SIP Voip implementation For ECF

Abstract

Session Initialization Protocol(SIP) is an Application Layer protocol which supports many services such as VoIP, IM, Presence Information and etc. But currently it's widely used for applications based on Voice calls and Video Calls.Eclipse Communication Framework is consists of many protocol implementations such as XMPP, YaHoo, MSN and etc. And also it contactins providers developed based on the above protocol implementations. Hence my goal is to implement a provider for VoIP for ECF based on SIP protocol for ECF.

Deliverables

SIP Provider for ECF
RTP Provider for ECF
SIP Related tests
Relevant Documentations and User Guides.

Timeline

Time line for project Milestones is given below

Legend
Needs some research

Work in progress

Feature added

Milestone	Date	Planned/Completed/Progressing items
M1	June 24	Implementation of the SIP Provider
M2	July 4	Completing the tests for SIP Provider and fix bugs
M3	August 8	Implementation of RTP provider using FMJ
M4	August 10	Complete the tests for RTP Provider and fix bugs
M5	August 12	Integrating the SIP & RTP Providers and create a SIP softphone
M6	August14	Complete the tests for integrated system and fix bugs
M7	August 17	Create the JUnit tests for Complete SIP and RTP integareted SIP softphone
M8	April 2010	Implementing the ECF Call API for SIP softphone
M9	TBD	Implementing UI for SIP softphone
M10	October 2009	Contributing the SIP softphone to the ECF Codebase

Project Code Base

Currently the project code base is hosted at https://github.com/ECF/Call & Google code site.

Check https://github.com/ECF/Call site how to access the repository via git.

The Google Code is at code.google.com/p/voipimplementationforecfusingsip/

This code base is managed using Sub version version control system.

Code Base Access

Command-Line Access

Use this command to anonymously check out the latest project source code:

# Non-members may check out a read-only working copy anonymously over HTTP.
svn checkout http://voipimplementationforecfusingsip.googlecode.com/svn/trunk/ voipimplementationforecfusingsip-read-only

SVN GUI Based Access

Use the following URL to Access and checkout the Codebase anonymously using GUI based Sub version Clients such as TortoiseSVN

# Non-members may check out a read-only working copy anonymously over HTTP.
http://voipimplementationforecfusingsip.googlecode.com/svn/trunk/

Now this codebase is hosted at [https://github.com/ECF/Call] as well. New improvements and features will be added to this new repository.

Path is: call/sip/modules.

Running Test Cases

Since this is a VoIP provider, to test functinalities you need to have at least 2 participants. You may use a Remote Sip Softphone or a Sip Echo service as the remote participant.

I have created 5 sip accounts for ECF testers and following are the credentials for those sip accounts. You need these credentials to initiate calls, register your softphone and to recieve calls.

Account 1

Username: sip:2233375055@sip2sip.info

Password: 391hw952w9

Name: Eclipse ECF Sip Tester 1