28b22eba583a6fb855cb382eb2ac3

debug: Found 1 processor(s) for method INVITE
debug: received request=
BYE sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com SIP/2.0
Via: SIP/2.0/UDP 209.237.251.182:5060;branch=z9hG4bK7c10da18;rport=5060;received=209.237.251.182
From: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe
To: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

debug: Found 1 processor(s) for method BYE
debug: We seem to already have a tag in this dialog. Returning
debug: sent response SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.237.251.182:5060;branch=z9hG4bK7c10da18;rport=5060;received=209.237.251.182
From: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe
To: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 102 BYE
User-Agent: Info-Tell Operator1.2.1Windows Vista
Content-Length: 0

debug: Dispatching a CallParticipantChangeEvent event to 3 listeners. event is: CallParticipantChangeEvent: type=CallParticipantStatusChange oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Connected newV=net.java.sip.communicator.service.protocol.CallParticipantState:Disconnected for participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Disconnected
debug: Dispatching a CallParticipant event to 7 listeners. event is: CallParticipantEvent: ID=2 source participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Disconnected source call=Call: id=124598690248221078831 participants=0
Session.callParticipantRemoved: CallParticipantEvent: ID=2 source participant=sip:14063661383@phone.wenco.com <sip:14063661383@phone.wenco.com>;status=Disconnected source call=Call: id=124598690248221078831 participants=0
debug: Dispatching a CallChange event to 7 listeners. event is: CallChangeEvent: type=CallState oldV=net.java.sip.communicator.service.protocol.CallState:In Progress newV=net.java.sip.communicator.service.protocol.CallState:Ended
debug: Dispatching a CallEvent to 5 listeners. event is: CallEvent:[ id=3 Call=Call: id=124598690248221078831 participants=0]
21:28:40.722 WARNING: impl.media.CallSessionImpl.callStateChanged().2534 Stopping streaming.
Session.callEnded
Users.setCallEnded
Session without an interaction
Mailbox callEnded
Session.CallStateChange: net.java.sip.communicator.service.protocol.CallState:Ended
warn: Stopping streaming.
debug: received the following JMF SendStreamEvent - javax.media.rtp.event.StreamClosedEvent=javax.media.rtp.event.StreamClosedEvent[source = RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
Info: Creating datasource for:dsound://
MediaContol initiProcessor: net.java.sip.communicator.impl.media.codec.EncodingConfiguration@1e492d8
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.alaw.JavaEncoder is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.alaw.DePacketizer is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.alaw.Packetizer is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.ulaw.Packetizer is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.video.h264.JNIEncoder is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.video.h264.Packetizer is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.video.h264.JNIDecoder is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.video.ImageScaler is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.speex.JavaEncoder is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.speex.JavaDecoder is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaEncoder is already registered
debug: Codec : net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaDecoder is already registered
debug: Received a ControllerEvent: javax.media.StopByRequestEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Started,current=Prefetched,target=Prefetched,mediaTime=javax.media.Time@10a4a32]
debug: Received a ControllerEvent: javax.media.DeallocateEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Prefetched,current=Realized,target=Realized,mediaTime=javax.media.Time@11f91ac]
debug: Received a ControllerEvent: javax.media.ControllerClosedEvent[source=com.sun.media.content.unknown.Handler@cf9bf0]
debug: Received a ControllerClosedEvent
debug: Received a ControllerEvent: javax.media.StopByRequestEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Started,current=Prefetched,target=Prefetched,mediaTime=javax.media.Time@101f287]
debug: Received a ControllerEvent: javax.media.DeallocateEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Prefetched,current=Realized,target=Realized,mediaTime=javax.media.Time@1578426]
debug: Received a ControllerEvent: javax.media.ControllerClosedEvent[source=com.sun.media.content.unknown.Handler@126456a]
debug: Received a ControllerClosedEvent
debug: generated via headers:Via: SIP/2.0/UDP 192.168.2.95:5060

debug: generated contactHeader:Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>

debug: sent request=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 3 REGISTER
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKde0facaff8dcb78368e5794fb057eee4
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip" <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Content-Length: 0

debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKde0facaff8dcb78368e5794fb057eee4;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bKde0facaff8dcb78368e5794fb057eee4;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>;tag=as15958e4b
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5,realm="phone.wenco.com",nonce="2997f0aa"
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: Authenticating a Register request.
debug: trying to authenticate using : MD5, kim_sip, phone.wenco.com, true, 2997f0aa, 00000001, xyz, REGISTER, sip:phone.wenco.com, , null
debug: Created authorization header: Authorization: Digest username="kim_sip",realm="phone.wenco.com",nonce="2997f0aa",uri="sip:phone.wenco.com",response="a2031b9798b755065bd1f42a8717a844",algorithm=MD5

debug: Returning authorization transaction.
debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bK02bab5577b988cb0ad4ee57808efbb74;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.95:5060;branch=z9hG4bK02bab5577b988cb0ad4ee57808efbb74;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>;tag=as15958e4b
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Date: Fri, 26 Jun 2009 03:28:51 GMT
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER

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#9

Hey Kim,

Just had a look at your traces. There seems to be audio flowing both
ways so I don't think there's a capture/encoding problem at the SC side.

I've extracted the media from your traces and the audio that leaves SIP
Communicator sounds perfectly well. The one that comes back from
asterisk however sounds choppy indeed. This makes me think there might
be something wrong happening on the remote side.

Hope this helps
Emil

kim Fairchild wrote:

···

Emil Ivov wrote:

Thanks for responding so quickly.

I am sorry, I don't know what you mean by "which packet are you
using..". Package? I am running with current code from about 2 weeks ago.

I am including the wireshark capture and the output from the console of
Netbeans (I modified the logger to output there).

I started wireshark and then SC. I placed one outgoing call. And then I
quit.

Hey Kim,

Which packet are you using to run SIP Communicator? Could you please
send your complete log file as well as a wireshark capture from a
session with bad audio?

Thanks
Emil

kim Fairchild wrote:
  

Hello,

I have been able to make and receive calls with good quality for some
time but when I updated to new sources, I have bad outgoing connections.
The audio from Sip-Communicator is choppy and gets more and more laggey.
Looking at the debug output, it says that it is able to transmit in:

debug: We will be able to transmit in:
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp

Sorry, I am confused by this. Because of copyright issues, mpegaudio has
to be separately downloaded doesn't it? Would appreciate any help in
figuring this out. I include the debug output below.

BTW, is Stun disabled by default? How does SC do the NAT traversal
without it?

thanks, Kim

deploy-os-specific-bundles:
run:

Welcome to Felix.

debug: Service Impl:
net.java.sip.communicator.impl.configuration.ConfigurationActivator [
STARTED ]
debug: Using config file in $HOME/.sip-communicator:
C:\trunk.pure\sip-communicator.xml
debug: Service Impl:
net.java.sip.communicator.impl.configuration.ConfigurationActivator
[REGISTERED]
Info: Resource manager ... [REGISTERED]
Info: UI Service...[ STARTED ]
Info: UI Service ...[REGISTERED]
Info: Resource registered
net.java.sip.communicator.plugin.defaultresourcepack.DefaultColorPackImpl@1995d80
Info: Resource registered
net.java.sip.communicator.plugin.defaultresourcepack.DefaultImagePackImpl@6e293a
Info: Resource registered
net.java.sip.communicator.plugin.defaultresourcepack.DefaultLanguagePackImpl@1319c
Info: Resource registered
net.java.sip.communicator.plugin.defaultresourcepack.DefaultSettingsPackImpl@1479feb
Info: Resource registered
net.java.sip.communicator.plugin.defaultresourcepack.DefaultSoundPackImpl@1592174
Info: Default resources ... [REGISTERED]
debug: Started.
debug: SIP Protocol Provider Factory ... [REGISTERED]
debug: SIP Communicator Version: sip-communicator-1.2.1
debug: Started.
created a deviceConfiguration
MediaControl created
debug: Media Service ... [REGISTERED]
Starting mediaServiceimpl
deviceConfiguration.initialize
debug: Started.
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.icq
accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.icq
accounts
IOException in readRegistry: java.io.EOFException
debug: SIP Protocol Provider Factory ... [REGISTERED]
debug: Discovered 2 stored net.java.sip.communicator.impl.protocol.sip
accounts
debug: Loading account
net.java.sip.communicator.impl.protocol.sip.acc1244603766502
Info: Gibberish protocol implementation [STARTED].
Info: SSH protocol implementation [STARTED].
Info: Stun server address(null)/port(null) not set (or invalid).
Disabling STUN.
Info: Network Address Manager ...[ STARTED ]
Info: Network Address Manager Service ...[REGISTERED]
Info: Zeroconf protocol implementation [STARTED].
Info: IRC protocol implementation [STARTED].
Info: DICT protocol implementation [STARTED].
debug: Service Impl:
net.java.sip.communicator.impl.contactlist.ContactlistActivator [ STARTED ]
debug: Starting the meta contact list implementation.
debug: Service Impl:
net.java.sip.communicator.impl.contactlist.ContactlistActivator [REGISTERED]
Info: RSS protocol implementation [STARTED].
debug: User-Agent set to Info-Tell Operator/1.2.1
debug: New history service registered.
debug: Starting the msg history implementation.
Info: Message History Service ...[REGISTERED]
debug: New history service registered.
debug: Starting the call history implementation.
Info: Call History Service ...[REGISTERED]
Info: Audio Notifier Service...[ STARTED ]
Info: Audio Notifier Service ...[REGISTERED]
debug: Service Impl:
net.java.sip.communicator.impl.keybindings.KeybindingsActivator [ STARTED ]
Info: Notification Service...[ STARTED ]
Info: Notification Service ...[REGISTERED]
debug: Registering default event
IncomingMessage/PopupMessageAction/null/null
debug: Registering default event
IncomingMessage/SoundAction/resources/sounds/incomingMessage.wav/null
Info: Looking for Audio capturer
Info: DirectSound Capture Supported = true
debug: Registering default event IncomingCall/PopupMessageAction/null/null
debug: presence initialized with :true, true, 30, 3600 for null
debug: Registering default event
ProactiveNotification/PopupMessageAction/null/null
debug: Registering default event
SecurityMessage/PopupMessageAction/null/null
debug: Registering default event
CallSecurityOn/SoundAction/resources/sounds/zrtpSecure.wav/null
debug: Registering default event
CallSecurityError/SoundAction/resources/sounds/zrtpAlert.wav/null
Info: UI Service...[ STARTED ]
Info: UI Service ...[REGISTERED]
debug: Stored account for id SIP:kim_sip@phone.wenco.com for package
net.java.sip.communicator.impl.protocol.sip
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP:kim_sip@phone.wenco.com
debug: All contacts loaded for account SIP:kim_sip@phone.wenco.com
debug: Loading account
net.java.sip.communicator.impl.protocol.sip.acc1244609754970
debug: presence initialized with :true, true, 30, 3600 for null
debug: Stored account for id SIP:doha_amber@phone.wenco.com for package
net.java.sip.communicator.impl.protocol.sip
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP
debug: Service is a protocol provider.
debug: Handling registration of a new Protocol Provider.
debug: Adding protocol provider SIP:doha_amber@phone.wenco.com
debug: All contacts loaded for account SIP:doha_amber@phone.wenco.com
debug: Discovered 0 stored
net.java.sip.communicator.impl.protocol.jabber accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.msn
accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.yahoo
accounts
debug: Discovered 0 stored
net.java.sip.communicator.impl.protocol.gibberish accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.ssh
accounts
debug: Discovered 0 stored
net.java.sip.communicator.impl.protocol.zeroconf accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.irc
accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.dict
accounts
debug: Discovered 0 stored net.java.sip.communicator.impl.protocol.rss
accounts
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event:
net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.branding.AboutWindowPluginComponent@a6e0a9]
Info: ABOUT WINDOW ... [REGISTERED]
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event:
net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.branding.AboutWindowPluginComponent@f81402]
Info: CHAT ABOUT WINDOW ... [REGISTERED]
Info: Swing Notification ...[ STARTING ]
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event:
net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.updatechecker.UpdateCheckActivator$UpdateMenuButtonComponent@154ea79]
Info: No config file specified for update checker. Disabling update checks
debug: Updates are disabled. Faking latest version.
Info: Swing Notification ...[REGISTERED]
Info: added the following popup handler : SIP Communicator popups
21:28:14.218 INFO: doha.impl.phone.PhoneActivator.start().59 Phone
Service ...[Starting]
21:28:14.218 FINEST: doha.impl.phone.PhoneActivator.start().62 [entry] start
Info: Systray Service...[ STARTED ]
Info: Systray Service ...[REGISTERED]
PhoneActivator Start!
Info: Phone Service ...[Starting]
OperatorServiceImpl started!
Session created
startListener
Session.startListener turned on!
Session.startListener turned on!
users.getCallerIDDocument
loadDom: c:\db\admin\callerids.xml
Opening connection to: jdbc:mysql://ljade.com:3306/mydatabase
1 Database.java:412 open()
2 Database.java:763 getXML()
3 Database.java:750 getXMLFile()
4 DomServiceImpl.java:546 loadDom()
5 Users.java:241 getCallerIDDocument()
6 Users.java:37 <init>()
7 OperatorServiceImpl.java:108 <init>()
8 PhoneActivator.java:64 start()
9 SecureAction.java:589 startActivator()
Opening SQL Database connection...
debug: The provider changed state from: RegistrationState=Unregistered
to: RegistrationState=Registering
debug: The provider changed state from: RegistrationState=Unregistered
to: RegistrationState=Registering
debug: generated max forwards: Max-Forwards: 70

debug: generated via headers:Via: SIP/2.0/UDP 192.168.2.95:5060

debug: generated contactHeader:Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>

Info: DirectSoundAuto: Committed ok
Info: JavaSound Capture Supported = true
Info: JavaSoundAuto: Committed ok
debug: No FMJ javasound detected:
net.sf.fmj.media.cdp.javasound.CaptureDevicePlugger
Info: Looking for video capture devices
Info: Scanning for configured Audio Devices.
debug: Found 2 capture devices: [Ljavax.media.CaptureDeviceInfo;@8d41f2
Info: Found DirectSoundCapture as an audio capture device.
Info: Scanning for configured Video Devices.
Info: No Video Device was found.
defaultMediaControl.initialize
MediaControl initialized:
net.java.sip.communicator.impl.media.codec.EncodingConfiguration@1e492d8
debug: Adding package : net.java.sip.communicator.impl
debug: Adding package : net.sf.fmj
debug: Registering new protocol prefix list : [javax, com.sun, com.ibm,
net.java.sip.communicator.impl, net.sf.fmj]
Info: Creating datasource for:dsound://
MediaContol initiProcessor:
net.java.sip.communicator.impl.media.codec.EncodingConfiguration@1e492d8
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.alaw.JavaEncoder is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.alaw.DePacketizer is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.alaw.Packetizer is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.ulaw.Packetizer is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.video.h264.JNIEncoder is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.video.h264.Packetizer is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.video.h264.JNIDecoder is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.video.ImageScaler is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.speex.JavaEncoder is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.speex.JavaDecoder is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaEncoder is
successfully registered
debug: Codec
net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaDecoder is
successfully registered
connection: com.mysql.jdbc.Connection@149249e
... SQL Database connected in 0.863 seconds
query: select * from Callerids where ider = 'admin_callerids_0'
Local file is up-to-date localMark: 232 serverMark: 232
Cache.initializing...
Initial Interaction created - used for plugin
Info: trying database in phoneactivator
Info: Phone Service ...[REGISTERED]
Phone Opened Connection
21:28:15.184 INFO: doha.impl.phone.PhoneActivator.start().95 trying
database in phoneactivator
21:28:15.185 INFO: doha.impl.phone.PhoneActivator.start().104 Phone
Service ...[REGISTERED]
21:28:15.185 FINEST: doha.impl.phone.PhoneActivator.start().109 [exit] start
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event:
net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.whiteboard.WhiteboardMenuItem@58d7c2]
Info: WHITEBOARD... [REGISTERED]
Info: Handling registration of a new Account Wizard.
Info: Handling registration of a new Account Wizard.
Info: Handling registration of a new Account Wizard.
Info: Handling registration of a new Account Wizard.
Info: Handling registration of a new Account Wizard.
Info: Handling registration of a new Account Wizard.
Info: Handling registration of a new Account Wizard.
Info: Loading gibberish account wizard.
Info: Handling registration of a new Account Wizard.
Info: Gibberish account registration wizard [STARTED].
Info: Loading ssh account wizard.
Info: Handling registration of a new Account Wizard.
Info: SSH account registration wizard [STARTED].
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event:
net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.callhistoryform.ExtendedCallHistorySearchItem@959fa1]
Info: EXTENDED CALL HISTORY SEARCH... [REGISTERED]
Info: Loading rss account wizard.
Info: Handling registration of a new Account Wizard.
Info: RSS account registration wizard [STARTED].
Info: Loading zeroconf account wizard.
Info: Handling registration of a new Account Wizard.
Info: Zeroconf account registration wizard [STARTED].
Info: Loading irc account wizard.
Info: Handling registration of a new Account Wizard.
Info: IRC account registration wizard [STARTED].
Info: Handling registration of a new Plugin Component.
debug: Will dispatch the following plugin component event:
net.java.sip.communicator.impl.gui.event.PluginComponentEvent[source=net.java.sip.communicator.plugin.contactinfo.ContactInfoMenuItem@423606]
Info: CONTACT INFO... [REGISTERED]
debug: Found 2 already installed providers.
debug: Adding protocol provider SIP
debug: Adding protocol provider SIP
debug: Service Impl:
net.java.sip.communicator.plugin.keybindingchooser.KeybindingChooserActivator
[ STARTED ]
Info: PREFERENCES PLUGIN... [REGISTERED]
Info: Handling registration of a new Account Wizard.
Started the CMSActivator
Info: EXTENDED CALL HISTORY SEARCH... [REGISTERED]
Info: SIMPLE ACCOUNT REGISTRATION ...[STARTED]
debug: Starting the ShutdownTimeout service.
Info: Mailbox plug-in...[STARTING]
21:28:15.370 INFO: org.osgi.framework.BundleActivator.start().75 Mailbox
plug-in...[STARTING]
debug: Starting the mailbox implementation.
debug: Found 2 already installed providers.
debug: Adding protocol provider SIP
debug: Adding protocol provider SIP
warn: Missing resource for key: outgoing
21:28:15.374 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: outgoing
21:28:15.795 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: browse
21:28:15.796 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: incoming
warn: Missing resource for key: browse
warn: Missing resource for key: incoming
21:28:15.942 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: browse
warn: Missing resource for key: browse
warn: Missing resource for key: waitTime
21:28:15.944 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: waitTime
warn: Missing resource for key: maxMessageTime
21:28:15.953 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: maxMessageTime
warn: Missing resource for key: confirm
21:28:15.955 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: confirm
warn: Missing resource for key: default
21:28:15.955 WARNING:
impl.resources.ResourceManagementServiceImpl.getI18NString().451 Missing
resource for key: default
fileLocation: null
21:28:15.959 INFO: org.osgi.framework.BundleActivator.start().89 Mailbox
plug-in...[STARTED]
Info: Mailbox plug-in...[STARTED]
debug: sent request=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Content-Length: 0

debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>;tag=as61098680
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
WWW-Authenticate: Digest
algorithm=MD5,realm="phone.wenco.com",nonce="2c5e294e"
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: Authenticating a Register request.
debug: trying to authenticate using : MD5, kim_sip, phone.wenco.com,
true, 2c5e294e, 00000001, xyz, REGISTER, sip:phone.wenco.com, , null
debug: Created authorization header: Authorization: Digest
username="kim_sip",realm="phone.wenco.com",nonce="2c5e294e",uri="sip:phone.wenco.com",response="5e9d462a45fd6f9abbe705e4b09a7d0d",algorithm=MD5

debug: Returning authorization transaction.
debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKd01785901b2b5e8e6974fd2708407ef9;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKd01785901b2b5e8e6974fd2708407ef9;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>;tag=as61098680
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Expires: 3600
Contact:
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Date: Fri, 26 Jun 2009 03:28:25 GMT
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: enter registered state
debug: The provider changed state from: RegistrationState=Registering
to: RegistrationState=Registered
debug: The provider changed state from: RegistrationState=Registering
to: RegistrationState=Registered
debug: Dispatching Provider Status Change. Listeners=2
evt=ProviderPresenceStatusChangeEvent-[OldStatus=PresenceStatus:Offline,
NewStatus=PresenceStatus:Online]
debug: status dispatching done.
debug: Dispatching stat. msg change. Listeners=2
evt=java.beans.PropertyChangeEvent[source=net.java.sip.communicator.impl.protocol.sip.ProtocolProviderServiceSipImpl@cb2185]
debug: status dispatching done.
debug: generated via headers:Via: SIP/2.0/UDP 192.168.2.95:5060

debug: generated contactHeader:Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>

debug: Dispatching a CallParticipant event to 0 listeners. event is:
CallParticipantEvent: ID=1 source
participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Unknown source call=Call:
id=124598690248221078831 participants=1
debug: Dispatching a CallParticipantChangeEvent event to 1 listeners.
event is: CallParticipantChangeEvent: type=CallParticipantStatusChange
oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Unknown
newV=net.java.sip.communicator.service.protocol.CallParticipantState:Initiating
Call for participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Initiating Call
debug: Dispatching a CallEvent to 5 listeners. event is: CallEvent:[
id=1 Call=Call: id=124598690248221078831 participants=1]
outgoingCallCreated: null
Session.CallTracker: Call: id=124598690248221078831 participants=1
ready: false
created OutgoingCallTracker: [c:\db\sounds\phoneReminder.wav]
Session insureAudioFilesExists is not implemented
session. Call is connected, playing: [c:\db\sounds\phoneReminder.wav]
Session mediaService.playAudio not implemented
debug: registering format ilbc/rtp, 8000.0 Hz, 16-bit, Mono,
LittleEndian, Signed with RTP manager
debug: registering format ALAW/rtp, 8000.0 Hz, 8-bit, Mono, Signed with
RTP manager
debug: registering format speex/rtp, 8000.0 Hz, 8-bit, Mono, Signed with
RTP manager
debug: registering format H264/RTP with RTP manager
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182,
skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182,
skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182,
skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Stun is disabled for destination phone.wenco.com/209.237.251.182,
skipping mapped address recovery (useStun=false, IPv6@=false).
debug: Will create media descs with: audio public
address=/192.168.2.95:5000 and video public address=/192.168.2.95:5002
debug: We will be able to transmit in:
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: unknown encoding format mpegaudio/rtp
debug: received response=
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bK0df49cdfbd46c87bf8ff5c79f2eb5ec6;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>;tag=as6567575c
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Proxy-Authenticate: Digest
algorithm=MD5,realm="phone.wenco.com",nonce="47aca73a"
Content-Length: 0

debug: Found 1 processor(s) for method INVITE
debug: Authenticating an INVITE request.
debug: trying to authenticate using : MD5, kim_sip, phone.wenco.com,
true, 47aca73a, 00000001, xyz, INVITE, sip:14063661383@phone.wenco.com, v=0
o=kim_sip 0 0 IN IP4 192.168.2.95
s=-
c=IN IP4 192.168.2.95
t=0 0
m=audio 5000 RTP/AVP 0 8 97 3 5 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=zrtp-hash:1.10
1d486d9ff6653a351616ab66d6649d10a82f8adc55c82d12ec05674a7e36bc39
m=video 5002 RTP/AVP 99 34 26 31
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1
a=recvonly
a=zrtp-hash:1.10
873daca7f9d570514de22e9000f568944ab28b22eba583a6fb855cb382eb2ac3
, null
debug: Created authorization header: Proxy-Authorization: Digest
username="kim_sip",realm="phone.wenco.com",nonce="47aca73a",uri="sip:14063661383@phone.wenco.com",response="60a31fd99e629e6415bd6fdd27f4a01d",algorithm=MD5

debug: Returning authorization transaction.
debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:14063661383@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method INVITE
debug: Dispatching a CallParticipantChangeEvent event to 3 listeners.
event is: CallParticipantChangeEvent: type=CallParticipantStatusChange
oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Initiating
Call
newV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting
for participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Connecting
debug: received response=
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:14063661383@209.237.251.182>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 2603 2603 IN IP4 209.237.251.182
s=session
c=IN IP4 209.237.251.182
t=0 0
m=audio 10066 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

debug: Found 1 processor(s) for method INVITE
debug: received the following JMF SendStreamEvent -
javax.media.rtp.event.NewSendStreamEvent=javax.media.rtp.event.NewSendStreamEvent[source
= RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address
        RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
debug: Dispatching a CallParticipantChangeEvent event to 3 listeners.
event is: CallParticipantChangeEvent: type=CallParticipantStatusChange
oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting
newV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting*
for participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Connecting*
debug: Dispatching a CallChange event to 7 listeners. event is:
CallChangeEvent: type=CallState
oldV=net.java.sip.communicator.service.protocol.CallState:Initializing
newV=net.java.sip.communicator.service.protocol.CallState:In Progress
Session.CallStateChange:
net.java.sip.communicator.service.protocol.CallState:In Progress
Session. newvalue
debug: call connected. starting streaming
debug: Transaction terminated for req=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 1 REGISTER
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKa299b316f166a9bd955239c862c99e95
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: Transaction terminated for req=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 5074ecf055ae252450fe4ebb4bd5df41@0:0:0:0:0:0:0:0
CSeq: 2 REGISTER
From: <sip:kim_sip@phone.wenco.com>;tag=e5d5cecc
To: <sip:kim_sip@phone.wenco.com>
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKd01785901b2b5e8e6974fd2708407ef9
Authorization: Digest
username="kim_sip",realm="phone.wenco.com",nonce="2c5e294e",uri="sip:phone.wenco.com",response="5e9d462a45fd6f9abbe705e4b09a7d0d",algorithm=MD5
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received a new incoming stream.
javax.media.rtp.event.NewReceiveStreamEvent[source = RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address
        RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
debug: Received a ControllerEvent:
javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Unrealized,current=Realizing,target=Realized]
debug: Received a ControllerEvent:
javax.media.RealizeCompleteEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Realizing,current=Realized,target=Realized]
debug: A player was realized and will be started.
debug: Received a ControllerEvent:
javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Realized,current=Prefetching,target=Started]
debug: Received a ControllerEvent:
javax.media.PrefetchCompleteEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Prefetching,current=Prefetched,target=Started]
debug: Received a ControllerEvent:
javax.media.StartEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Prefetched,current=Started,target=Started,mediaTime=javax.media.Time@38c8c5,timeBaseTime=javax.media.Time@53b2c]
debug: Received a StartEvent
warn: container is null
debug: received request=
OPTIONS
sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com
SIP/2.0
Via: SIP/2.0/UDP
209.237.251.182:5060;branch=z9hG4bK1f60dfbe;rport=5060;received=209.237.251.182
From: "asterisk" <sip:asterisk@209.237.251.182>;tag=as5228afbf
To:
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>
Contact: <sip:asterisk@209.237.251.182>
Call-ID: 7e794f3d5fea2d1b5340ed4c43518400@209.237.251.182
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 26 Jun 2009 03:28:34 GMT
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Content-Length: 0

debug: Found 1 processor(s) for method OPTIONS
debug: Dialog terminated for req=gov.nist.javax.sip.stack.SIPDialog@e9493a
debug: received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:14063661383@209.237.251.182>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 2603 2604 IN IP4 209.237.251.182
s=session
c=IN IP4 209.237.251.182
t=0 0
m=audio 10066 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

debug: Found 1 processor(s) for method INVITE
debug: Dispatching a CallParticipantChangeEvent event to 3 listeners.
event is: CallParticipantChangeEvent: type=CallParticipantStatusChange
oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Connecting*
newV=net.java.sip.communicator.service.protocol.CallParticipantState:Connected
for participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Connected
debug: received a new incoming stream.
javax.media.rtp.event.NewReceiveStreamEvent[source = RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address
        RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
debug: Received a ControllerEvent:
javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Unrealized,current=Realizing,target=Realized]
debug: Received a ControllerEvent:
javax.media.RealizeCompleteEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Realizing,current=Realized,target=Realized]
debug: A player was realized and will be started.
debug: Received a ControllerEvent:
javax.media.TransitionEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Realized,current=Prefetching,target=Started]
debug: Received a ControllerEvent:
javax.media.PrefetchCompleteEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Prefetching,current=Prefetched,target=Started]
debug: Received a ControllerEvent:
javax.media.StartEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Prefetched,current=Started,target=Started,mediaTime=javax.media.Time@1540a77,timeBaseTime=javax.media.Time@7b2d29]
debug: Received a StartEvent
debug: Transaction terminated for req=
INVITE sip:14063661383@phone.wenco.com SIP/2.0
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
From: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
To: <sip:14063661383@phone.wenco.com>
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>
Content-Type: application/sdp
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bK8af6f076971a62df8811f2dd68dd48ff
Proxy-Authorization: Digest
username="kim_sip",realm="phone.wenco.com",nonce="47aca73a",uri="sip:14063661383@phone.wenco.com",response="60a31fd99e629e6415bd6fdd27f4a01d",algorithm=MD5
Content-Length: 432

v=0
o=kim_sip 0 0 IN IP4 192.168.2.95
s=-
c=IN IP4 192.168.2.95
t=0 0
m=audio 5000 RTP/AVP 0 8 97 3 5 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=zrtp-hash:1.10
1d486d9ff6653a351616ab66d6649d10a82f8adc55c82d12ec05674a7e36bc39
m=video 5002 RTP/AVP 99 34 26 31
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1
a=recvonly
a=zrtp-hash:1.10
873daca7f9d570514de22e9000f568944ab28b22eba583a6fb855cb382eb2ac3

debug: Found 1 processor(s) for method INVITE
debug: received request=
BYE
sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com
SIP/2.0
Via: SIP/2.0/UDP
209.237.251.182:5060;branch=z9hG4bK7c10da18;rport=5060;received=209.237.251.182
From: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe
To: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

debug: Found 1 processor(s) for method BYE
debug: We seem to already have a tag in this dialog. Returning
debug: sent response SIP/2.0 200 OK
Via: SIP/2.0/UDP
209.237.251.182:5060;branch=z9hG4bK7c10da18;rport=5060;received=209.237.251.182
From: <sip:14063661383@phone.wenco.com>;tag=as5f83f2fe
To: <sip:kim_sip@phone.wenco.com>;tag=ce7b1a74
Call-ID: 19643ce99ef64e53200f3f5845abd8d3@0:0:0:0:0:0:0:0
CSeq: 102 BYE
User-Agent: Info-Tell Operator1.2.1Windows Vista
Content-Length: 0

debug: Dispatching a CallParticipantChangeEvent event to 3 listeners.
event is: CallParticipantChangeEvent: type=CallParticipantStatusChange
oldV=net.java.sip.communicator.service.protocol.CallParticipantState:Connected
newV=net.java.sip.communicator.service.protocol.CallParticipantState:Disconnected
for participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Disconnected
debug: Dispatching a CallParticipant event to 7 listeners. event is:
CallParticipantEvent: ID=2 source
participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Disconnected source call=Call:
id=124598690248221078831 participants=0
Session.callParticipantRemoved: CallParticipantEvent: ID=2 source
participant=sip:14063661383@phone.wenco.com
<sip:14063661383@phone.wenco.com>;status=Disconnected source call=Call:
id=124598690248221078831 participants=0
debug: Dispatching a CallChange event to 7 listeners. event is:
CallChangeEvent: type=CallState
oldV=net.java.sip.communicator.service.protocol.CallState:In Progress
newV=net.java.sip.communicator.service.protocol.CallState:Ended
debug: Dispatching a CallEvent to 5 listeners. event is: CallEvent:[
id=3 Call=Call: id=124598690248221078831 participants=0]
21:28:40.722 WARNING: impl.media.CallSessionImpl.callStateChanged().2534
Stopping streaming.
Session.callEnded
Users.setCallEnded
Session without an interaction
Mailbox callEnded
Session.CallStateChange:
net.java.sip.communicator.service.protocol.CallState:Ended
warn: Stopping streaming.
debug: received the following JMF SendStreamEvent -
javax.media.rtp.event.StreamClosedEvent=javax.media.rtp.event.StreamClosedEvent[source
= RTPManager
        SSRCCache com.sun.media.rtp.SSRCCache@1f0d7f5
        Dataport 0
        Controlport 0
        Address
        RTPForwarder com.sun.media.rtp.util.PacketForwarder@f23491
        RTPDemux com.sun.media.rtp.RTPDemultiplexer@1593ce6]
Info: Creating datasource for:dsound://
MediaContol initiProcessor:
net.java.sip.communicator.impl.media.codec.EncodingConfiguration@1e492d8
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.alaw.JavaEncoder is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.alaw.DePacketizer is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.alaw.Packetizer is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.ulaw.Packetizer is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.video.h264.JNIEncoder is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.video.h264.Packetizer is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.video.h264.JNIDecoder is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.video.ImageScaler is already
registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.speex.JavaEncoder is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.speex.JavaDecoder is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaEncoder is
already registered
debug: Codec :
net.java.sip.communicator.impl.media.codec.audio.ilbc.JavaDecoder is
already registered
debug: Received a ControllerEvent:
javax.media.StopByRequestEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Started,current=Prefetched,target=Prefetched,mediaTime=javax.media.Time@10a4a32]
debug: Received a ControllerEvent:
javax.media.DeallocateEvent[source=com.sun.media.content.unknown.Handler@cf9bf0,previous=Prefetched,current=Realized,target=Realized,mediaTime=javax.media.Time@11f91ac]
debug: Received a ControllerEvent:
javax.media.ControllerClosedEvent[source=com.sun.media.content.unknown.Handler@cf9bf0]
debug: Received a ControllerClosedEvent
debug: Received a ControllerEvent:
javax.media.StopByRequestEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Started,current=Prefetched,target=Prefetched,mediaTime=javax.media.Time@101f287]
debug: Received a ControllerEvent:
javax.media.DeallocateEvent[source=com.sun.media.content.unknown.Handler@126456a,previous=Prefetched,current=Realized,target=Realized,mediaTime=javax.media.Time@1578426]
debug: Received a ControllerEvent:
javax.media.ControllerClosedEvent[source=com.sun.media.content.unknown.Handler@126456a]
debug: Received a ControllerClosedEvent
debug: generated via headers:Via: SIP/2.0/UDP 192.168.2.95:5060

debug: generated contactHeader:Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>

debug: sent request=
REGISTER sip:phone.wenco.com SIP/2.0
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 3 REGISTER
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKde0facaff8dcb78368e5794fb057eee4
Max-Forwards: 70
User-Agent: Info-Tell Operator1.2.1Windows Vista
Expires: 3600
Contact: "kim_sip"
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Content-Length: 0

debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKde0facaff8dcb78368e5794fb057eee4;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bKde0facaff8dcb78368e5794fb057eee4;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>;tag=as15958e4b
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
WWW-Authenticate: Digest
algorithm=MD5,realm="phone.wenco.com",nonce="2997f0aa"
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: Authenticating a Register request.
debug: trying to authenticate using : MD5, kim_sip, phone.wenco.com,
true, 2997f0aa, 00000001, xyz, REGISTER, sip:phone.wenco.com, , null
debug: Created authorization header: Authorization: Digest
username="kim_sip",realm="phone.wenco.com",nonce="2997f0aa",uri="sip:phone.wenco.com",response="a2031b9798b755065bd1f42a8717a844",algorithm=MD5

debug: Returning authorization transaction.
debug: received response=
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bK02bab5577b988cb0ad4ee57808efbb74;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:kim_sip@209.237.251.182>
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER
debug: received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.95:5060;branch=z9hG4bK02bab5577b988cb0ad4ee57808efbb74;received=206.81.217.93
From: <sip:kim_sip@phone.wenco.com>;tag=3887af51
To: <sip:kim_sip@phone.wenco.com>;tag=as15958e4b
Call-ID: 001c9512fca429a92eede0d6aa82d3b3@0:0:0:0:0:0:0:0
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Expires: 3600
Contact:
<sip:kim_sip@192.168.2.95:5060;transport=udp;registering_acc=phone_wenco_com>;expires=3600
Date: Fri, 26 Jun 2009 03:28:51 GMT
Content-Length: 0

debug: Found 1 processor(s) for method REGISTER

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--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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