VoIP Codecs + Bandwidth = Your Call Quality

The minimum VoIP speed is determined by the Codecs that your provider uses. Ideally, you would have the right match for the amount of compression that your codec uses, and the amount of bandwidth you have. The most popular codec is called G711, which uses no compression at all. Other codecs use compression at a trade-off of sound quality.

VoIP providers assume you have a broadband Internet connection. DSL, which stands for Digital Subscriber Link, cannot reach 10 Mbps, while typical cable connections can reach up to 30Kbs. Keep in mind that VoIP is full-duplex, meaning that it can both receive and send data at the same time. ISPs emphasize high download speeds for streaming and downloading music and videos, but for VoIP, you need to send data at a high rate, so you have to consider the upload speed as your benchmark. Also remember that 1Mbps = 1024 Kbps.

Codecs work by encoding and decoding audio signals. The most commonly used codec is called G.711, and uses 64 kilobits per second plus additional overhead for safety, which can work out to an IP bandwidth of 80-90Kbps. But, G.711 uses no compression. You can use other codecs that compress data to use less bandwidth, at a lower quality, but still suitable for most conversations.

In order to have a VoIP call that is comparable in quality to the PSTN, you would be using the codec called G.729. G.729 is the codec favored by business VoIP providers as a “bandwidth saver,” so that they can connect slower connections. G.729 compresses 64Kbps into only 8Kbs, a compression ratio of 8 to 1, but in practice you get bandwidth savings of about 3 or 4 to 1. G.729 codecs use about 24-30Kbps.

If you are willing to sacrifice call quality, your provider may use a codec called G.723.1. The compression ratio can be as high as 12 to 1. G.723.1 compresses the signal down to either 5.6 or 6.4 kilobits per second, resulting in a bandwidth requirement of a lean 16.27Kbps or 17.07Kbps, respectively. In Megabits per second, that’s as low as 0.0159.

All of this assumes that you will dedicate 100% of your resources to VoIP. Fortunately, the FCC has a handy table of recommended minimum speeds for other activities. In order to do basic email and web browsing, and even streaming radio, you need only half a Megabit per second.

There is good news on the horizon for the bandwidth-conscious. The Xiph.org foundation has released a royalty-free codec known as Opus, based partly on Skype’s video and audio codec, SILK. OPUS has a variable bit rate, and can go as low as G.729 quality to as high as stereo iTunes music quality, depending on the available bandwidth.

VoIP is capable of higher quality calls than the PSTN, but it is a flexible enough technology that it can work under less than optimal conditions. Your provider may be able to manually switch between G.711 and another, lower-quality bandwidth, and in the future, the codec you use will automatically adjust to available bandwidth.