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Home News VoIP Free Your Phone Calls With FreeSWITCH!
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Free Your Phone Calls With FreeSWITCH!

Since SIP became the commonly used protocol to handle phone calls and other communication and Internet connects almost all households with a reasonable QoS, I ask myself: What is actually the role of the traditional telephone network operators? Everyone can become SIP service provider! Servers and bandwidth are cheap.
Well they have one important asset: Well known and globally coordinated and routed E.164 telephone numbers plus interconnection between all telephony exchanges.

The traditional voice service operators could close their business immediately, if everyone would use Skype and Skype terminals would be used and available everywhere. Or if everyone used Google voice. Or if everyone used "Line", or xxx, or yyy. And that's the point. There are many alternatives and they all are islands. To be reachable I would have to maintain at least a handful of user IDs and different clients.

SIP seems to emerge as the single most used protocol to handle voice and other sessions over IP. There are plenty of clients, some mobile phones have build-in SIP clients and SIP desktop phones became cheap as well. SIP addressing allows email like URIs as well as using the traditional E.164 phone numbers. The question remains who will be the "man in the middle" and operates the SIP infrastructure? In theory this man in the middle would not even be needed for simple UA to UA calls. The user agents can find each other using the DNS. But in practise we want to have voice mail, terminal portability, need transcoding, want to conference, and more.

So plenty of SIP service providers pooped up. They don't need much, some servers and Internet connectivity. They don't have to worry about the last mile. But they want to make money, of course their effort must be paid. They live from the fragmentation of voice services. IP to IP calls are free, but calls to PSTN will be charged, naturally. As more and more users are going IP, the offer of free IP to IP calls can not be kept any more. Earlier many SIP service providers offered free inter-provider calls and allowed incoming and outgoing SIP calls. As the share of paid PSTN calls goes down, those operators become more restrictive. Actually with many there is no big difference if you use them or a traditional PSTN operator. Only the last mile is IP and they are a bit cheaper for this. Otherwise, normal PSTN E.164 numbers are used and inter-operator calls are charged, no matter if they go via PSTN or are pure IP.

Fair enough that SIP service providers earn money with their services. But I wanted to unleash the full potential if the Internet and SIP without any restrictions and free of charge!
Hosting already a couple of websites on a VPS, I wanted to utilise this server as a private SIP PBX as well. I wanted to be reachable over the Internet / SIP using my well known email address under my own domain, e.g. email: sp@example.com and voice: sip:sp@example.com! No operator is needed, voicemail, conferencing, multiple terminals with parallel alerting. This is all possible with a piece of SW on my already existing VPS! Furthermore calls have even a better quality than with other operators as wide band codecs like G.722 are supported.

I described the basic set-up here.

As not everyone would call me in the future under sip:sp@example.com, of I still have traditional telephone numbers. But even using those, the call could be free! If the originating side (PSTN switch or VoIP operator) is clever enough to check the enum databases, it would find my SIP address linked to E.164 numbers and the call could go free of charge via IP instead of via PSTN or PLMN. Unfortunately most operators don't use public enum servers. They have their own internal enum systems. They actually like to charge subscribers PSTN fees for terminating calls and receive termination fees for incoming calls. Therefore they have not much interest in a public enum system under subscriber control.
Although the enum check should happen in the PSTN exchange or be done by the SIP proxy (or IMS S-CSCF), some SIP clients do this already themselves!
One example is the Linphone. Lets say someone is Sipgate.de subscriber and uses the Linphone. He wants to call me on my mobile phone. Normally this call would go to Sipgate and Sipgate would terminate this call to Vodafone and charge about 10c per minute for this. Now Linphone finds out that there is a SIP address in the e164.arpa database for my mobile phone number. Instead of setting up the call with +49172......, the call will be set-up using sip:sp@example.com. The call reaches my FreeSWITCH softswitch on my VPS and both, my desktop phone at home and my SIP client on my mobile phone (using UMTS or WLAN) will ring in parallel. The call would be free and even have a better quality (HD codec), if the Internet QoS is sufficient. Unfortunately sipgate does not forward SIP URI. So some more intelligence on the client side would be needed to make this scenario possible and chose another SIP operator or route the call directly. But I am sure that we will see this developing soon.