Are you ready for “VoIP”?

VoIP (Voice over Internet Protocol) is a prevalent buzz word in the telecommunications industry today. VoIP includes technologies that use the Internet Protocol’s connections to exchange fax, voice and other forms of communication that were traditionally completed on Public-Switched Telephone Networks (PSTN).

There are several factors to be considered to successfully complete the Voice over Internet Protocol puzzle.

Quality of Service (QoS) is one of the most important factors for VoIP. The term refers to the perceived quality of speech and the methods used to provide good quality speech transmission.

QoS specifies rules about which traffic has priority on your network. Used correctly it is awesome, but it could render your router useless if done incorrectly. Be sure to choose a solution that offers you the critical control you require for successful VoIP deployment.

There are several factors that affect speech quality, and several mechanisms that can be used to ensure QoS:

If at any point the usage on the network exceeds the available bandwidth, the users will experience delay, also known as latency. In more traditional uses of an IP data network, the applications can deal with this latency. If a person is waiting for a web page to download, they will accept a certain amount of wait time. This is not so for voice traffic. Voice is a real-time application, which is sensitive to latency. If the round-trip voice latency becomes too long (250 ms, for example), the call quality would usually be considered to be poor. Another important thing to remember is that packets can get lost. IP is a best effort networking protocol. This means the network will try its best to get your information there, but there is no guarantee.

Delay is the time required for a signal to traverse the network. In a telephony context, end-to-end delay is the time required for a signal generated at the talker’s mouth to reach the listener’s ear. Therefore end-to-end delay is the sum of all the delays at the different network devices and across the network links through which voice traffic passes. The impact of latency on network throughput can be temporary (lasting a few seconds) or persistent (constant) depending on the source of the delays. Many factors may contribute to end-to-end delay. The buffering, queuing, and switching or routing delay of IP routers primarily determines IP network delay. Specifically, IP network delay is comprised of the following:

Packet capture delay is the time required to receive the entire packet before processing and forwarding it through the router. This delay is determined by the packet length and transmission speed. Using short packets over high-speed networks can easily shorten the delay but potentially decrease network efficiency. Packet delay variation (PDV) is the difference in end-to-end one-way delay between selected packets in a flow with any lost packets being ignored. The effect is sometimes referred to as jitter.

Delay variation is the difference in delay exhibited by different packets that are part of the same traffic flow. High frequency delay variation is known as jitter. Jitter is caused primarily by differences in queue wait times for consecutive packets in a flow, and is the most significant issue for QoS. Certain traffic types-especially real- time traffic such as voice, are very intolerant of jitter. Differences in packet arrival times cause choppiness in the voice.

All transport systems exhibit some jitter. As long as jitter falls within defined tolerances, it does not impact service quality. Excessive jitter can be overcome by buffering, but this increases delay, which can cause other problems. With intelligent discard mechanisms, IP telephony/VoIP systems will try to synchronize a communication flow by selective packet discard, in an effort to avoid the “walkie-talkie” phenomenon caused when two sides of a conversation have significant latency.

Some systems incorporate a Jitter Buffer to avoid these problems.

Switching/routing delay is the time the router takes to switch the packet. This time is needed to analyze the packet header, check the routing table, and route the packet to the output port. This delay depends on the architecture of the switches/routers and the size of the routing table.

Due to the statistical multiplexing nature of IP networks and to the asynchronous nature of packet arrivals, some queuing, thus delay, is required at the input and output ports of a packet switch. This delay is a function of the traffic load on a packet switch, the length of the packets and the statistical distribution over the ports.

Designing very large router and link capacities can reduce but not completely eliminate this delay.

IP is an unreliable protocol which means that in some circumstances packets of data can be discarded (dropped) by the network. This usually occurs when the network is particularly busy. Loss of multiple packets of a voice stream may cause an audible pop that will become annoying to the user. To maintain voice quality, packet loss should not exceed around 1% of all packets. Obviously this figure should be as close to 0% as possible.

A CODEC, which stands for coder-decoder, converts an audio signal (your voice) into compressed digital form for transmission (VoIP) and then back into an uncompressed audio signal for replay. It’s the essence of VoIP.

The CODEC used will affect the voice quality due to the different compression algorithms used, and the amount of bandwidth required. For example, on a low bandwidth WAN link, using a high bandwidth CODEC (such as G.711) may cause “choppy” speech as the WAN link will suffer from congestion. In this case, a lower bandwidth CODEC (such as G.729) may be more appropriate.

Available bandwidth has a major influence on voice quality in VoIP networks. Bandwidth is usually expressed in the number of bits per second (bps) that can be transmitted over a network link. The amount of bandwidth is usually limited by the service provider or the physical cables that are used for transmission.

So, now can you answer this: “Are you ready for Voice over Internet Protocol?” Let us help you with the details!