SIP Servlets proxy performance vs Kamailio

So how do SIP Servlets based proxies compare to each other and how do they compare to Kamailio ?

I had a project recently in which I needed to verify how different (free) SIP Proxies compare to each other performance wise, and was a little bit surprised to find out that there is very little information out there on the Internet. I decided to test out a few..

What I tested was three different SIP Servlets implementations and Kamailio.

My test setup...

The machine on which I run the tests was a Ubuntu 12 running on a Parallels (Mac) virtual machine with 2 i5 2.60HGHz cores and 3GB of RAM.

SIP Proxies:

• Mobicents SIP Servlets ver. 3.0.564
• Sailfin ver. 1.5 (9.1.1)
• Cipango ver. 2.0.0
• Kamailio ver. 4.2.2

The tests were run with SIPP client against a SIPP server.

The SIP Servlets code I used was very straight forward (taken from Cipango's load test app):

...

   Proxy proxy = req.getProxy();

   proxy.setRecordRoute(true);

   proxy.setSupervised(true);

   proxy.proxyTo(req.getRequestURI());

...

Kamailio configuration was taken from the example that comes with the distribution. The only difference was that I have disabled accounting.

Testing was done with SIPP as UAC and UAS. The scenario was:
INVITE >
100 <
180 <
200 < 
BYE > 
200 <

Results...

Sailfin

Mobicents

Cipango

Kamailio

This is what htop shows with 2500 CPS SIP traffic on Kamailio.

Amongst the SIP Servlets in my setup the worst performance was with Sailfin, the best with Cipango. Mobicents was somewhere in the middle.

And as expected Kamailio outperformed SIP Servlets as a Proxy. I could probably get higher results with Kamailio with assigning more shared memory for the server (it run with 512 MB).

test

just testing

Agile in telecommunications environment

Have you tried applying Lean, Agile methodologies in a telco environment ?

Here is our (translated) presentation from Agile Warsaw where we outline some of the challenges, experiences and solutions we have had with Agile management in a pretty large telco project.

Presentation Agile Telco

View more presentations from dmielnik.

Have you had similar problems ? How did you approach them ?

Freeswitch performance

I was recently testing Freeswitch and the performance of mod_conference to see how many concurrent channels it can handle. Take a look at the results.

The test environment was an Intel quad core machine with 8 GB and FreeSWITCH Version 1.0 installed. The load was generated by SIPP running on a different machine.

On this standard server, and just a little bit of tweaking, with 2000 calls in 50 different conference rooms with each conference being recorded to a file - the call quality was still ok and cpu was 50% idle.

Here are the SIPP result screens





Each call lasted about 4 and a half minutes with more or less 10% of that time prerecorded alaw sound was streamed by SIPP at different intervals throughout the call. 2002 calls were connected at peak, with 2000 generted by SIPP and 2 calls were placed manually to test the call quality.

I would say the results are quite impressive, what do you think ?

End of Cash?

An excellent blog post by Tomi Ahonen. This is the first of a series on evolution of money from physical to electronic to mobile.

Interesting and informative article covering subjects like history of money, cheques, credit cards, mobile money, virtual money and various examples of mobile money initiatives around the world including how they approach the issue in South Korea.

Go ahead and read the article http://communities-dominate.blogs.com/brands/2010/10/end-of-cash-first-blog-in-a-series-examining-the-pending-doom-of-minted-coins-and-printed-banknotes.html.

Asterisk 1.8.0 released

Asterisk 1.8.0 was released 2 days ago. The first thing that came to my mind when I saw the main feature list was that it is catching up on Freeswitch.

As on http://www.asterisk.org/node/51454, the main features include

Secure RTP
IPv6 Support in the SIP channel driver
Connected Party Identification Support
Calendaring Integration
A new call logging system, Channel Event Logging (CEL)
Distributed Device State using Jabber/XMPP PubSub
Call Completion Supplementary Services support
Advice of Charge support

The complete list of new features can be found on http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup (supposibly, because I can't open the link)

The full changelog is here http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0

really cool demo by Twilio

Twilio steals the show at New York Tech Meetup with this live demo.

You usually tend to think twice, or more before giving a live demo because that's when it hurts when things go wrong - and that's when they usually do.

Check out the demo by Twilio developer to large tech audience.

Watch live streaming video from nytechmeetup at livestream.com