If you're going to support voice over IP on your LAN, you need to know your VoIP basics. This isn't easy - confusion abounds when it comes to VoIP. Let's start with the obvious but often misunderstood: VoIP telephony is not the same as digital voice traffic, nor is it restricted to either local networks or wide-area networks like the Internet.

What exactly is VoIP? It is a type of digital voice traffic running over TCP/IP. You can do it over private local networks, public wide-area networks (the Internet), or even private wide-area networks. And it's not the only (digital) game in town - there are plenty of other types of digital voice traffic using other architectures. While telcos have been handling digital traffic for years, it wasn't until quite recently that they started running voice over IP.

The promises of VoIP are legion, but the reality is often somewhat more mundane. For carriers, VoIP offers the possibility of increased efficiency, new feature possibilities, and integrated voice, video, and data. For businesses and consumers, it offers the possibility of new carrier choices.

IP and TCP Basics

The reason for VoIP's growing popularity is simple: it rides on top of IP networks, which comprise both the backbone and the back alleys of the Internet. IP networks have successfully elbowed their way into corporate LANs, and most IT pros know how to work with and support IP applications.

The protocol combination generally used for VoIP is H.323, an ITU-T standard for multimedia videoconferencing on packet-switched networks. H.323 is a stack in its own right (a combination of other protocols) that governs terminals, gatekeepers, gateways, and MCUs involved in a transmission. H.323 should not be confused with its close cousins: H.324 is for point-to-point videoconferencing over POTS lines, and H.320 is for ISDN.

Meeting WAN Expectations

The major challenge for VoIP is latency. In addition to network traffic, VoIP's privacy requirements put an extra burden on quality of service.

Because IP networks are packet-switched (not circuit-switched like conventional voice networks), the individual packets constituting a voice "signal" can travel separate paths to a destination and can arrive for reassembly out of order. The efficiency and reliability gains of such networking were the goals of packet-switching in the first place, but they don't help out much when it comes to the quality of a VoIP conversation.

Variable routes and out-of-sequence packets might cut it for email and Web browsing. They even cut it for IP broadcast audio and video (thanks to buffers). But they just don't work for real-time full duplex voice - at least, not without some help. In an ordinary phone conversation, there's no time to wait for the retransmission of lost packets. The conversation must go on. Lost packets remain lost, and the conversation gets choppy as a result. Ten years ago, people might have put up with it for first-world international telephony. But today, we have grown accustomed to high QoS (quality of service) at low cost. Blame it on those old "pin drop" commercials.

Ironically, in the WAN arena, it is because we're used to the crystal-clear quality of digital calls across dedicated circuits that VoIP doesn't yet meet our expectations. Imagine if we were still calling long distance over amplified, noise-filtered high-latency analog circuits when VoIP came along. We would have probably rushed to even the earliest incarnations of VoIP.

However, long-distance VoIP has found a niche for itself in third-world markets, where telephone services are expensive and, in many cases, difficult to procure. In North America and Western Europe, telephony is so affordable and widely available that VoIP is hard-stretched to provide a price cut.

In markets where the price of telephony is high, a VoIP connection can be dramatically lower than traditional options. Most people erroneously think long-distance VoIP must run over the public Internet, but a number of long distance carriers are already providing service over privately managed VoIP networks.

The Problem with Connectionless IP Networks

The big problem with IP networks, and the Internet in particular, is lost packets and the resultant latency in the "voice signal" received at the other end. In a low-traffic environment, unassisted VoIP might work fine. However, once traffic heats up, voice conversations suffer while other applications do not.

Most applications are not as delay-sensitive as VoIP. Unlike other traffic types, a little delay equals a major annoyance in VoIP.

So, we prioritize. Voice can be treated as data, but it is a special type of data. As George Orwell might have written, all data is equal, but voice data is more equal than others. Not only must it stream (which means it must minimize lost packets) but it must stream in two directions simultaneously.

Like TCP/IP, the H.323 protocol is actually a stack of other protocols. Originally accepted in 1996, it was geared towards LANs and it was understood at the time that H.323 would not provide a guaranteed QoS. Lots of experimentation on different platforms (LAN, private WAN, public Internet) and in different applications (PC-to-PC, PC-to-phone, phone-to-PC) led to the adoption of version 2 in 1998.

The transport protocol used in H.323, RTP (Real Time Protocol), gives priority to delay-sensitive traffic like voice and video. This was a step in the right direction. However, while RTP attempts to control different traffic streams, it cannot actually guarantee on-time packet delivery.

For private bandwidth-managed networks, this need not be a barrier. However, it is surely a problem in unpredictable, congested network environments - namely, the public Internet.

That's where RSVP, or Resource Reservation Protocol, is supposed to come in. It essentially lets you reserve a user-to-user slice of bandwidth. For the duration of a voice conversation, RSVP creates a virtual circuit. The ability of H.323 v.2 client and network devices to support RSVP, or other protocols and techniques for managing QoS, is crucial for widespread adoption and for tweaking the utmost efficiency out of a network while still retaining quality.

The LANs First

A crucial component of any VoIP system running over a LAN is a gateway to the PSTN. A VoIP PBX can be installed with a gateway directly linking it to the PSTN so that any outgoing calls made on the PBX are immediately routed to the PSTN. Or a VoIP PBX can be connected to an IP telephony backbone. It, in turn, could connect to the PSTN to carry a long distance call. Or, it could carry the call itself if it can terminate at or near the destination local exchange.

Despite the difficulties, products are already available that let you expand a VoIP-based PBX to create voice VPN over the public Internet.

Hardware or Software

Options abound when it comes to VoIP products. The first VoIP products were all software, but they've grown up a bit since then. Though run-of-the-mill PC platform options are out there, robust business products are built on NT, Unix, or Solaris platforms.

There's also an increasing number of embedded offerings. This is one area where DSPs (digital signal processors) are expected to play a big role. Built from the ground up to handle analog information, DSPs will likely replace programmable hardware in VoIP client devices. On the back end, we're seeing more and more embedded products both for carriers and for individual office installations.

The Future

Eventually, carriers will implement VoIP all the way to the local loop, even in residential areas. Why? Because broadband Internet connections will finally do an end-run around local telephony loops. Demand for inexpensive broadband technologies and increases in public and private Internet backbone bandwidth will further fuel VoIP adoption.

The timeline for all this? It depends on what kind of infrastructure you've already got in place. If what you currently have works, and it's cheap, then there's less incentive to build. "For the next ten years," says Pierce, "we will be living in a hybrid world." For the time being, packetized and circuit-switched voice will likely coexist in the marketplace and even on LANs, for better or for worse.


      
                  
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