VoIP, or Voice over IP, has become very popular over the last few years, and most businesses we come in contact with who are considering purchasing a new phone system want VoIP in some form, or at least want to know that the system they buy can support it.
(If you’re new to VoIP, feel free to read my other post on the different types of VoIP. You can also check out our web site for more information on ESI Communications Server phone systems, which support VoIP.)
To be sure, VoIP gives tremendous advantages over traditional solutions, and we are very excited to offer it. It should be noted, though, that with the nature of VoIP, a little more planning is required, and there are even times where VoIP may not make sense in your installation.
In contrast to traditional telephones, which usually use a pair of dedicated wires to transmit voice, IP phones can use the same network connection that your computer uses. This is because in VoIP, your voice is broken up into many small bits (packets) at lightning speed and sent quickly to its destination over an IP network (such as your LAN), where the voice packets are reassembled and played back to the recipient. This VoIP technology offers several applications that traditional telephony can’t match, but it also has the potential of causing some issues if you’re not prepared.
Jitter
When voice packets travel over a network to their destination, some packets may take a different route than others (routers determine this based on what appears to be the most efficient route at the particular millisecond that each packet passes through). Because there can be some variability in the time it takes for a packet to reach its destination, this can create some issues. If a particular packet does not reach its destination in time to be assembled at the other end in the right order, or if it does not arrive at all, it is not part of the voice conversation, and the person at the other end will get missing chunks or “stuttering” in the conversation. Phone systems generally address jitter, which occurs in all IP-based conversations to a greater or lesser degree, with an internal jitter buffer, which helps to smooth things out. No phone system can cope with excessive jitter, however (the larger a jitter buffer is in order to cope with jitter, the more it contributes to latency; see below).
Latency
Latency is the average time it takes for IP voice packets to get from one end to the other on an IP network. The greater the latency, the longer it takes for you to hear what the other person is saying, and vice-versa. In VoIP, there is always some level of latency, which is not necessarily a problem. For most people, latency above 250 milliseconds will make for a noticeable delay and will start affecting the conversation (generally in the form of people talking over each other).
Packet Loss
As mentioned, the voice packets in a conversation can (and usually do) take different paths to their destination. Sometimes, a packet may reach an unintended “dead end” and not make it to its destination, or a router along the way will become overwhelmed with traffic and drop some packets. This is normal, and a few packets will almost always be dropped from a given conversation. If the percentage of packets being dropped approaches 1%, however, it will make a noticeable difference in the conversation. Packet loss above 2% may result in the conversation quality being unacceptable.
Echo
Echo happens all the time in telephony, in fact, almost constantly, though you almost never notice it. In simplified terms, echo tends to happen due to an impedance mismatch on phone lines. This is common. In traditional telephony, the reason we almost never notice it is because when your voice bounces back to you and you hear it, there is a delay of only a few milliseconds. Thus, instead of perceiving it as echo, you perceive it as if it is something called sidetone (this is when you hear yourself talk on the phone, which gives you the comfort that you’re not on a “dead” line). With IP communications, voice takes longer to get from point A to point B and back (because at each router along the way, there is a small delay as the router processes it). Because of this increased delay, any echo is more noticeable because there is a greater time between when you speak and when you hear your voice in your own ear. Most phone systems address echo through the use of echo cancelers. ESI uses sophisticated hardware-based echo cancelers in all its phone system products.
Jitter, latency, and packet loss can usually be addressed by obtaining a managed network switch and/or a good router (such as Cisco). This equipment offers something called QoS, or Quality of Service, which essentially looks at network traffic and determines which packets are voice traffic, and gives voice the highest priority. Even when using a Remote IP phone or ESI-Link (ESI’s site-to-site IP connectivity technology) over the public Internet, we find that, most often, it is the internal network where most issues lie if there is trouble. Having indicated that, voice quality is obviously dependent on the quality of the Internet connection as well. We have found in general that if using the public Internet to carry your IP traffic, T1 is generally the best way to go, followed by cable modem, followed by DSL. Satellite modems are NOT acceptable for VoIP traffic, nor are other forms of wireless Internet (such as microwave-based technologies) since they tend to be “bursty”. Of course, while it is a more expensive option, a point-to-point T1 connection would be the best guarantee of quality, and this is often the best choice for full site-to-site connectivity. Another good choice would be an MPLS circuit, which has QoS capabilities.
When determining in what fashion it makes sense to deploy IP, one should consider the extra time and effort it will take to ensure that things run smoothly, and compare it with the benefits you will receive. With local phones in the office, this can be a toss-up. This is because, all else being equal, IP phones generally cost more than their digital phone counterparts (although with ESI Communications Server phone systems, IP pricing is very competitive with its digital counterpart if you’re buying 20 phones or more). Also, as mentioned above, the IP phones’ voice quality and connectivity can be subject to network conditions. Keep in mind that if your network goes down for whatever reason, it will bring down the phones as well, which will certainly add to the stress of your IT person or team. If you have digital phones, on the other hand, they will continue to operate normally in the case of a LAN outage.
Many people tout using IP phones on their LAN because of the convenience of moving phones around without having to pay for a phone technician to visit your site. With ESI’s Communications Servers, this is now a moot point, as even digital phones can be swapped easily without a site visit or wiring change. Still, many IT professionals prefer working with IP phones, because they understand them and how they operate, and local IP phones are certainly a viable option.
Where IP applications tend to make the most sense are for Remote IP phones (for telecommuters or executives), IP “soft” phones (for “road warriors”), and multiple-site connectivity where each site has its own phone system. If you could benefit from any of these scenarios, it’s well worth the effort to take the plunge.
