Factors that affect VoIP call quality

* What can make VoIP call quality unacceptable?

Following up on the VoIP quality issues that we discussed in last week's newsletter, this week we'd like to take a deeper dive into some of the possible problems that can make VoIP call quality unacceptable.

First, let's start with one of likely suspects in the user equipment category that turns speech into a VoIP packet - the CODEC. The CODEC (short for coder/decoder) helps converts sound waves into digital packets so the packet can be transmitted across a digital transmission line and then be decoded on the other end back into sound. CODECs also compress the packet to gain maximum efficiency from the network. How well the CODEC converts speech to digital packets (and back again) is a possible factor that can affect call quality.

One of two available protocols is typically used by VoIP CODECS as defined by ITU-T standards G.7.11 and G.729. Using a Mean Opinion Score (MOS), with a 5.0 MOS as perfect quality, a G.7.i codec can achieve a 4.4 MOS and a G.729 CODEC can achieve a 4.2 MOS, according to VoIP quality experts at Brix Networks.

Another contributing factor can result from packet loss or discard somewhere between the calling parties. Because VoIP is highly compressed by the CODEC, packet discard can "throw away" a "lot of speech" as opposed to an uncompressed sound wave. The more highly compressed the voice packet, the greater the amount of conversation lost when a packet is discarded. Since packet discard are likely on a best Internet service, this factor is one of the most common causes of poor quality VoIP.

Latency is most frequently caused by the laws of physics governing the speed on light - the greater the distance between calling parties, the greater the latency. Note that both VoIP calls and traditional phone calls are bounded by latency and latency can be increased by the call routes chosen by both networks.

Finally, jitter can also affect speech quality; jitter is the variable latency between packets. Jitter is more common in IP-based speech because the path for voice packets across the network may not always follow the same route. The buffers commonly used in IP networks can also increase packet-induced jitter.

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Copyright © 2006 IDG Communications, Inc.

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