Getting started with OpenSource VoIP
This is more like a Hello World sort of writeup but it defines and explains the purpose of this blog site. The two main reasons why I like to write this blog:
1 - Contribute back to internet from where I've learnt this all myself.
So to be a good resource for anyone looking for help.
2 - Keep documenting stuff for my own future use. I'm not good with remembering things and mostly come visit my blog and refresh memory.
It has been about 5 years since I started my first blog article on starting with VoIP @ SaevolGo Blogger. The purpose was to get any person to read my blog and he/she could just start doing any basic VoIP things by just reading it.
That page has about 50 or so different articles and people love reading them it has been mentioned at OpenSIPSs home page as well - which is a big thing for me. That also got me involved in publishing my first and only book on Citrix XenServer by PacktPublishing, again a big achievement for a small time blogger like me.
Most important of all I've been applauded A LOT by readers there and I feel encouraged to contribute more and more. I had a big discontinuity gap on my blog, about 2+ years...Software and Operating systems have updated since then and I think its a good idea to start from scratch again.
At the very basic I expect anyone reading this page to have installed Linux based operating systems once and know a few basic commands to navigate and check things.
This blog area is all about practical stuff, understanding what is something and how to get it done. This does not necessarily have to be production grade stuff but atleast works and again motive is to make anybody do this on their own, get some idea, and then translate it into their environment.
Here are the two major areas we will cover in future blog posts: 1- Media Servers We will start from installing VoIP telephony servers like Asterisk, FreeSWITCH, and maybe SEMS. Once installed we will familiarize with their different components, making some dialplans, defining users, and establishing calls.
Once the basics are covered I will try to cover some advance topics from both Asterisks and FreeSwitch like Asterisk Realtime, Call Queues, Conferences, ODBC interfacing, AGI & AMI. From FreeSWITCH will do writing call logic in LUA, Learning about ESL, Integrating FreeSWITCH with Databases using XML_CURL.
2- SIP Proxies
OpenSIPS and Kamailio are the two major names in this field and we will learn how to install them, then will pick one of the two and get some closure with writing some logic in it. Understanding SIP protocol is as important so the posts related to these will also include some SIP signalling diagrams as well so everyone knows what we are trying to achieve.
After learning how to control a SIP proxy we will then see how to integrate our Media Servers with a SIP proxy. A very basic example is to use OpenSIPS as a Load Balancer in front of multiple Media-Servers to achieve High Availability by fail-over as well as increasing capacity at the Media Servers layer.
Some advances topics may include using RTPproxy, using Radius with OpenSIPS, making some basic PBX functionality inside a SIP Proxy, use of different modules etc.
I hope I be more consistent and accurate in explaining the scenarios. Wish me Luck :)
P.S: (Promise to Self) Do not repeat anything form my old blog page http://saevolgo.blogspot.ca