It seems everywhere you turn, you’re bombarded with pitches to dump your local phone company and make all your calls via your broadband connection. (Actually, the cell phone folks also make a similar pitch, but that’s another article.)
The thing that makes this all possible is a technology that’s actually been around awhile called Voice over IP or VoIP. In this NTK series, I’ll attempt to turn down the hype and give you information intended to help you to understand what VoIP will and won’t do for you. I’ll also show you how to choose the right solutions, should you decide to take the plunge.
VoIP – also called Internet telephony or IP telephony – is the popular term for a number of technologies that enable voice to be carred via the Internet’s IP-based data network instead of networks dedicated to voice. This doesn’t necessarily mean that a VoIP call is the only time that your voice is carried in digital form from your mouth to the listener’s ear. Both local and long-distance “traditional” phone companies have used combinations of analog and digital networks to carry calls over the PSTN (Public Switched Telephone Network) for a number of years without making a big deal of it.
The move to digital by the telephone companies (telcos) was primarily cost driven for purposes of “pair gain” – the ability to squeeze more subscribers onto a single circuit. (Your telephone uses only two wires – a “pair” in telephone terms – that enable your phone to do its job.) These digitized voices were carried over telephone service providers’ private networks, however, never touching what we know as the public Internet.
The key differences with VoIP are that the digital conversion takes place before your voice leaves wherever you’re making the VoIP call from, and your phone lines (and local telephone company) aren’t involved – unless you’re using them for a DSL-based broadband connection. With VoIP, your digitzed voice usually travels over the public Internet, although this isn’t a VoIP requirement. Many companies still make significant revenue selling VoIP services that travel over private networks owned by them or their customers.
It’s interesting to note that the recent frenzy is actually VoIP’s second attempt at mass-market acceptance. The first consumer VoIP wave actually took place in early 2001 with Net2Phone and Linksys partnering on the BEFN2PS4. This was the first product from a major consumer networking product company to enable consumers to use a regular analog phone to make digital calls over the Internet.
Unfortunately, for a number of reasons, VoIP didn’t hit critical mass and the hype died down. Net2Phone is still in the VoIP business, but is definitely taking a lower-key approach this time, leaving the hype generation (and big advertising budgets required) to Vonage and others.
NOTE: As this article was written, the BEFN2PS4 was still up on Linksys’ website. But the company told me it has been discontinued, replaced by the RT31P2.
There are two main elements involved in a VoIP call:
- Analog-to-Digital (and vice-versa) conversion of voice (media)
- Call setup (signaling)
These are the same things involved in a traditional PSTN-based call – but the VoIP implementations are very different. In a traditional call, there’s no voice conversion involved for the media (at least not the subscriber end) and most of the heavy lifting is in the call setup.
In a VoIP call, the media portion involves not only the actual A-D and D-A conversion, but compression (to minimize the bandwidth required) and packetization (to get the voice data into IP form and onto the network). These elements are handled via a wide (and growing wider) range of stand-alone devices and combinations of software and hardware that I’ll get back to later. These devices are known by different names, but I’ll use the term VoIP phone when referring to them.
In most VoIP phones, the RTP (Real-time Transport Protocol) protocol is used to get the digitized voice packets into IP form for transmission over the network. Actually, RTP packets are enclosed in UDP packets which are sent and received via a range of ports. UDP is used because it has less overhead than TCP/IP and is more suited to the “real-time” delivery requirements of voice (vs. plain data).
If you want to get really picky about the whole thing, what actually happens is that two ports are used in each call direction. One is the RTP “media”, i.e. voice stream, the other an RTPC (Real-time Control Protocol) stream for Quality of Service (QoS) and media control.
No matter what you use for a VoIP phone, it will support some number of codecs (coder / decoder). VoIP codecs perform the same function as those used in DVD and digital music players – analog to digital and digital to analog conversion of the source material – but are optimized for the smaller bandwidth required for voice. As in video and music applications, various voice codecs have different advantages and disadvantages.
The most important codec-related criteria that you’ll need to worry about is whether your VoIP phone supports the codecs required by your VoIP service provider or used by the party that you’re calling. In most cases, VoIP phones support multiple codecs and automatically negotiate the best one to use, much like a dial-up connection figures out the best connection rate (and modem standard) to use.
Call setup is also different for VoIP, with the specifics depending on the VoIP protocol being used. For the home and small office VoIP applications that I’ll be focusing on, the two most often encountered protocols are H.323 and SIP (Session Initiation Protocol). H.323 is the older protocol and a standard from the telecom world maintained by the International Telecommunications Union (ITU). (If you’ve used Netmeeting, you’ve used H.323.) SIP is the up-and-coming newer kid on the standards block, born out of the Internet world and maintained by the Internet Engineering Task Force (IETF).
It’s not my goal to get into a religious battle of which standard is better (although you might want to read this Winnetmag article that describes what it says is Microsoft’s reasons for switching from using H.323 in NetMeeting to SIP in Windows Messenger). But it seems that mass-market VoIP products are opting for SIP compatibility over H.323, although some products support both.
One negative that SIP and H.323 share is that they have problems working through the NAT-based routers that are used to share your Internet connection. The root of the problem is the use of random multiple ports to carry the voice part of a VoIP connection. There’s a lot of work going on to eliminate, or at least reduce, this problem. But for now, you’ll need to pay attention to your VoIP phone or adapter’s installation instructions if you’re using a router.
As with VoIP phones, there is a wide range of hardware and software products for taking care of VoIP signaling duties. Although SIP phones (“end points” or “User Agents” in SIP jargon) can work on a peer-to-peer basis, most practical use will involve proxy and registrar network elements. These usually are combined into one piece of equipment commonly referred to as a SIP server, which can be a dedicated hardware device or computer running SIP server applications.
The H.323 world has its own terminology, of course, with user devices referred to as terminals (or endpoints) and the other essential piece called a gatekeeper. A gatekeeper is a logical entity and provides call control services for the terminals including address resolution, authorization, and authentication services, and call logging.
Optional H.323 elements are gateways and multipoint control units (MCU). As you can infer from the name, gateways translate call signaling and media transmission when terminals need to reach each other via other network types (Internet, PSTN, etc.) or segments of the same network. MCUs are perhaps the most specialized H.323 element and are required only when three or more H.323 terminals need to connect for a multipoint conference.
Before we move on, here are two other terms you’re sure to encounter (actually the definitions come from here) in your VoIP product explorations:
FXO (Foreign Exchange Office)
An FXO interface connects to the Public Switched Telephone Network (PSTN) central office and is the interface offered on a standard telephone.
FXS (Foreign Exchange Station)
An FXS interface connects directly to a standard telephone and supplies ring, voltage, and dial tone.
Physically, both these interfaces usually appear as standard RJ11 connectors that are appropriately marked Line (for FXO) and Phone (for FXS) or something similar.
Is VoIP for me?
Since the media hype engine for VoIP is turned up pretty high, I’m not going to spend any time repeating the pitches that you probably know by heart for considering VoIP. Instead, I’ll give you a few reasons below for not using VoIP.
Not the only way to lower long distance costs
Since long distance telephone service is available in so many other ways for very little money, if saving money on your long distance bills is your only justification for using VoIP, you may want to reconsider. I use a 3.5 cents / minute AT&T card from Sam’s Club that suits my usage just fine and don’t have to worry about lugging a VoIP adapter with me on business trips. Most cell phone plans bundle in more nationwide minutes that you can eat in a month, too.
Of course the tipping point in this equation is constantly changing, with the going monthly rate for an all-you-can-eat US and Canada plan from Vonage and AT&T at $25 as I write this in October 2004. If you do a lot of international calling, be sure to check your prospective providers’ rates because these calls may not be included in the basic monthly rate.
911 service gotchas
Whether or how well 911 emergency services work depends mostly on your VoIP service provider. So be sure you understand whether 911 services will work and whether there is an extra charge for them before you sign up. (This LightReading story has more details.) Note also that since your VoIP service depends on both power and Internet working it might not be available when you need it.
One thing that used to be true about POTS (Plain Old Telephone Service) phones, is that they worked when the power went out. Of course, since most of us now use cordless phones that need to be plugged in, this advantage has vanished unless you keep a batteryless phone around for use during power blackouts or have your phone on a UPS. But VoIP phones not only need your electric company to be on the job, but your broadband service, too. And getting both these to be 24 / 7 / 365 reliable may be more than your local companies are capable of.
Variable Voice Quality
This is another criteria that has changed over time since entire generations have grown up enduring the generally crappy (yes, that is a technical term) voice quality foisted on us by cell service providers. It’s a hopeful sign that users have not had their voice quality expectations lowered by VoIP companies…not yet at least… and expect the same quality as they’ve had on their POTS phones. Whether you get good quality depends on many network performance factors, some of which you can control (on your LAN) and many that you cannot (at your ISP and on the Internet).
Too many cooks
There are more companies involved in providing VoIP service than POTS and anyone who has wrestled with DSL problems knows the joy of being bounced from company to company when problems arise. If it’s important to you to get phone service problems resolved quickly, or you have had less than excellent experience dealing with your broadband’s tech support folks, then better pass on VoIP.
- Wrong number
Any ITSP that offers service connecting to PSTN phones will offer you a choice of phone numbers in various area codes. You might not, however, be able to get a number in your local calling area. So it’s possible that people calling you even from next door might have to pay for a long-distance call. Note also that your number won’t appear in local phone books (which isn’t necessarily a negative).
Consumer VoIP – Types of Service
So you’ve decided to forge ahead and try VoIP. What you do next depends on how comfortable you are with futzing with networking product selection and setup and how many lines you are looking to get. To start out simply, I’ll concentrate on VoIP products aimed at average not-especially-technical consumers. I’ll call this Consumer VoIP for lack of a better term.
The early days of Internet telephony were pretty much limited to tinkerers, teens and ‘tweens. These folks got a thrill using their computers equipped with a headset (or just microphone and speakers) suitable sound board and VoIP or “soft phone” software to directly communicate with callers with similar setups via the Internet (Figure 1). Call quality wasn’t great, but since the phone company meter wasn’t running, callers could talk as long as they wanted (or could stand the lousy quality).
Figure 1: Computer-to-Computer VoIP
This Free VoIP (also referred to as Computer-to-Computer) is still alive and well, but is now thought of more as enhanced instant messaging instead of telephone replacement. This method has also found a place in computer gaming, allowing players to verbally abuse each other while doing battle. The highest-profile exceptions to these uses are currently Skype (from the creators of KaZaA) and Earthlink, who are trying to revive the “free phone” angle. But even Skype is now pitching its pay-for SkypeOut service that connects to the good ol’ PSTN world. Earthlink isn’t offering VoIP / PSTN service yet, but says you can connect to anyone else using SIP-based services.
For now, most of the world still uses PSTN-based telephones, and people with VoIP service want to be able to call regular phone users and vice-versa. So all of the Consumer VoIP companies (Vonage, VoicePulse, Packet8) and telco consumer VoIP products (AT&T’s CallVantage, Verizon’s VoiceWing) support to / from PSTN calling (VoIP / PSTN service).
Tip: Terminology isn’t always consistent, but companies selling VoIP service are frequently referred to as Internet Telephony Service Providers (ITSP).
Most ITSP’s will require you to have a broadband connection to sign up. This isn’t because a VoIP connection takes up a lot of bandwidth, but more due to other things you may be doing on your broadband connection. Streaming video, long downloads and peer-to-peer file sharing, for example, can eat up enough bandwidth to adversely affect VoIP voice quality. Some suppliers such as Voiceglo (which uses H.323) and Packet8 (which uses SIP) let you use a dial-up connection, but recommend using broadband for “best performance”.
Consumer VoIP – Connection alternatives
All the “big name” Consumer VoIP offerings include the correct equipment you’ll need to use the service in handy packages that you can purchase at your favorite electronics retailer. These bundles are intended for self-installation by non-technical consumers, so include hardware that in many cases is already configured – at least for “simple” set-ups that don’t include multiple computers or routers.
In most cases the equipment is an FXS adapter, more commonly known as an Analog Telephone Adapter or ATA (Figure 2).
Figure 2: VoIP using ATA
An ATA is pretty much what it sounds like – a device that adapts or converts standard analog telephone equipment for use with VoIP. This means it will have one or more RJ11 (4 wire) telephone jacks to connect your analog phone stuff and an RJ45 (8 wire) Ethernet jack for connection to your LAN or broadband modem. ATAs allow any analog phone equipment to be used with VoIP service, including fax machines, answering machines and cordless phones. ATAs take care of both media and signaling VoIP duties, so must support the protocol(s) used by your ITSP.
In VoIP’s early days, the ATA supplied was typically a Cisco ATA-186 since less expensive alternatives weren’t easy to come by. But once the consumer networking product companies smelled the blood in the VoIP waters, they started pumping out products that have pushed the higher-priced alternatives aside.
A product that you’re sure to be seeing a lot will be the Linksys PAP2 (pictured at left), an ATA that provides connection for two standard phones or phone devices that Verizon recently announced will be supplied to its VoiceWing customers.
According to the folks over at Voxilla, the PAP2 is actually a repackaged Sipura SPA-2000 (more details here). Even though Linksys is mum on the issue, this Sipura press release (PDF) says their technology has been licensed by Linksys for the PAP2, RT31P2 router with built-in 2 port ATA that I mentioned earlier and WRT54GP2, basically an 11g wireless version of the RT31P2. As I was writing this article, AT&T announced that they’ll be bundling both Linksys VoIP routers with their CallVantage service.
Using an analog phone plus ATA isn’t the only way that ITSPs let you connect. Most also support soft phones like XTen’s X-PRO and X-Lite (pictured right), but may charge an additional monthly fee for their use. Soft phones require only a computer to run on (and a suitable headset or integrated microphone and speaker) and can be convenient for VoIP users on the go. There are even soft phones that run on PocketPCs, for an even more mobile VoIP solution.
Another point to consider is whether to get an ATA that includes an FXO interface. The FXO interface connects to your PSTN phone line and lets you use your phone with your normal telco connection. The feature is suggested if you aren’t ready to take the VoIP plunge entirely and will be keeping your regular phone connection. If your VoIP service goes down, or if you’d just rather use your telco line for a call, the built-in FXS – FXO converter will keep you connected. Low-cost ATA’s with FXO connectors aren’t easy to come by, however, and you’ll have to do your own set up, since none of the major ITSPs offer them as part of their bundles.
The ZyXEL P-2002L VoIP ATA with PSTN Lifeline automatically switches connected phones over to your normal telco connection when it is powered down or your VoIP service isn’t available. It also lets you dial a prefix to use the PSTN connection even when VoIP is active and automatically route 911 calls to your normal telco line.
Consumer VoIP’s Dirty Little Secret
Although there are hundreds of ITSP’s to choose from (thousands if you count the “Mom & Pop” resellers), using one of the heavily-advertised majors may give you a deja vu of dealing with your favorite cell phone company. Companies like Vonage, AT&T and Verizon are spending big bucks to promote their VoIP services and want you (actually your money) to stay with them.
Given that they’re the biggest (at least as I write this), Vonage comes under heavy fire from users for its practice of locking its ATAs so that they can only be used with Vonage’s service – just like cellular companies do with their phones! A suit about this practice has even been filed against Vonage by competitor SIPphone, alleging deceptive packaging and advertising.
The criticism the company has received for this practice has been somewhat mitigated by Vonage providing a $15 unlock service for its Cisco ATA-186’s, even though you’ll also have to forfeit a $40 termination fee to keep the ATA.
Tip: You’ll find many Google references to a reset-to-factory defaults procedure for Vonage ATA-186’s. But Vonage implemented password-protection for the unlock procedure around July 2003 that defeats the process. This page has a good summary of where things stand with unlocking an ATA-186.
Since it’s still early for AT&T CallVantage and Verizon’s VoiceWing, I haven’t been able to find complaints about their locking devices to their services. But the FAQ for both services list a charge for not returning the device when you discontinue service.
Note also that the larger consumer ITSPs generally won’t let you use your own VoIP hardware. But there are options such as BroadVoice’s Bring Your Own Device plan that let you use any compatible device (in this case SIP). And VoicePulse’s VoicePulse Connect provides commercial-grade origination and termination that supports both SIP and IAX and lets you use any compatible device. But it doesn’t have the fancy features offered by VoicePulse’s normal VoIP plans.
In Table 1, I’ve summarized some of the “gotchas” that you should be aware of with some of the larger consumer ITSPs.
|$60. Can avoid if equipment returned.
|$50. Refunded if equipment returned within 14 days.
|$59 non-refundable if termination within first 12 months.
|Yes, but only on additional accounts for full service.
Available for single account with VoicePulse Connect SIP / IAX origination / termination service
|None for most plans if equipment returned.
|$40. Refunded if equipment is returned.
|Key: BYOD = Bring Your Own Device MBG= Money Back Guarantee
Table 1: Consumer ITSP gotchas
You now have a basic understanding of VoIP and should be better able to choose a “consumer” VoIP service if that fits your needs. But there are many more aspects to the world of VoIP!
In Part 2, we’ll move from the world of neat and tidy Consumer VoIP to the more challenging unbundled and small business VoIP segments.
I’ll also cover what you need to know about implementing Quality of Service (QoS) and Class of Service (CoS) and show you how Open Source alternatives are alive and well in VoIP land!
To Explore Further
Concisely-written non-technical explanation of VoIP with good links.
I used this and its linked pages a lot in writing this article. Lots of good information via its well-organized linked articles.
Roberto Arcomano’s widely linked article is somewhat dated, but the Overview and Technical Info sections have clear, concise explanations of VoIP basics.
What is SIP?
Very understandable Network World article that will have you understanding SIP’s pieces and how they play together in no time.
Getting Started with SIP
Another Network World piece that describes how to get a feel for how SIP works and how to work it into your existing PSTN setup if you’re a small business or just play with it if you like to tinker.
VoIP Service Provider List
Community-maintained list of VoIP telephony providers and most likely to be up-to-date. Doesn’t consistently provide info on the protocols supported by each provider.
SIP service providers
Not a huge list, but at least you know they all support SIP!