Project ideas for Google Summer of Code 2009

Its that time of the year when students across the globe gets an opportunity to participate in Google Summer of Code and get paid for executing the project that they always wanted to. This year too RedHat/JBoss has participated and has been accepted as Mentoring Organization.

Follow the RedHat page for GSoC 2009 here

There are many ideas that have been listed by JBoss here

The Mobicents Media Server team too has identified following Project Ideas where we are ready to mentor the students

Project Ideas

• Add Support for RTCP protocols to Mobicents Media Server
• Support for SRTP protocol to Mobicents Media Server
• The Automatic Speech Recognition engine to Mobicents Media Server. You can look at Sphinx-4 project which is open source

In addition the VoIP PBX project from Mobicents Sip Servlets listed on same page too is of interest to us. It uses Media Server for all of its media processing.

However these are not the only topics, if you have something in your mind and would like to execute as project on Mobicents Media Server, please feel free to drop us a line at amit dot bhayani at gmail dot com or oleg dot kulikoff at gmail dot com



All the best!

Mobicents Media Server 1.0.0.CR5 a step closer to GA (Stable) release

With the latest release of Mobicents Media Server 1.0.0.CR5 its just an inch away from finish line - GA (Stable) release. The CR4 was completely overhauled to achieve greater performance and CR5 is bug fix reported from CR4. In doing so there were number of modifications made such that the MSC API from Application Developers perspective doesn't change. The Announcement and IVR endpoint's in new MMS doesn't have transcoding capacity now. This was required to boost the performance. Also most of the time in Telco world the application developer knows precisely which codec will be used by end User Agent (UA) and hence use the pre-encoded audio file such that Audio Player reads this file, creates RTP packets and sends across to UA. If this is not the case than the application just has to front-end the Ann/IVR Endpoint with Packet Relay endpoint which still has transcoding capacity. The Announcement endpoint is capable of generating DTMF Tones. The Conference Endpoint is not modified and it too has transcoding capacity just like pre CR4. The Conference Endpoint has the DTMF detection and Audio Player such that Application can directly detect DTMF on Conf Endpoint or play the announcement.

We are very happy to announce that MMS can now easily achieve 100 concurrent announcements with transcoding. This test was done on my laptop which is 2GB Dell D620 mahine. We still have to do a test without transcoding and on real servers. If you want to test the performance of MMS in your environment please feel free to drop a line at amit dot bhayani at gmail dot com

For further details please look at http://groups.google.com/group/mobicents-public/browse_thread/thread/e6dd1ee52a357f53

Performance of Mobicents Media Server 1.0.0.CR3

Yesterday I did a performance test of Mobicent Media Server 1.0.0.CR3 on my laptop (Dell D620, Dual Core, 2GB RAM) with server and mms load test client running on same machine. The test was made for 50 concurrent announcements of an average length of announcement 3.74 Seconds. The codec used by UA test client was G711 A-law. The test lasted for 34487595 seconds or 9.58 hours (I stopped test and machine after this as I wanted to move to office ;) ). Number of announcements completed were 231202 out of which only 292 failed which means failure rate of 0.1262 %. The number of successful announcement were 231198. The load on CPU was on an average 60%. This indicates that max load on MMS can be further increased and hence more concurrent announcement

If we calculate the Erlang Number based on this statistics 3.74 (avg announcement in sec) * 231202 (completed announcement ) * 1000 (convert to milli sec ) / 34487595 = 25E. The call set-up and tear down takes time. The signaling used was MGCP stack on client/test tool side and MGCP RA + mgcp-controller-sbb on MMS server side.

These are definitely much better numbers compared to what MMS was capable of before CR3 release.

Performance is always relative to what you are trying to achieve. Its directly proportional to environment, application and fine tunning in addition to what one is trying to measure. Try the MMS performance in your environment, the load test tool is available at http://code.google.com/p/mobicents/source/browse/#svn/trunk/servers/media/test-suite

Soon we will execute load test on actual server and expecting the numbers to be much greater than reported here ... stay tunned!

JAIN SLEE / SIP Servlet becoming obvious choice

Recently one of the Indian ISV's MoBitE Technologies started providing consultancy on JAIN SLEE. If you visit Mobicents Partners pages, it already lists 4 companies around the globe who is working on Mobicents suite of products.

The Operators are getting aggressive on their plan of embarking on Next Generation Network and JAIN SLEE becomes the obvious choice. While Mobicents/RedHat provides the infrastructure, Mobicents Partners can provide the development help to Operators. Mobicents is increasing its range of products from core JAIN SLEE Server, SIP Servlet Server, Media Server, XDM Server and now the latest addition is Presence Server. With so many products and many standards it makes sense to involve the experts who understands the products better. The RedHat ISV Partnership program makes sure that these partners are up-to the mark and end customers can rely on them.

Review of book 'Internet Multimedia Communications Using SIP'

Last week I completed book Internet Multimedia Communications Using SIP from Rogelio Martinez Perea. This book is definitely recommended for beginners trying to learn SIP, even veterans can clarify their doubts using this book. The book points to all the RFC and JSR's where ever applicable and hence acts as good reference. Book refers to Mobicents as open source implementation of JAIN SLEE and NIST SIP stack as implementation of JAIN SIP. In fact all the examples are built using the JAIN SIP stack and Java Media Framework (JMF) for media. This shows how Open Source is making an impact in VoIP world :)

The part that I felt could be improved is the explanation of SIP Transaction Sub Layer.