Category Archives: Jitsi

Jitsi Setup

Go to http://jitsi.org/ in your browser. Download the program onto your computer. Jitsi is an audio/video and chat application that supports protocols such as SIP, XMPP/Jabber, AIM/ICQ, Windows Live, Yahoo! and many other useful features.

When Jitsi is installed, go to the applications folder and open it.

Select the SIP account type in the Network pull down in the new account window. It should look like this

In the Sip id box, add the username & server detail we sent to you in the email (for example foo@ostel.co).

In the Password box, Add the password we sent to you in the email.

Click the Advanced box

Click the Connection tab. Fill out the form fields to match this screenshot

Click the Next button. Confirm the settings are correct on the next screen. Click Sign In

You should see it read "Registering" for a few seconds until the bar to the right of your account name turns Green and reads SIP ON Online.

Congratulations! You've successfully signed in.

You can fetch a package or subscribe to the repository.

For details on the current status with the official Debian packaging, please refer to Debian Bug report logs.

Known issues making a ZRTP initiation between clients.

Current successes for ZRTP: Jitsi OSX -> Groundwire / Ostel -> Jitsi OSX

Current failings for ZRTP: Groundwire -> Jitsi OSX / Jitsi OSX -> Ostel

Secure video calls, conferencing, chat, desktop sharing, file transfer, support for your favorite OS, and IM network. All this, and more, in Jitsi - the most complete and advanced open source communicator.

The ever growing use of Voice over IP (VoIP) and other media applications triggered a more widespread use of the Real-time Transfer Protocol (RTP). Thise protocols is the workhorse for VoIP applications. Many VoIP applications send RTP data over the public Internet in clear, thus the data is not protected from eavesdropping or modification. Therefore most VoIP applications are regarded insecure today. During the last years several activities started to enhance the security of RTP.

The Secure Real-time Transfer Protocol (SRTP) enhances security for RTP and provides integrity and confidentiality for RTP media connections. To use SRTP in an efficient way VoIP applications should be able to negotiate keys and other parameters in an automatic fashion.

ZRTP is a protocol that negotiates the keys and other information required to setup a SRTP audio and video session

While it is important to look at the technology, the protocols and alike, it is also important to look at the implications a specific technology may have on its implementation, deployment, and usability. Usability is of major importance for VoIP peer-to-peer applications: these applications are mainly used by non-IT persons. Therefore the handling must be simple, easy to use, and shall not require special infrastructure or registration.

http://jitsi.org/index.php/Documentation/ZrtpFAQ

Original post:
Jitsi - OSTN - Guardian Project Open Dev

Jitsi Original author(s) Emil Ivov Developer(s) Jitsi Team and Contributors Initial release 2003(2003) Stable release 2.8 (build.5426) (March19, 2015; 11 months ago(2015-03-19)) [] Preview release 2.9 (nightly) [] Development status Active Written in Java Operating system Linux, Mac OS X, Windows (all Java supported) Size 52.4 MB Windows (bundles its own private JRE)[1] 78.8MB Mac OS X (includes private JRE)[2] 22MB Linux 65MB source code[3] Available in Asturian, English, French, German, Bulgarian, Japanese, Spanish, Italian, Romanian, Greek and 25 more Type Voice over Internet Protocol / instant messaging / videoconferencing License Apache Website jitsi.org

Jitsi (formerly SIP Communicator) is a free and open source multiplatform[4]voice (VoIP), videoconferencing and instant messaging application for Windows, Linux, Mac OS X and Android. It supports several popular instant-messaging and telephony protocols, including open recognised encryption protocols for chat (OTR) and voice/video/streaming and voice/video conferencing (SIP/RTP/SRTP/ZRTP), as well as built-in IPv6, NAT traversal and DNSSEC. Jitsi and its source code are released under the terms of the Apache Software Licence.[5]

Work on Jitsi (then SIP Communicator) started in 2003 in the context of a student project by Emil Ivov at the University of Strasbourg.[6] It was originally released as an example video phone in the JAIN-SIP stack and later spun off as a standalone project.[7]

Originally the project was mostly used as an experimentation tool because of its support for IPv6.[8][9] Through the years, as the project gathered members, it also added support for protocols other than SIP.

Jitsi has received support from various institutions such as the NLnet Foundation,[10][11] the University of Strasbourg and the Region of Alsace[12] and it has also had multiple participations in the Google Summer of Code program.[13][14]

In 2009, Emil Ivov founded the BlueJimp company which has employed some of Jitsis main contributors[15][16] in order to offer professional support and development services[17] related to the project.

In 2011, after successfully adding support for audio/video communication over XMPPs Jingle extensions, the project was renamed to Jitsi since it was no longer a SIP only Communicator.[18][19] This name originates from the Bulgarian (wires).[20]

On November 4, 2014, Jitsi + Ostel scored 6 out of 7 points on the Electronic Frontier Foundations secure messaging scorecard. They lost a point because there has not been a recent independent code audit.[21]

On February 1, 2015, Hristo Terezov, Ingo Bauersachs and the rest of the team released [22] version 2.6 from their stand at the Free and Open Source Software Developers European Meeting 2015 event in Brussels. This release includes security fixes, removes support of the deprecated MSN protocol, along with SSLv3 in XMPP. Among other notable improvements, the OS X version bundles a Java 8 runtime, enables echo cancelling by default, and uses the CoreAudio subsystem. The Linux build addresses font issues with the GTK+ native LookAndFeel, and fixes some long standing issues about microphone level on call setup when using the PulseAudio sound system. This release also adds the embedded Java database Hyper SQL Database to improve performance for users with huge configuration files, a feature which is disabled by default. A full list of changes is [23] available on the project web site.

Jitsi supports multiple operating systems, including Windows as well as Unix-like systems such as Linux, Mac OS X and BSD. Beta packages built for Android are available[24] but the projects roadmap describes the porting to Android as on hold.[25] It also includes:[26]

The following protocols are currently supported by Jitsi:[4]

Jitsi is mostly written in Java[31] which helps reuse most of the same code over the various operating systems it works on. Its GUI is based upon Swing. The project also uses native code for the implementation of platform specific tasks such as audio/video capture and rendering, IP address selection, and access to native popup notification systems such as Growl.

The project uses the Apache Felix OSGi implementation[32] for modularity.

Among others Jitsi uses the JAIN-SIP protocol stack for SIP support and the Jive Software Smack library [33] for XMPP.[34]

As Jitsi can handle IPv6 it is especially interesting for direct PC-to-PC (peer-to-peer) communication, for instance, if both sides were trapped behind NAT routers, but could obtain a reachable IPv6 address via a tunnel-broker.[citation needed]

The Jitsi community has also completed an ICE implementation called ice4j.org, which it uses to provide NAT traversal capabilities, and assist IPv4 to IPv6 transition.[35]

Audio systems supported are PortAudio, PulseAudio and WASAPI (Windows Audio Session API).

Go here to see the original: Jitsi WOW.com

More here:
Jitsi WOW.com | Prometheism.net

Clearly, Skype is the world's most popular voice and video chat client and it supports a wide variety of platforms. But if you are looking for an alternative that affords you privacy and more control, Jitsi is the answer. Touted by security conscious netizens like the Tor Project's own Jacob Appelbaum as an ideal alternative to proprietary software like Skype, Jitsi lets you audio and video conference on your terms using your own private Session Initiation Protocol (SIP) server or any SIP service.

Jitsi is written in Java for cross-platform compatibility with other operating systems. Although this can translate into a slightly sluggish experience and a rather bland user interface, the software is remarkably flexible in its own right. In addition to supporting traditional SIP for online communications, Google Talk's protocol (XMPP) is also supported out of the box for audio and video chats as well as AIM, ICQ, Facebook, Yahoo and MSN. Jitsi also provides a means to encrypt VoIP traffic using SRTP or ZRTP encryption methods, which is something Skype doesn't provide and is a rarity amongst most SIP / VoIP clients today.

With the emergence of IPv6 connectivity, Jitsi is capable of initiating direct connect VoIP sessions, simply by providing the appropriate IPv6 address of the machine to connect to. For anyone behind NAT on a home router using IPv4 connections, tunnel brokers that provide IPv4 to IPv6 address translation can be used to get around pesky limitations imposed by NAT. Basically, this means that anyone from outside your network will typically have no problem reaching you.

As far as downsides go, the only issue I see with Jitsi, at least on the SIP side of the equation, is that SIP isn't nearly as commonplace as something like Skype and it may require additional VoIP configuration knowhow. With XMPP support available, at least Jitsi users can leverage their Google contacts right out of the box and get an experience similar to a more traditional SIP setup. Obviously though, Google will have access to your conversation data, whereby setting up a true SIP session gives you more control over your privacy.

For what it's worth, Jitsi is a fairly decent VoIP and chat client. Though not as shiny looking as its proprietary competitor Skype, Jitsi is still worth giving a try. It's free and if you don't have a resource-constrained system, the software should suit your online communication needs fairly nicely. Just be sure that you have the latest version of Oracle's Java Runtime Environment for best results. With Microsoft taking control of Skype and casting doubt on its future as a cross-platform service as well as the possibility of a programmed "back door" into the network, Jitsi could potentially fill the void nicely as an eventual replacement.

Read more from the original source:
Review: Jitsi the ultimate SIP voice and video client ...

Jitsi Original author(s) Emil Ivov Developer(s) Jitsi Team and Contributors Initial release 2003(2003) Stable release 2.8 (build.5426) (March19, 2015; 11 months ago(2015-03-19)) [] Preview release 2.9 (nightly) [] Development status Active Written in Java Operating system Linux, Mac OS X, Windows (all Java supported) Size 52.4 MB Windows (bundles its own private JRE)[1] 78.8MB Mac OS X (includes private JRE)[2] 22MB Linux 65MB source code[3] Available in Asturian, English, French, German, Bulgarian, Japanese, Spanish, Italian, Romanian, Greek and 25 more Type Voice over Internet Protocol / instant messaging / videoconferencing License Apache Website jitsi.org

Jitsi (formerly SIP Communicator) is a free and open source multiplatform[4]voice (VoIP), videoconferencing and instant messaging application for Windows, Linux, Mac OS X and Android. It supports several popular instant-messaging and telephony protocols, including open recognised encryption protocols for chat (OTR) and voice/video/streaming and voice/video conferencing (SIP/RTP/SRTP/ZRTP), as well as built-in IPv6, NAT traversal and DNSSEC. Jitsi and its source code are released under the terms of the Apache Software Licence.[5]

Work on Jitsi (then SIP Communicator) started in 2003 in the context of a student project by Emil Ivov at the University of Strasbourg.[6] It was originally released as an example video phone in the JAIN-SIP stack and later spun off as a standalone project.[7]

Originally the project was mostly used as an experimentation tool because of its support for IPv6.[8][9] Through the years, as the project gathered members, it also added support for protocols other than SIP.

Jitsi has received support from various institutions such as the NLnet Foundation,[10][11] the University of Strasbourg and the Region of Alsace[12] and it has also had multiple participations in the Google Summer of Code program.[13][14]

In 2009, Emil Ivov founded the BlueJimp company which has employed some of Jitsi's main contributors[15][16] in order to offer professional support and development services[17] related to the project.

In 2011, after successfully adding support for audio/video communication over XMPPs Jingle extensions, the project was renamed to Jitsi since it was no longer "a SIP only Communicator".[18][19] This name originates from the Bulgarian "" (wires).[20]

On November 4, 2014, "Jitsi + Ostel" scored 6 out of 7 points on the Electronic Frontier Foundation's secure messaging scorecard. They lost a point because there has not been a recent independent code audit.[21]

On February 1, 2015, Hristo Terezov, Ingo Bauersachs and the rest of the team released [22] version 2.6 from their stand at the Free and Open Source Software Developers' European Meeting 2015 event in Brussels. This release includes security fixes, removes support of the deprecated MSN protocol, along with SSLv3 in XMPP. Among other notable improvements, the OS X version bundles a Java 8 runtime, enables echo cancelling by default, and uses the CoreAudio subsystem. The Linux build addresses font issues with the GTK+ native LookAndFeel, and fixes some long standing issues about microphone level on call setup when using the PulseAudio sound system. This release also adds the embedded Java database Hyper SQL Database to improve performance for users with huge configuration files, a feature which is disabled by default. A full list of changes is [23] available on the project web site.

Jitsi supports multiple operating systems, including Windows as well as Unix-like systems such as Linux, Mac OS X and BSD. "Beta" packages built for Android are available[24] but the project's roadmap describes the porting to Android as "on hold".[25] It also includes:[26]

The following protocols are currently supported by Jitsi:[4]

Jitsi is mostly written in Java[31] which helps reuse most of the same code over the various operating systems it works on. Its GUI is based upon Swing. The project also uses native code for the implementation of platform specific tasks such as audio/video capture and rendering, IP address selection, and access to native popup notification systems such as Growl.

The project uses the Apache Felix OSGi implementation[32] for modularity.

Among others Jitsi uses the JAIN-SIP protocol stack for SIP support and the Jive Software Smack library [33] for XMPP.[34]

As Jitsi can handle IPv6 it is especially interesting for direct PC-to-PC (peer-to-peer) communication, for instance, if both sides were 'trapped' behind NAT routers, but could obtain a reachable IPv6 address via a tunnel-broker.[citation needed]

The Jitsi community has also completed an ICE implementation called ice4j.org, which it uses to provide NAT traversal capabilities, and assist IPv4 to IPv6 transition.[35]

Audio systems supported are PortAudio, PulseAudio and WASAPI (Windows Audio Session API).

Go here to see the original:
Jitsi - WOW.com

Openfire Jabber/XMPP is a server written in JAVA. This is free software and is also official support. Management has a WEB panel and it works on 9090 (http) and 9091 (https) ports.It supports Plugins(extensions), SSL/TLS, can connect to the database(Oracle, MSSQL, PostgreSQL, DB2, Sybase ASE, MySQL or internal database HSQLDB) via JDBC, can connect LDAP groups and filter by groups, has the ability of users registration based on to different sources and supporting different languages. Most of the management is done via the web interface. The official website is http://www.igniterealtime.org/.

Features:

Supported client programs:

Add SRV records in your DNS server as follows: openfire IN A 94.20.81.149 _jabber._tcp.jabber.unixmen.com. IN SRV 0 0 5269 jabber.unixmen.com. _xmpp-client._tcp.jabber.unixmen.com. IN SRV 0 0 5222 jabber.unixmen.com. _xmpp-server._tcp.jabber.unixmen.com. IN SRV 0 0 5269 jabber.unixmen.com.

Before all configuration we will create MySQL database, user and password because we will use this in next configurations: mysql -uroot -p mysql> CREATE DATABASE openfire; mysql> GRANT ALL PRIVILEGES ON openfire.* TO [emailprotected] IDENTIFIED BY 0penfire0bepassword; mysql> FLUSH PRIVILEGES;

Before the installation absolutely update the ports. cd /usr/ports/net-im/openfire Go to the port folder make config choose the modules make install install

echo openfire_enable=YES >> /etc/rc.conf Add to the startup /usr/local/etc/rc.d/openfire start Start the daemon

sockstat -l | grep openfire Check for listen openfire java 56187 26 tcp4 *:9090 *:* openfire java 56187 29 stream (not connected)

Then go to the http://openfire.unixmen.com:9090 page. You will get the following page (select English and click the continue button):

Write domain name and click the Continue button:

To select different type of database select the Standart Database Connection and click the Continue button:

The selection MySQL database. Write username, password and database URL as the following syntax and click the Continue button: jdbc:mysql://localhost:3306/openfire?rewriteBatchedStatements=true:

Select Default and click Continue button:

In the opened page add the email for administrator account, type password twice and click the continue button. Administrator login name will be admin:

Last configuration page will be as below. Go to the Login to the admin console for login:

Write admin user name and password for this user:

At the end the opened page will be as below:

Create some users as the following template:

Add group:

Then go to this group and add created users to this group:

Now configuration for client program. Download Spark client program from http://www.igniterealtime.org/downloads/download-landing.jsp?file=spark/spark_2_7_0.exe link and install.

As the following screen configuring user faxri.iskandarov:

Check monitoring service plugin (Must be installed):

Then go to the Server -> Archiving -> Archiving Settings section and select logging between our XMPP clients (as the following screen):

Even you are able to get meetings through web. For this go to the http://openfire.unixmen.com:7070/jitsi/apps/ofmeet link. This channel will not be crypted. For crypted channel go to the http://openfire.unixmen.com:7443/jitsi/apps/ofmeet link.

Note: If you are using Jitsi client program you dont need any plugin for call and any SIP number. Because Jitsi client can call with audio/video over XMPP like as Microsoft Lync and you can share your desktop like as Skype. You can download jitsi client program via https://jitsi.org/Main/Download official page. We will configure jitsi program at next sections.

In general plugin configurations under the Server tab. Also go to the Server -> Jitsi Videobridge section. Add the SIP username, password and SIP registration server and click save button:

In the Sessions -> Tools -> Send Message section you can send broadcast message to all users. As the following screen:

If we want to set SIP number for each user, before this we must add XMPP users to our system and then go to the Server -> Phone -> Add new Phone Mapping section and create SIP users(SIP and XMMP on the same server). For example we will add SIP number for existing namaz.bayramli XMPP user.

Then download Jitsi XMPP/SIP client program to your Windows machine and configure as follows (The official page: https://jitsi.org/Main/Download : File -> Add new account > XMPP -> XMPP Username Password -> Add

As you see XMPP user namaz.bayramli is ready:

Then click File -> Add contact and add the credentials as the screen, click Add button (Of course, user exists in our system):

The previous configuration we did for [emailprotected] and added to his user list [emailprotected]. At the end call from one client to another with audio/video and share your desktop:

This is desktop sharing:

And if you want to control other point from jitsi client, just select Enable desktop remote control checkbox. After that you can control other point as teamviewer.

For example if you want to use SIP configuration together XMPP, choose again Tools -> Options -> Add -> SIP and write SIP username and password (As the follows page). Just change domain name to yours:

As you see XMPP and SIP accounts is together:

Read more:
OpenFire Jitsi as Skype(desktop sharing) and Temviewer ...

An instant messenger client that offers support for text, audio and video based communication along with a wide range of helpful features

Jitsi is an application designed to offer you a simple and fun way in which you can keep in touch with the people in your life.

It offers you chat, video and audio communication, all of which are possible through a comprehensive and good looking graphic interface. It supports protocols like XMPP, Jabber, SIP, AIM/ICQ, Yahoo, Windows Live and others.

As is characteristic to nearly all IM applications, Jitsi offers you a main window that contains your contacts list from where you can perform various tasks. You can change your status, call a friend or send a file. Everything about the application is straightforward and user-friendly.

Contacts can be placed into custom groups, renamed and relocated at any time. You can edit their info and start a secure chat with them. With Jitsi its possible to make audio and video calls, perform desktop streaming, make audio conference calls and record them, as well as encrypt all your calls.

It proves itself to be a reliable means of communication for all kinds of environments, home, school and even business.

The level of security that Jitsi offers is one you should not overlook. It provides encrypted password storage, call authentification, call encryption and DNSSEC support.

As far as instant messaging goes, Jitsi offers you a lot of functions from the chat window. You can invite more people to join in, call a certain contact, initiate a video call, send a file, start secure chatting and of course insert various types of emoticons.

In case you are busy or away from the computer, Jitsi provides auto answer and call forwarding to any other accounts that are added to the application.

In closing, if youre looking for an environment that brings together all the major chatting platforms then you can try Jitsi.

Read more:
Jitsi Download - Softpedia

Atlassian just became a WebRTC superpower. Sort of.

The news is out. Yesterday, Atlassian announced through their HipChat blog that they acquired Blue Jimp:

Weve acquired Blue Jimp, the mastermind team behind the Jitsi Community.

The title itself speaks volumes of Atlassians needs and intent: HipChat acquires Blue Jimp & Jitsi.org

TechCrunch did the usual coverage of this, while the rest of the tech media was silent.

To understand what Blue Jimp did along with an analysis of how this affects the WebRTC ecosystem, Id recommend Chad Harts post.

I do like to reflect on a few issues with this acquisition though:

Why was this so important for HipChat? Heres a session Jonathan Nolen, HipChats Product Manager, gave at our Kranky Geek event on June 2014:

This is the 16th WebRTC acquisition, and the 3rd one this year. Interesting times.

Want to make the best decision on the right WebRTC platform for your company? Now you can! Check out my WebRTC PaaS report, written specifically to assist you with this task.

Go here to read the rest:
Why did Atlassian Acquire Jitsi? (Hint: WebRTC Multiparty ...

Jitsi (antes SIP Communicator) es una aplicacin de videoconferencia, VoIP, y mensajera instantnea para Windows, Linux y Mac OS X.

Es compatible con varios protocolos populares de mensajera instantnea y de telefona. Distribuido bajo los trminos de la GNU Lesser General Public License, Jitsi es software libre y de cdigo abierto.[1]

Jitsi soporta varios sistemas operativos, incluyendo Windows, as como sistemas de tipo UNIX, como Linux, Mac OS X y BSD. Tambin incluye:[2]

Los siguientes protocolos estn soportados por Jitsi:

Jitsi est escrito sobre todo en Java,[3] que ayuda a reutilizar la mayor parte del cdigo en los distintos sistemas operativos que trabaja. El proyecto tambin utiliza cdigo nativo para la ejecucin de tareas especficas de plataforma, como captura y procesado (renderizado) de audio/video, seleccin de direcciones IP, y acceso a sistemas de notificacin nativos popup como Growl (bramido).

El proyecto utiliza la aplicacin Apache Felix OSGi[4] para modularidad.

Entre otros, Jitsi utiliza la stack (pila) de protocolos JAIN-SIP para el soporte de SIP y la biblioteca Smack para XMPP.[5]

El hecho de que Jitsi maneja correctamente IPv6 es especialmente interesante para la comunicacin directa de PC a PC, por ejemplo, si ambas partes estn "atrapadas" detrs de routers NAT, pero se puede obtener una direccin IPv6 accesible a travs de un tnel-corredor.

La comunidad Jitsi ha completado tambin una aplicacin llamada ice4j.org, que se utiliza para proporcionar capacidades de NAT transversal, y ayudar a la transicin de IPv4 a IPv6.[6]

El trabajo en Jitsi (entonces SIP Communicator) comenz en 2003 en el marco de un proyecto estudiantil de Emil Ivov en la Universidad de Estrasburgo.[7] Fue lanzado originalmente como un videotelfono de ejemplo en el stack de JAIN-SIP y ms tarde empez a dar pasos como un proyecto independiente.[8]

Originalmente el proyecto se utiliz sobre todo como una herramienta de experimentacin debido a su apoyo a la IPv6.[9][10] A travs de los aos, segn el proyecto aada ms miembros colaboradores, se fue agregando soporte para ms protocolos, aparte de SIP.

Jitsi ha recibido apoyo de diversas instituciones como la Fundacin NLnet,[11][12] la Universidad de Estrasburgo y la regin de Alsacia[13] y ha participado en varias ocasiones en el verano de Google Code[14][15]

En 2009, Emil Ivov fund la empresa BlueJimp que ha empleado a algunos de los principales contribuyentes de Jitsi[16][17] con el fin de ofrecer apoyo profesional y servicios de desarrollo[18] relacionados con el proyecto.

El 11 de marzo de 2011, tras agregar soporte para audio/vdeo a travs de extensiones XMPP Jingle exitosamente, el proyecto fue renombrado a Jitsi puesto que ya no era "slo un comunicador SIP (SIP Communicator)".[19][20]

Continue reading here:
Jitsi - Wikipedia, la enciclopedia libre

Stefan Kelemen Stefan.Kelemen at 1und1.de Thu Aug 21 13:01:25 CEST 2014 That means lync is not supported by jitsi?Am Donnerstag, 21. August 2014, 18:18:54 schrieb Ingo Bauersachs:> On 2014-08-21 15:26, Stefan Kelemen wrote:> > Hi,> > > > i want to connect with our company lync server, but what i try, it> > doesnt work> > > > this message is the only thing i see with debug on -------> > impl.protocol.sip.ProtocolProviderServiceSipImpl.register().349 No> > address found for> > ProtocolProviderServiceSipImpl(skelemen at uc.mycompany.org (SIP)) -------> > > > Envirnment:> > The Lync Server is not the the uc.mycompany.org, its> > lync-pool01.Local.domain> > > Port is Standart 5060> > I can spaek per telnet with the lync Server> > Im using Jitsi 2.5.52.79 64Bit> > on a Debian Linux 64Bit> > > > Any Ideas> > Lync uses proprietary extensions that make it incompatible to any standard> SIP client. The most prominent are the authentication (NTLM/Kerberos) and> media transport over TCP.> > > regards> > Stefan> > Ingo> > > _______________________________________________> users mailing list> users at jitsi.org> Unsubscribe instructions and other list options:> http://lists.jitsi.org/mailman/listinfo/users-- "Seems to me, that you should "fummeln" on some more points."----Stefan KelemenSWIS OperatingSoftware Infrastructure Operations More information about the users mailing list

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[jitsi-users] SIP - Lync Connect deosnt work

Unbeknownst to many people, there are a growing number of free stand-alone VoIP clients, some of which arent half bad. Today Im going to be doing an in-depth look at one of these free downloadable clients, Jitsi, which is described as an audio/video Internet phone and instant messenger that supports some of the most popular VoIP and instant messaging protocols such as SIP, Jabber, AIM/ICQ, MSN, etc

The list is extensive, but it had me at SIP and Jabber.

Jitsi, which is written mostly in Java, is a free and open source VoIP, and instant messaging application for Windows, Mac, and Linux. Its currently in alpha. Stable releases come out every so often while nightly builds are released several times a day. When appropriate, users are automatically prompted to download and install the latest build (or you can just tell it to do this all without asking).

What separates this application from others like it is the inclusion of enterprise VoIP features such as attended and blind call transfer, call recording, call encryption, conferencing, and video calls.

This version of the application looks and feels great. The main UI is simple and clean, the pop-up call handling screen is easy to use, and the instant messaging feature is handled nicely. Jitsi certainly aims to accomplish a lot. While you can almost expect a few glitches here and there, it is certainly worth trying out.

[ Relevant Sidenote: This review was conducted on a Macbook Pro. ]

As usual, I am going to do a quick walk through of how to setup OnSIP with Jitsi. A lot of these steps apply no matter which VoIP provider youre using so I noncustomers will also find this useful. Youre going to need your user credentials. They can be found in your OnSIP admin portal under users. Here is an example of the fields you will need:

Setting Up VoIP Calling

Open up Jitsi and select +Add New Account under File. You should see a screen pop up that looks like this:

Select SIP as your choice from the options provided in the Network dropdown menu, and then hit Advanced in the lower left corner.

Youll be taken to another menu with 3 parts: Account, Connection, and Presence. Account is pretty self-explanatory. Under SIP id, youll want to input your entire SIP address. Password is your SIP password, and display name can be anything you want.

Next, in Connection, input your Proxy/Domain in the field marked Registrar, and your Auth Username into the field marked Authorization name. Youll want to uncheck Configure proxy automatically if it isnt already, and type sip.onsip.com into the field labeled Proxy if you are an OnSIP customer (Port 5060). Make sure that preferred transport is UDP and that the Keep alive method is Register.

In Presence, simply check Enable presence (SIMPLE) and leave everything else unchecked.

Hit the Next button. Youll be taken to a summary page where you can go over your settings one last time before you sign in.

Go into the Jitsi preferences. You should see a screen that looks something like the image above, with a list of all your active and inactive accounts. Select Audio and make sure that the codecs (or encodings) enabled are G722, PCMU, PCMA, and telephone-event.

Setting Up XMPP

Setting up IM is even easier. Here Ill show you how to get your my.OnSIP contacts in Jitsi. Once again, select +Add New Account under File. This time, youll want to select Jabber in the Network dropdown menu, and hit Advanced in the lower left corner. Youll be taken to another menu with 3 parts: Account, Connection, and Advanced. In Account, input your my.OnSIP login credentials. Skip the Connection section since you dont need to change anything there and uncheck the three options you see in Advanced (Use ICE, Auto discover STUN/TURN servers, and Use Jitsis STUN server in case no other servers are available). Click Next at the bottom of the menu, and then Sign In on the summary page that follows.

At Junction Networks, we put each of the phones we use through a multi-step interoperability test in which we apply ~30 test cases. An example of a test case would be the following:

Test phone calls phone B

B picks up

B puts Test phone on hold

B calls phone C

C picks up

B transfers test phone to C

Call must be transferred correctly to C. B must be released correctly after the transfer. When C picks up, audio must work in both ways between test phone and C. When test phone is on hold, there is no audio between it and phone B.

Build 3132 passed our test cases with no issues.

When I first installed Jitsi a couple of months ago, there was so much static that having an intelligible conversation was impossible. Whatever the issue was, it has since been patched and resolved.

Jitsi supports G.711 as well as the G.722 wideband codec. Narrowband calls sound about as good as a regular landline call.

High definition calls with the Jitsi sound absolutely fantastic. You can get HD VoIP calls as long as the person youre on the call is also using an HD capable device. I heavily recommend using a USB headset when making calls with a soft phone on your computer to get the optimum experience. You can pick up a good headset for less than $30.

For something that costs the end user nothing, Jitsi is a surprisingly good attempt at a unified communications client. I like to think of it as a bare-bones version of Microsoft Lync that doesnt cost me $700+ to setup, and $100 per download.

The main user interface of Jitsi looks a lot like any other IM client, except that you can have a dedicated section for voice contacts in your consolidated buddy list. Clicking on what looks like a small watch face will take you to your call history. You can conveniently redial from this screen. Right next to the watch face button is a search field, which will draw from both your contacts list and your call history. This field will also act as your dialer. Start typing in any number or SIP address, and a small green handset will appear that you can click to initiate the call.

Every contact in your buddy list and call history menus can be dragged and dropped into an ongoing call. What do I mean by that? With Jitsi, every call gets its own pop up window. Its here that youll find all of your call handling options: dialpad, create a conference call, hold, mute, record, video, desktop share, transfer, etc. Dragging and dropping people from your buddy list or call history menu into an ongoing call automatically creates a conference call. This seems to work without a hitch, and youre not just limited to a 3-way conference.

The image above shows the popup window you see during each call. You can have several calls going at once (simply call another number or SIP address using the dialer field in the main Jitsi UI and any active calls you have at the time will automatically be put on hold), and each one opens up a new window. Ill very briefly go over some of the functions of interest.

Youll notice that almost everything you can do with Jitsi is laid out in a row at the bottom. At the very left is a button that looks like an old school rotary dialer. This will append a numpad to the bottom of the window so that you can interact with attendant menus, etc. Next is your conference button. This brings up a window that you can use to invite multiple people to the call at the same time.

The next three buttons are self-explanatory: hold, mute, record (you can designate which file you want to save your recordings in the Advanced section of the application preferences).

Next is the button to turn on the video. Supported video compression formats include H.263 and H.264. Ill admit that I havent spent too much time testing out video calls on Jitsi, but the few video calls I have done (on Wifi, with just the built-in iSight camera on my Macbook and H.264 selected) were better than I was expecting. No experience-ruining frame rate or picture resolution issues here. I did try doing a video call with a coworker on her Counterpath soft phone and we werent able to get it working, despite the fact that they were using the same codec. We will do more testing and Ill update this review with our findings. Also keep in mind that a lot of factors will affect the quality of your video calls, and many of the problems you or I experience may have very little to do with the application. We plan to include video calling cases as part of JN interoperability test in the near future for applicable user agents.

According to the Jitsi development roadmap, there are tentative plans to implement multi-party video conferencing in Q1 2011.

Finally, Jitsi users can easily conduct blind and attended transfers. If only one call is active, clicking on the transfer button brings up a window where you can quickly input the transfer destination and send the caller on his/her way. If you have multiple calls active, clicking on the transfer button will open up a dropdown menu that includes all your active calls so that you can quickly conduct an attended transfer. Of course you can also choose to transfer to another number as well.

Now lets talk about some of the stuff that doesnt work quite as well.

If youre a my.OnSIP user, then you might be used to having the ability to click-to-dial and IM the same contact. You dont really get the same experience with Jitsi. My.OnSIP uses XMMP for IM and OnSIP uses SIP for voice, which means that youll have to have two separate accounts, and two separate contact lists for the same group of people. It can get especially confusing if the two types of contacts for one person look exactly the same. Long story short: Remember to use your SIP account for calling and your Jabber (XMMP) account for IM.

Adding phone numbers to the voice contacts could be better streamlined. Here is what the add contact form looks like:

Youll notice that you only get to specify the contact name. It actually works fine if youre adding a SIP address. If I type jondoe@example.onsip.com into the contact name field, Jitsi will know to use that as the SIP address, and will even cut off the domain in my contact list so that only jondoe is displayed. Adding actual telephone numbers is a little annoying since the contact name field is really the what to dial field. Sure you can go back after the contact is added and rename the number to a persons name but this seems like an unnecessary step.

Since Jitsi is a project that is literally updated several times every day, I dont think a Final Thoughts section is necessarily appropriate. The application has come a long way in a very short time, and there are big plans for the coming year. We expect a lot of updates and fine-tuning.

I would recommend giving this soft phone a download if you do not already have one on your computer, or if youre completely new to VoIP and SIP and just want a way to test out IP calling. Its free so what have you got to lose?

More here:
Jitsi (Build 3132)