Obtaining and setting up Jitsi to participate in the Hawaii IPv6 Task Force SIP conference.

Jitsi Web Site

This page provides guidance for obtaining and setting up Jitsi, a free SIP client for Windows, Mac OS, and Linux. You may have a SIP client that you prefer, and you should be able to sort out for yourself how to configure your client using these instructions. Several other SIP clients will have similar setups.

Download Jitsi for your OS

You can select an appropriate package for download from the Jitsi download page:      https://jitsi.org/index.php/Main/Download

Set Up Jitsi with your SIP Extension Info

The information below is intended to enable you to participate in the SIP conference call which will be part of an upcoming Hawaii IPv6 Task Force Meeting. See this page for information about upcoming meetings.

For this part, you will need a SIP extension on the uhnet.net SIP server. You can obtain a temporary extension through the web form at: http://asterix.uhnet.net/cgi-bin/signup.pl.

This extension will allow you to call other extensions on the same server, but it will not enable you to call anyone else. You may wish to exchange extension numbers with someone else in order to test your SIP client.

You are also responsible for your own sound gear. In order to hear audio, you must have speakers connected and working (if you have a laptop, they're probably bult in, and ready to go), and if you want to be heard in the conference, you will need a working microphone. Many laptops also come with built-in microphones.

When you start up Jitsi for the first time you will see the following panel:

The only protocol that needs to be enabled for our purposes is SIP. In the "Username:" field type the extension number (somenumber@uhnet.net) you received by email. In the "Password:" field, fill in the extension secret from the same email. Once you have filled in those two fields, click the "Sign In" button in the lower right corner.

Jitsi should start up and look like the picture below. In order to test, you can enter 9999 in the "enter name or number field, and then click on the green phone handset to the right. Assuming that you have speakers connected and the volume turned up, you should hear a message welcoming you to the conference.

Note that the aforementioned green phone shows up only after you start typing in the number field.

Mute Your Microphone, Please

Although Jitsi, and similar programs can do a decent jobs suppressing echos and noise, you may wish to be a good SIP citizen and mute your microphone when you don't intend to speak. This can be accomplished in the call window that pops up when you start a call. Click on the microphone icon to prevent your local noises from getting into the conference. If you do this, you will have be able to click again to un-mute, and talk in the conference. If your background noise or an echo disrupts the conference, a conference admin may need you mute you at the server, and you will not be able to speak.

Using headphones instead of speakers is a very simple, effective way to ensure that you are not the source of echoes.

You may also wish to adjust the volume by clicking on the headphones icon.

If you click the button on the bottom that looks like a "pause" button on a DVD player, your call will be put on hold, and sound will be muted on the speakers and the microphone.

The number for the meeting conference will be sent to all registrants on the day of the coming meeting.

We look forward to hearing you!

Footnote: although you can do person-to-person calls including video between extensions, you will not be able to use video during the conference. You can watch the video from the meeting room on Ustream.tv. See the meeting page for more information.

Ancillary Footnote: No instructions for other clients yet, but I have used Counterpath X-Lite and CSipSimplei for Android with success. (On CSipSimple, I used user=<number> server=asterix.uhnet.net, and enabled ICE. )