VoIP: On the Rise Despite Shortcomings
“With voice over IP, you can have long distance for less – even for free!” That is the Siren’s song being played by VoIP, the Internet’s newest promise. Millions of consumers with broadband connections are responding. And there’s no end in sight; it’s expected that 32 million Internet phone lines will be in use by 2009 (Source: Gartner).
Akin to the cellular phenomenon, consumers are rushing to Voice over Internet Protocol (VoIP) despite the fact that audio quality and reliability are not yet up to traditional landline telephony standards (see “I want my V-O-I-P” section for quality analysis). Downtime and quality aside, the value proposition of VoIP has clearly resonated with consumers. During 2005, U.S. subscriptions to VoIP calling plans, which cost as little as $20 to $25 per month for unlimited domestic long distance (LD), more than tripled, from 1.3 million to 4.5 million (Source: TeleGeography).
Seductive as the savings message may be to consumers, the lure of VoIP is even stronger for businesses – whose monthly LD bills are often in the hundreds or even thousands of dollars. Many of these businesses have exacerbated toll charges because they either pay for calls between branch offices or have call centers, which typically incur heavy line usage. For such firms, slashing LD expenses could literally drop thousands of dollars a month straight to the bottom line. By the same token, the more dependent a business is on telephone communications, the less it can afford to compromise on the quality of its phone connections. Poor audio quality can undermine productivity and customer satisfaction; dropped calls cost sales and money; a total telecom service outage might well be disastrous.
I Want My V-O-I-P
It’s important to approach VoIP with your eyes wide open – to understand the technology involved, the investment required, and an affordable migration path to VoIP that will deliver value to your business, both short- and long-term, without
compromising the quality of your communications. But first, let’s talk quality. A study by Internet performance monitor Keynote Systems found that consumer VoIP service reliability improved from 96.9 percent to 99.1 percent between June
2005 and January 2006. 99.1 may sound good, but actually it is significantly lower than the 99.999 percent reliability rating that people have grown used to with the plain old telephone system (POTS), also referred to as the public-switched telephone network (PSTN). To put these figures into perspective, 96.9 percent uptime actually
equates to over 22 hours of downtime per month and 99.1 percent uptime still equates to more than 6.5 hours of downtime a month. Compare those numbers to 99.999 percent uptime, which equates to just 26 seconds of downtime per month, and you can see the impact of a few decimal points!
Packets 101: What Makes VoIP Vulnerable
As most people understand, VoIP is telephone calling over the Internet. So, it may appear that switching from the POTS to VoIP is a fairly straightforward proposition for the average small or medium size business (SMB). In a typical SMB, all employees have PCs, which are connected together via a local area network (LAN). This LAN is then connected to a router/firewall which talks to the Internet via what is called a wide-area network, or WAN. So, if you put a VoIP-capable phone on each desk and plug it into your LAN, the call goes out your WAN and voilà – IP telephony, right? Wrong.
Many people don’t realize that the Internet was *not* originally built for telephone calling. Neither were most LANs – yes, this means you! Even if you have a great LAN, it is unlikely that your WAN, or the WAN of your Internet Service Provider (ISP) is ready to go. In fact, the Internet Protocol itself was not really designed for real-time communication of any sort – especially the streaming nature inherent in audio, video, or online gaming.
Let’s get geeky. IP-based networks divide information – for example, an email message or Web page – into thousands of small chunks of data called “packets”. Each packet has a “header” which contains both the sender’s and the receiver’s Internet address; this enables the packets, which are transmitted separately, to be reassembled at the receiving end. Think of packets like ants, each of which knows how to get back to their anthill, but doesn’t necessarily need to get back at the same time or by the same route. This ability for a packet to “choose its own route and time” is why data transmission over the Internet is so efficient.
See, when data packets carrying email or web pages arrive slowly, or out of order, it is usually not a big deal. So what if it takes you an extra second to download your email? Space out for a minute, watch the paperclip, you know the drill. But this method of “out of order” packet delivery spells disaster for real-time protocols that need packets to arrive at the right time, in the right order, all the time. Welcome to the stringent demands of transmitting real-time audio.
OK, let’s adjust our pocket protectors and really talk about this; during a VoIP call, speech is captured as analog information by a phone, then converted into digital information, compressed, and divided into packets for transmission. This whole process is relatively fast, easy and reliable. The potential problems happen at the receiving end, when the packets must be reassembled in the correct order, absolutely error-free, and reconverted from their digital form into a seamless audio stream – all in real time.
If there are any substantial glitches in transmission, a VoIP call “breaks up” like the reception from a distant radio station. As engineers say, the audio stream stutters. As the people on the call will tell you, it sounds like gibberish. Houston...we.. av...a..ro..bl...em. In the worst case, which is not all that rare, the entire call is dropped – that is, cut off – because the transmission becomes so overwhelmed by problems that the connection simply fails. Dropped calls can mean lost revenue and/or dissatisfied customers!
Is Your Network Ready for VoIP? Probably Not
Here’s a fact: the LANs and Internet connections (WANs) used by most SMBs are simply not ready to handle VoIP. The basic firewalls commonly used for security and virus protection often cause VoIP calls to break up. The low cost routers from the local computer store often don’t have the horsepower to drive quality VoIP calls. LANs can also become congested, especially when users are transferring large files across the internal network, such as when sending or receiving emails with large files attached, downloading documents, doing file backups or copying media files.
Of course, your VoIP network includes not only your LAN but also your WAN. Your WAN begins with your broadband modem and ends with your broadband Internet provider. Most people don’t understand that just having a broadband connection is not enough. You actually need a *high quality* connection to deliver the call quality you need to run a real business. Many SMBs connect to the Internet via DSL or cable, most often with inexpensive modems. While such connections work fine for web browsing and email, they are not designed to handle VoIP transmissions, much less the combination of voice and data.
Similarly, many of the WANs now in use by ISPs were designed and built before the advent of IP telephony. They weren’t originally designed to meet the demanding requirements of error-free, reliable VoIP transmission. In fact, most of them actually run on a business model designed for oversubscription, which results in frequent latency and jitter (see sidebar: “Into the Weeds: VoIP explained”). Even the ISPs that aren’t oversubscribed have not yet deployed true quality-of-service (QoS) technology to ensure that voice packets get priority over data packets across their networks.
As if ensuring you have a great LAN and WAN was not enough, you also have to choose a high quality VoIP provider (VSP or ITSP). [Is your acronym meter at full capacity yet?] Much like LD providers, which deliver long distance services over the PSTN, VSPs provide you with VoIP calling over the Internet. Like ISPs, not all VSPs are created equal in terms of network strength, proximity to the PSTN backbone and, of course, good old fashion customer service. The VSP industry is a new one, so be sure to choose carefully. Remember a great LAN and WAN don’t mean anything if you have a weak VSP!
VoIP without hype. What business need to know. Part 2.
As most people understand, VoIP is telephone calling over the Internet. So, it may appear that switching from the POTS to VoIP is a fairly straightforward proposition for the average small or medium size business (SMB). In a typical SMB, all employees have PCs, which are connected together via a local area network (LAN). This LAN is then connected to a router/firewall which talks to the Internet via what is called a wide-area network, or WAN. So, if you put a VoIP-capable phone on each desk and plug it into your LAN, the call goes out your WAN and voilà – IP telephony, right? Wrong.
Many people don’t realize that the Internet was *not* originally built for telephone calling. Neither were most LANs – yes, this means you! Even if you have a great LAN, it is unlikely that your WAN, or the WAN of your Internet Service Provider (ISP) is ready to go. In fact, the Internet Protocol itself was not really designed for real-time communication of any sort – especially the streaming nature inherent in audio, video, or online gaming.
Let’s get geeky. IP-based networks divide information – for example, an email message or Web page – into thousands of small chunks of data called “packets”. Each packet has a “header” which contains both the sender’s and the receiver’s Internet address; this enables the packets, which are transmitted separately, to be reassembled at the receiving end. Think of packets like ants, each of which knows how to get back to their anthill, but doesn’t necessarily need to get back at the same time or by the same route. This ability for a packet to “choose its own route and time” is why data transmission over the Internet is so efficient.
VoIP vs. IP Telephony:A moment of distinction
Recently, VoIP has become a catch-all buzzword. Yet, it is important to distinguish between VoIP, which is a digital transport vehicle for phone calls, and IP telephony, which is a digital phone system based on Internet standards. This is important, because businesses stand to benefit from both VoIP and IP telephony – in substantially different ways.
VoIP is a method of digitizing your voice so that it can be transmitted across the Internet to save money on toll charges. Whereas IP telephony is a way of digitizing your phone system so that it can leverage the Internet, your computer, and your other business software applications (CRM, CTI, Outlook) to increase productivity within your business. VoIP is actually a subset of IP telephony – look at it this way: VoIP is an “arrow in the quiver” of IP telephony.
Any IP telephony system will use VoIP as a way of transmitting voice, in some manner or another (SIP, Skinny, MGCP, H.323). But, an IP telephony system, such as an IP-PBX, goes far beyond cheap phone calls; it enhances business productivity by providing additional features that weren’t available or affordable with legacy phone systems.
See, when data packets carrying email or web pages arrive slowly, or out of order, it is usually not a big deal. So what if it takes you an extra second to download your email? Space out for a minute, watch the paperclip, you know the drill. But this method of “out of order” packet delivery spells disaster for real-time protocols that need packets to arrive at the right time, in the right order, all the time. Welcome to the stringent demands of transmitting real-time audio.
OK, let’s adjust our pocket protectors and really talk about this; during a VoIP call, speech is captured as analog information by a phone, then converted into digital information, compressed, and divided into packets for transmission. This whole process is relatively fast, easy and reliable. The potential problems happen at the receiving end, when the packets must be reassembled in the correct order, absolutely error-free, and reconverted from their digital form into a seamless audio stream – all in real time.
If there are any substantial glitches in transmission, a VoIP call “breaks up” like the reception from a distant radio station. As engineers say, the audio stream stutters. As the people on the call will tell you, it sounds like gibberish. Houston...we.. av...a..ro..bl...em. In the worst case, which is not all that rare, the entire call is dropped – that is, cut off – because the transmission becomes so overwhelmed by problems that the connection simply fails. Dropped calls can mean lost revenue and/or dissatisfied customers!
Is Your Network Ready for VoIP? Probably Not
Here’s a fact: the LANs and Internet connections (WANs) used by most SMBs are simply not ready to handle VoIP. The basic firewalls commonly used for security and virus protection often cause VoIP calls to break up. The low cost routers from the local computer store often don’t have the horsepower to drive quality VoIP calls. LANs can also become congested, especially when users are transferring large files across the internal network, such as when sending or receiving emails with large files attached, downloading documents, doing file backups or copying media files.
Of course, your VoIP network includes not only your LAN but also your WAN. Your WAN begins with your broadband modem and ends with your broadband Internet provider. Most people don’t understand that just having a broadband connection is not enough. You actually need a *high quality* connection to deliver the call quality you need to run a real business. Many SMBs connect to the Internet via DSL or cable, most often with inexpensive modems. While such connections work fine for web browsing and email, they are not designed to handle VoIP transmissions, much less the combination of voice and data.
Into the Weeds: VoIP explained
Let’s talk protocol! This will help us understand why the real-time demands of VoIP put such a strain on the Internet and your LAN. As mentioned previously, the Internet was originally designed so that packets could arrive “out of order” and be reassembled by the client. The protocol most commonly used to transmit these packets is called TCP/IP.
Yet, TCP/IP is rarely used for VoIP packets because this protocol was not designed for real-time communications. Instead, VoIP often uses the UDP protocol. UDP tends to carry low overhead, making it a good choice for voice calls. But, the low overhead of UDP also makes it sensitive to network conditions. Therefore,VoIP-over-UDP can sound poor when it encounters any of following conditions:
Latency is the time it takes for a data packet to make a round trip between the sending and receiving location. When the average latency of a VoIP connection is above 200ms, call quality suffers. The best VoIP connections have latency under 80ms. Email and web access can gracefully handle latency of 400ms.
Jitter is the result of variance in latency between subsequent packets. For example, if you ping a network and get results such as 90ms, 92ms, 89ms, the network is jitter-free or, to be exact, it has jitter of 3ms (variance between 92ms and 89ms). But if your pings looked like: 50ms, 70ms, 190ms, then your network has jitter of 120ms. Jitter that exceeds 100ms degrades the quality of a VoIP call. Jitter under 50ms is gracefully handled by most IP phone systems.
Packet loss occurs most commonly when an Internet network is congested. Under such conditions, packets are often simply discarded. TCP/IP automatically retransmits these lost packets, but VoIP-over-UDP will not. Packet loss will create “stuttering”, or in extreme cases, “silent gaps” in your phone call.
Similarly, many of the WANs now in use by ISPs were designed and built before the advent of IP telephony. They weren’t originally designed to meet the demanding requirements of error-free, reliable VoIP transmission. In fact, most of them actually run on a business model designed for oversubscription, which results in frequent latency and jitter (see sidebar: “Into the Weeds: VoIP explained”). Even the ISPs that aren’t oversubscribed have not yet deployed true quality-of-service (QoS) technology to ensure that voice packets get priority over data packets across their networks.
As if ensuring you have a great LAN and WAN was not enough, you also have to choose a high quality VoIP provider (VSP or ITSP). [Is your acronym meter at full capacity yet?] Much like LD providers, which deliver long distance services over the PSTN, VSPs provide you with VoIP calling over the Internet. Like ISPs, not all VSPs are created equal in terms of network strength, proximity to the PSTN backbone and, of course, good old fashion customer service. The VSP industry is a new one, so be sure to choose carefully. Remember a great LAN and WAN don’t mean anything if you have a weak VSP!
VoIP without hype. What business need to know. Part 2.
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