SIP COMMUNICATOR - a JAIN-SIP Video Phone for the People!

Emil Ivov, Network Research Team (ULP Strasbourg)
 

1. What's in this package?

SIP COMMUNICATOR is a JAVA based SIP User Agent with audio/video capabilities built on top of the JAIN-SIP-1.1 API and the Java Media Framework. The phone is a pure JAIN-SIP application: it does not need any proprietary nist-sip classes in addition to those defined in JAIN-SIP 1.1, you can substitute the NIST-SIP stack by another JAIN-SIP compliant stack and plug the phone on top of it. This application was successfully tested against Microsoft Messenger, Ubiquity's Helmsman User Agent, and NIST's SIP proxy, and registrar server. So far, SipCommunicator has been tested to run under Windows XP and RedHat 8.0 Kernel-2.5.65. Note that there is an RTP problem with 2.4.x linux kernels and sessions are often dropped.

2. How to build SIP COMMUNICATOR

You need to have Apache Ant (http://ant.apache.org/) installed and in your path. Use ant clean and ant make to respectively clean old class files and build the project. Note that you should manually set the jmf property in the build.xml file.

3. Configuration

You could run SipCommunicator without any configuration as it assigns default values to all mandatory properties. To change any of these you could either use the configuration frame (on the menu bar, Call->Configure) or manually modify the SipCommunicator.properties file. The latter could be either in the nist-sip-1.2/src/net/java/sip/communicator directory or in $HOME/.sip-communicator. SipCommunicator uses the following properties:


4. Running SIP COMMUNICATOR


5. A brief description of SIP COMMUNICATOR's features

Currently SIP COMMUNICATOR is able to handle calls, participate in unicast audio/video sessions, register itself with a registrar server, and display incoming MESSAGE request. More features, (such as NOTIFY & SUBSCRIBE support) are planned for the near future (i.e. next couple of months). If you would like to see any particular features, currently non-implemented in SipCommunicator, we will be glad to discuss that and eventually add them.

To call someone you need to enter their SIP URL in the top left corner field of the SipCommunicator (the only editable field actually :) ) and press Dial. Examples of correct addresses are:

sip:emil_ivov@yahoo.com
emil_ivov@yahoo.com
sip:emil_ivov@130.79.90.142:5060

In other words - valid SIP URLs according rfc3261 with the only difference that the scheme is not mandatory (a default sip: prefix is added by SipCommunicator)

If this is not the first time you are running the phone you might find it handy to use the dial history (press the black arrow).

Use Answer and Hangup to respectively answer and end calls.

Your registration status as well as the address, port and transport you are currently using is shown in the SipCommunicator's status bar. 

SIP COMMUNICATOR works fine on both IPv6 and IPv4

The Menus

Well not much in here. Call->Configure let's you configure the application. Note that you should restart the application for the changes to take effect.Call->Exit exits (weird, isn't it). Tools->View Traces starts NIST's SIP TraceViewer. Help->About, shows my name and the name of the place where I work :).

6. About

 

Title: SIP COMMUNICATOR
Description: JAIN SIP Audio/Video Phone Application
Version 1.1
Organization: Network Research Team
LSIIT Laboratory
Louis Pasteur University, Strasbourg, France
Division Chief: Thomas Noel
Legal Notice This document and the accompanying software is provided under the terms of the Apache Software License
   
Author: Emil Ivov

 

7. Acknowledgements

I would like to thank a lot Mudumbai Ranganathan, for agreeing to include the SIP COMMUNICATOR in JAIN SIP's RI, I really appreciate the confidence, Ranga, and thank him as well (this time as a developer) for the great SIP stack that he and his team have provided "for the people".

A BIIIG  thank you goes as well to Thomas Noel (my boss :) ) first of all for providing me with everything I needed to work on the SIP COMMUNICATOR, secondly for letting me work on it for longer than initially previewed and most of all for letting me contribute it to the nist-sip community.