Daily Archives: August 30, 2015

This setup guide is based on the Jitsi stable version 1.0-beta1-nightly.build.3593 on the Windows platform. Other platforms should have a similar infterface. This setup guide assumes you have properly installed the application on your operating system of choice and have gotten any audio issues resolved before attempting to configure Jitsi for use with Callcentric.

Help / Support: Jitsi documentation

***If this is not the first time running Jiitsi then on the main interface, please click on Tools > Options (on Mac OSX, please click on Jitsi > Preferences):

Then choose Add to add a new account. Select SIP from the dropdown list and continue to Step 2.

Username: This is either the default extension 1777MYCCID OR 1777MYCCIDEXT, where 1777MYCCID is the 1777 number assigned to you by Callcentric and EXT is the three digit extension you are trying to register this UA to.

For example: 17770001234101 would register to extension 101 on account 17770001234.

You cannot register to your account using only the extension number.

Password: Enter your extension SIP Password here. Your extension SIP password is the password you created for the extension you are trying to use. You may edit the SIP password you wish to use in by logging into your My Callcentric account and clicking on the Extension menu link and then modifying the appropriate extension.

After specifying your username/ password, click on the Sign in button on the bottom of the Sign in screen.

On the main interface, please click on Tools > Options (on Mac OSX, please click on Jitsi > Preferences). From there, please select your Callcentric profile and click on the EDIT button on the bottom of the Options page:

Keep alive method: REGISTER Keep alive interval: 60 Disable/uncheck: Enable support to encrypt calls

Click on the Next button, which should bring you to the Summary window.

Click on the Sign in button to finish.

PCMU/8000 telephone-event/8000

Once you have made these changes, please click on the Video tab and disable all video codecs.

After making the above changes, please close out the Options window, and go back to the main interface.

Go here to read the rest:
Jitsi Configuration and Review - Callcentric

jitsi.org/http://jitsi.org/logo/jitsi_logo.svg/Wikimedia Commons

About.com Rating

Visit Their Website

Jitsi is a Java-based communication tool that offers a lot of features and allows SIP-based calls. Jitsi supports voice and video calls and gives all the functionalities of an Instant Messaging software. It offers a nice interface and runs on many platforms. It also offers conference calls over SIP and allows you to connect to many other networks including Facebook, Google Talk, Windows Live, Yahoo!, AIM and ICQ.

Jitsi is a nice tool for integrating all your communication needs into one application. Jitsi is open source and is therefore free.

Pros

Review

Jitsi offers a very simple but nice-looking interface, with basic features and easy controls for configuring the tool and communication. Download and installation are straightforward and easy, and it is also easy to configure your SIP settings. You can use Jitsi with any SIP account.

Jitsi supports many IM protocols and works with many other networks, including Facebook, Google Talk, Windows Live, Yahoo!, AIM and ICQ. This makes it possible to call and contact your friends without having to change tool.

Jitsi is free and open source. Having a look the source code of tools like this is an interesting adventure for programmers who want to work on VoIP applications. Being Java-based, the application works on most operating systems.

Jitsi offers privacy and encryption for calls. It uses the ZRTP encryption, and is one of the rare SIP clients to offer something for security.

With Jitsi, you can use your computer and Internet connection to make free voice and video calls through SIP. Just get a SIP address and register with Jitsi. You can then communicate with your other friends using SIP or with people on the other networks mentioned above. You can also use Jitsi with Google Voice to call regular landline and mobile numbers.

Jitsi supports voice communication, video conferencing, chat, IM network, file transfer and desktop sharing.

The application is a little bit slow compared to others of the kind, due to it being Java-based and therefore interpreted. But if you have a powerful computer, you wont notice it. Also, you dont get to more detailed options where you can select codecs and other technical things. But you will be happy if you are not geeky enough for that.

Visit Their Website

Go here to see the original:
Jitsi SIP Softphone Review - About.com Tech

Earlier last week n8fr8 suspected something changed on the ostel.co server, due to many users emailing support specifically about Jitsi connectivity to ostel.co. The common question was why did it work a few weeks ago and now it doesnt anymore?

The tl;dr follows, skip to keyword CONCLUSION to hear only the punch line.

To support n8fr8s hypothesis, there was a small change to the server but I want convinced it effected anything since all my clients continued to work properly, including Jitsi. Obviously something had changed but none of us knew what it was. After some testing we discovered the problem was related to insecure connections from Jitsi to UDP port 5060 on ostel.co. Secure connections (on TCP port 5061) continued to work as expected.

To make matters more confusing, I could register and make calls with two different clients (CSipSimple and Linphone) on the same network (my home ISP, Verizon FiOS) using an insecure connection to ostel.co on UDP port 5060.

At this point I was like WTF?

I went back to the server, diffed all the configs, checked server versions, connected with every client I could find that would run on any of my computers. The only change was a Kamailio upgrade from 4.0.1 to 4.0.2. A minor point release. The problem with Jitsi remained. What could the server be doing to this poor client?

I did a packet trace on the ostel.co servers public network interface, filtered to dump packets only on UDP port 5060 that match my SIP username. I opened Jitsi and things got interesting. For the curious, heres the utility and options I used. If you are new to operating a SIP network, ngrep is an excellent tool for debugging.

ngrep -d eth0 -t -p -W byline foo port 5060

Ill include an excerpt (Ive included only the relevant headers for this issue) of the initial request from Jitsi. IP addresses and usernames have been changed to protect the innocent.

U 2013/07/19 22:17:34.920749 0.0.0.0:5060 -> 66.151.32.200:5060 REGISTER sip:ostel.co SIP/2.0. CSeq: 1 REGISTER. From: "foo" ;tag=1eb3467e. To: "foo" . Via: SIP/2.0/UDP 0.0.0.0:49152;branch=z9hG4bK-393535-2269e43afef0b312554eb419a8d0540e. User-Agent: Jitsi2.3.4752Linux. Contact: "foo" ;expires=600. .

# U 2013/07/19 22:17:34.921155 66.151.32.200:5060 -> 0.0.0.0:5060 SIP/2.0 401 Unauthorized. CSeq: 1 REGISTER. From: foo ;tag=1eb3467e. To: foo ;tag=e01f0de2cdfebbeefc5ff0c8eabbb8b3.2f1f. Via: SIP/2.0/UDP 0.0.0.0:49152;branch=z9hG4bK-393535-2269e43afef0b312554eb419a8d0540e;rport=5060. WWW-Authenticate: Digest realm=ostel.co, nonce=Uen0alHp8z4d6ePDl83RtMwARltAxzQu, qop=auth. Server: kamailio (4.0.2 (x86_64/linux)).

If you read the response, youll see Kamailio sent 401 Unauthorized. This is normal for SIP authentication. A second client request should follow it, which should contain an Authorization header with an md5 and a nonce. When Kamailio receives this request, checks the auth database and sends a 200 OK response, the client is authenticated.

The SIP dialog looks good but Jitsi continues not to register. The dialog flow is cut off after the 401 Unauthorized response. Its almost like something has blocked the response to the client.

Since I could register Linphone using the same account, I did the same trace for that client. Heres the excerpt.

U 2013/07/19 22:33:18.372770 0.0.0.0:42680 -> 66.151.32.200:5060 REGISTER sip:ostel.co SIP/2.0. Via: SIP/2.0/UDP 0.0.0.0:49153;rport;branch=z9hG4bK359459505. From: ;tag=142131416. To: . CSeq: 3 REGISTER. Contact: . User-Agent: LinphoneAndroid/2.1.2-1-g23b7fc0 (eXosip2/3.6.0). .

# U 2013/07/19 22:33:18.373112 66.151.32.200:5060 -> 0.0.0.0:42680 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 0.0.0.0:49153;rport=42680;branch=z9hG4bK359459505. From: ;tag=142131416. To: ;tag=e01f0de2cdfebbeefc5ff0c8eabbb8b3.4065. CSeq: 3 REGISTER. WWW-Authenticate: Digest realm=ostel.co, nonce=Uen4GlHp9u4FwHNY/uE1iQQNCfGHJiob, qop=auth. Server: kamailio (4.0.2 (x86_64/linux)).

This 401 Unauthorized response was received by the client and the follow up request with the Authorization header was sent with the correct digest. Linphone registered. I made a call. Everything worked fine. Indeed WTF?

I stared at these traces for a while to get a clue. Look again at the first line of the request from Jitsi. Youll see a timestamp followed by two IP:port pairs. Notice the port on the first IP is 5060 and the port on the second IP is also 5060. This means that the source port used by Jitsi on my home network is UDP port 5060. In order for a response to come back to Jitsi, it must enter my network on the same port it exited. Now read the top line of the response from Kamailio. Indeed, the server sent the response to UDP port 5060.

Now look at the same flow for Linphone. There is a very different source port in that dialog. In this case, Kamailio sent the response to UDP port 42680 and Linphone received it. Also notice the IP address used by Kamailio as the destination of the response is the same one in the dialog from Jitsi.

The question remained, why cant Jitsi get the same kind of SIP response on UDP port 5060? Why is Jitsi using a single source port for outgoing traffic anyway? That value can be dynamic. I configured Jitsi to use a different port for insecure SIP. It has an advanced configuration for SIP with the key SIP client port. I set this to 5062 (5061 is conventionally used for secure SIP traffic so I incremented by 2) and tried to register again.

SUCCESSSSSSSSSSSS!

To be thorough, I changed Jitsis SIP port again to a 5 digit number I randomly typed on my keyboard without looking.

SUCCESSSSSSSSSSSS!

So if Jitsi can register to Kamailio on any port other than UDP port 5060, WTF is going on? I had a suspicion. I tried one more test before I called it. I configured Jitsi to connect on TCP port 5060. It registered successfully. Now I know whats going on. I have a sad

CONCLUSION

My ISP, Verizon FiOS, has a firewall running somewhere upstream (it could be on the router they provided, I havent checked yet) that blocks incoming UDP traffic to port 5060. This probably falls under their TOS section which forbids running servers since Verizon provides voice services for an additional fee on top of data service, despite both running over the same fiber connection to my house. It seems like Verizon doesnt want their data-only customers to get in the way of that sweet cheddar delivery each month in exchange for phone service.

This sucks on two levels.

LEVEL 1

Why is my ISP censoring my incoming traffic when I have 5 mbps of incoming bandwidth? I assume the answer is because they can. *desolate frowny face*

LEVEL 2

Why doesnt Jitsi use a dynamic source port for SIP requests? I assume the answer is Jitsi is open source, why dont I change this and send a patch upstream?

Both levels are formidable challenges to overcome. Convincing Verizon to play nice on the Internet feels like a vanity project. Im writing that off. To make a change to the SIP stack in Jitsi is well within the area of the GP teams expertise, myself included but its not a trivial undertaking. Since this is a default configuration change there is probably a reason upstream devs made this choice so in addition to the programming work theres the work to convince the developers this would be a change worth a new release.

Since this is specific to Jitsi, Im going to follow up with the developers and see if I missed anything. Stay tuned for part two.

Thanks for listening. Stay safe!

Read the original:
Jitsi, ostel.co and ISP censorship | The Guardian Project

See also Jitsi provisioning via Tiki

Jitsi (previously SIP Communicator) is a cross-platform VOIP, videoconference, desktop sharing and chat client.

Jitsi is a core part of the Tiki Suite.

Protocols supported include SIP, XMPP/Jabber/Google Talk/Facebook chat, AIM/ICQ, Windows Live, Yahoo! Messenger and Bonjour.

Other features include: Call recording, Call & chat encryption, Noise suppression, Echo cancellation, File transfer, multi-user chat, Desktop streaming, Presence, Conference calls, Integration with Microsoft Outlook and Apple Address Book, Support for LDAP directories, Support for Google Contacts, On-line provisioning, Systray notifications, IPv6 support, Spell checker and many more.

License: LGPL http://jitsi.org/

Usage

To try out Jitsi via XMPP (with VOIP, videoconference, desktop sharing, etc.) you can use a Gmail account (which is also a XMPP account) or the jit.si service. Juts create an account and use that username at jit.si as jabber account in the Jitsi application.

If you are using Gmail, or Google for domains, use the Google Talk option in Jitsi. If you are using jit.si or a generic XMPP server, use the Jabber option in Jitsi.

A port for Android is in alpha

Jitsi is focused on a rich feature set, standards compliance and security.

There is the base package and 2 options. Many organizations will only need the base package, and they can add one or both options as they need them.

Advanced features (such as ZRTP encryption) require the client to support, but basic communication can be achieved by any XMPP client, including via a web interface.

This adds a web interface for Audio-Video-Chat collaboration with folks outside the team, by sending them a URL and using WebRTC

Alternatively, you can use BigBlueButton for this use case. BigBlueButton is more mature, very well integrated in Tiki and more focused towards online Learning. It is very easy to install, but it requires another server.

http://www.rtcquickstart.org/ICE-STUN-TURN-server-installation https://github.com/mozilla/togetherjs/issues/327

See the original post:
Jitsi | Tiki Suite