A lot of people have asked me about my home network configuration and VoIP telephone setup, so I thought I'd illustrate it here:

In the illustration, all blue arrows represent SIP communication with the annotation on the arrow representing the link layer.  The black arrows carry non IP voice traffic, potentially over GSM/UMTS or the fixed line PSTN.

At the center of everything is of course the VoIP server. In my case I'm running Asterisk on there, along with a Jabber server and a mail configuration making my mailboxes available over IMAP. A plugin installed in my OpenFire Jabber server is able to identify whether I'm on the phone or not and set my IM status accordingly. If I'm logged in to my Jabber account, I also get an IM message with extra details about incoming calls. Voicemails are stored in my IMAP mailbox and I get notified via SMS.

The top left part of the diagram illustrates the many SIP DIDs which I have registered against my server. You can often get free DIDs from providers such as SIPGate, IPKall, etc. People can thus call a local DID number and contact me, no matter where I might happen to be. There are two other possible routes for incoming calls - the Sipura 2000 and the MV370 - but we'll look at those in more detail later.

I also have multiple outgoing channels over which I can place calls into the PSTN, shown in the top right of the diagram. Then there's pure VoIP calls which are originating and terminating in the VoIP world in the right-middle of the diagram. These might be calls from VoIP networks such as Gizmo, other SIP calls, and ENUM routed calls. You can use ENUM (RFC 3761) to identify whether PSTN numbers you're trying to call might have a VoIP equivalent. If a mapped number exists, one can bypass the PSTN and route the call completely over VoIP.

Looking at the basic networking at home, I have a 25MBit Cable Modem connection to the outside world. This is connected to a WRT54GL running OpenWRT. Since the WiFi point is in the living room, I have a WET54G WiFi bridge in my study which I use to bridge the WiFi connection back to an Ethernet connection, made available through a Gigabit switch to the rest of the equipment in my study. I’ve got several VoIP devices interacting in interesting ways. First of all, all my computers are running softphone software such as X-Lite or SJPhone. Fine, softphones work, and they're available whenever you're at your computer, but sometimes you just need a real phone unit which is independent of your computer. That's where the Avaya 4620SW comes in - it's a very solid phone with a lot of functionality and it works fairly well against my Asterisk server except for the fact that the MWI indicator, as well as several other features of the phone, only work when connected against an Avaya SIP server. The Sipura SPA 2000 is a SIP ATA which lets one bridge SIP VoIP and the PSTN. Thus, incoming calls on PSTN are routed over VoIP to my server, and my server can also route calls over VoIP to the PSTN line on the Sipura. It also has an FXO port so that you can plug any standard telephone into the Sipura and have it connect to both the VoIP and PSTN lines. The PORTech MV370 is a similar device except that you place a GSM SIM in it and it bridges SIP VoIP with the GSM/UMTS network.

Finally, there's my mobile phone - a Nokia N95. The N95 has WiFi networking capabilities, and comes with a native SIP stack built-in.  When I'm at home, my mobile is registered over WiFi as a standard extension to my SIP network.  If I'm not at home but if there might be a WiFi network I can log on to, I can place and receive calls over my VoIP server. The N95 is my favourite phone so far. I used to be a very firm Siemens follower starting with the Siemens S25, but having used the N95 for close to a year I really have come to love this phone.

This equipment configuration lends itself to a number of interesting scenarios:

• On an incoming call, all registered extensions ring allowing me to answer the phone on any device.
• I don't need a cordless phone at home - my mobile phone is already my cordless phone for home.
• I can place calls over my home telephone line even when I'm not at home
• Calls to my home numbers can be answered from anywhere my mobile phone is logged on to a WiFi network.
• Place and receive calls over a second SIM, and not have to worry with the hassle of constantly changing SIMs.  The reason for the second SIM here is due to a mobile communications company here in Switzerland called Lebara who provide very competitive international call rates whereas Orange provides a good deal for local calls. 

There's a lot of potential here, limited only by your imagination in how you hook everything up. 

Finally, it’s out: the Gizmo Project - an easy to use, SIP-based VoIP program for everyone. It’s like Skype, except it’s based on open standards. After downloading and installing Gizmo, the first thing I did was to plug in my Asterisk SIP address into it and see whether it would dial, and it did! Similarly, Gizmo users can be dialled directly by their Gizmo username@proxy01.sipphone.com, or as gizmonumber@proxy01.sipphone.com. It provides PSTN termination (for a fee), and you can get a DID (Direct Incoming Dialling) number (again for a fee). One nice though though: if you don’t want a dedicated DID number, you can give out a shared Access number, and your Gizmo number. Users can then call in to the Access number, and when prompted, enter your Gizmo number to contact you.

All the features that they have, plus the fact that it interoperates with other SIP networks, definitely gives Gizmo the potential to be a Skype-killer - let’s wait and see what happens.

SIP to the masses!

A user on Voxilla has posted a set of parameters which allow you to configure the Sipura SPA 3000 Adaptor to correspond to UK BT-style regionalisation. A very helpful thread.

Hot on the heels of Vonage complaining that an ISP has blocked it’s services, an article at TechWeb mentions the possibility of Costa Rica possibly criminalising VoIP, or rather, the Instituto Costarricense de Electricidad (ICE), who are a telecommunications monopoly in Costa Rica, find it not right that 20% of international calls made from Costa Rica are VoIP. So far, most discussions about VoIP agree on the fact that it is a data service that should not be regulated.

After doing some more research on the capabilities of the Sipura ATA adaptors and the HandyTone range, I found that both now offer the capability to connect to standard PSTN lines as well as the VoIP connection, meaning you can buy one of these adaptors, plug it in between your phone and your phone line, and plug the adaptor into your router, and you’ve got one phone that can do both PSTN calls as well as VoIP calls.

The HandyTone 486 had the concept of a ‘lifeline’ PSTN connection, which would only activate if the 486 did not have power, so that you could still make normal PSTN calls. Revision 2 of the HandyTone though has introduced the feature that prefixing the number you want to call with “*00″ will cause the adaptor to route the call over the PSTN network, in place of the VoIP network. The phone will of course ring on an incoming call on either of the two lines.

The newly introduced Sipura 3000 also has the same capability described above that the HandyTone 486 has, with an additional twist - it has a built-in gateway which converts calls to or from your PSTN line to a VoIP channel! This means that phone calls coming in via your PSTN network can now be routed either to X-Lite or Asterisk. Whee! Of course, it works the other way around too - make a call from X-Lite or Asterisk, and have it go out through your PSTN line! Another interesting feature of the Sipura 3000 is that you can call into your adaptor by calling your normal PSTN number, authenticate yourself with a PIN, and then route your call over your VoIP connection.

I’ve been told that the next version/revision of the HandyTone would also introduce some features which the Sipura 3000 now has.

The BBC has an article about Internet telephony, which compares Skype and Vonage. Yes, Vonage have quietly introduced their services in the UK. I’d initially thought to myself that if Vonage ever were to start operations in the UK that I’d sign up with them, but I think they’ll find it hard to match the value-for-money you get with an unlimited GossipTel account. This push to VoIP is surely a good thing, but it shouldn’t happen indiscriminately. KPMG has recently released a white paper discussing the risks of VoIP (PDF Article from KPMG). Yes, the risks. Picture for a moment, that all your phone calls are going to be routed in an open format across the very hostile Internet. If you want to do any phone-banking, your account codes and PINs are being transmitted over the Internet which anyone could sniff! Or simpler, less hostile possibilities: you lose electricity, your internet provider has a failure, whatever: you instantly lose your phoning capabilities - no more 911 or 112 or 999. Beware where you tread.

Many of my friends have recently discovered that I have gone SIP crazy. I’d known for quite some time that SIP existed, but had never really got around to trying it out. After reading a post and associated comments at Slashdot, I finally decided to go and see what all the hype was about. I signed up initially with GossipTel for their Free, pay-as-you-go package. With this, you also get a Public Switched Telephone Network (PSTN) number with an 0870 prefix that other people can use to call you. Coupled with a soft-phone such as X-Lite from X-Ten, this means that your computer will ring when people call you.

I’ve already been on Skype for some time and had been using its SkypeOut service to make calls to PSTN phones, but the quality you get with SIP seems to be significantly better than that of Skype. This superior phone quality, combined with the fact that GossipTel are offering a package where you pay £20 a month and have unlimited minutes to call 35 countries, convinced me to become a subscriber with them. For signing up for a package, you have the additional benefit of receiving a local-area PSTN number, so that people no longer pay more for calling on an 0870 number. So now I have an 0207 number for London that people can call me on. I’ve also subscribed for SIP gateway services in USA and Germany, and have corresponding PSTN numbers in both countries, which are routed (for FREE) to my standard GossipTel account. Since GossipTel automatically forwards your incoming call after a certain number of seconds to any other number you might provide, my standard BT phone now rings even if you dial my US or German phone number. Whee!

I’ve also started investigating the technical aspects of SIP, and have already set up Asterisk (which is open-source and runs on Linux) as my VoIP server. Imagine the possibilities you have in being able to call into your Linux server with any standard phone…. *mad gleam*!