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 email@example.com).
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.
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
Jitsi ZRTP FAQ¶
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.