There’s a new generation and it won’t be denied. I have previously blogged on the exciting new FreeSwitch 1.0 release. I predicted it wouldn’t be long before others in the industry started seeing its worth and started making applications for it. Well, one major trend seems to be that the folks over at SipX Foundry, who make the very nice SipXecs, are a bit smitten with FreeSwitch, and what it can do for them. They’ve just created a conference server solution and next in line are an IVR and hints about a voicemail system based on FreeSwitch. They now have that feature/application server that they needed to complete their system. I expect to see the SipXecs project to use FreeSwitch more often.
Asterisk is starting to look like a first generation IP PBX. It is apparent now that its dominance in the FOSS VoIP market may be seeing its first real challenges by real players. Asterisk is going to have to start from scratch. They wouldn’t listen to their community, and may now pay the price for that hubris. Can they fix an inherently flawed design? Time will tell.
When I started with Asterisk, the learning curve was awfully steep. I didn’t know that much about telephony, and when I started with FreeSwitch I had to learn an entire new syntax and way of doing things (see -> steep learning curve). A lot of the people that are hanging on to Asterisk are doing so because they really don’t want to have to relearn something all over again. They are comfortable. Others have gained status in a community and don’t relish the idea of having to start at the bottom of the food chain again as noobs. They are comfortable. Asterisk gave me my start with VoIP, and I will always have place for it in my heart. But, I also realize that times and technology change, and so must I. Asterisk isn’t going anywhere anytime soon, but the writing is on the wall, and it reads “Niche Player.”
We here at VoIP Supply will have to react as the market changes or face the very real possibility that our competitors will do so first, and become the market leaders in doing so. To date, VoIP penetration has been limited by scalability. With new systems offering greater capacity the opportunity for growth is certainly there. The enterprise market that was previously cautious regarding VoIP may be a bit more open now. My point is not knocking Asterisk, simply categorizing it for what it is, a decent solution for the SMB/SOHO market. But, as my boss put it: “given the choice between a Mercedes and a Ford for the same price, I’ll take the Mercedes.” Others may feel the same.
UPDATE: The FreeSwitch group is getting ready to release version 1.0.1 soon that will improve the code’s stability and add Automatic Speech Recognition (ASR) and Text To Speech (TTS) to the code.
A little background on what others think of SipXecs is also in order. From the hyper-connected enterprise blog on TMCNet: “Asterisk may be older but sipXecs is better” (a Nortel Guy).
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Not sure what is so difficult with our spelling, but the project is called sipXecs and it is from SIPfoundry.
Hi,
That was a good article elevating sipxecs. But I want to know the real difference between freeswitch and sipfoundry. If you can help me understand this, I would be very grateful.
with regards,
suresh.
Hi Kevin,
I've got my book on sipXecs version 4.0 out now. Should help those new to sipXecs get going with the project.
https://www.packtpub.com/networking-and-servers/building-enterprise-ready-telephony-systems-sipxecs-40
Freeswitch is definitely enhancing the back-end functionality of the system.
Thanks,
Mike
I can tell you that sipXecs 4.x is a great product for a first timer of VoIP; however, the book on sipXecs 4.x above is definitely NOT the source to go by. It is just a rehash of what is already printed on the screens and what's on the wikis. Don't waste your money on the book and instead look at the wikis.