IP PBX – BPO Call center – Artatel
In the VOIP world, minimum VoIP speeds are determined by the Codec your provider uses. Ideally, you have an exact match for the amount of compression 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 the expense of sound quality.
What is a codec? codec is an abbreviation of compression & decompression, namely a system of file compression and non-compression of files which changes the audio signal and compresses it into digital data form for transmission and then returns it back to the form of an audio signal like the data sent. Codec functions to save bandwidth on the network.

VoIP providers assume you have a broadband internet connection. DSL, which stands for Digital Subscriber Link, cannot reach 10 Mbps, while a regular cable connection can reach up to 30Kbs. Keep in mind that VoIP is full duplex , meaning VoIP can 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 high speeds, so you should consider upload speed as your benchmark. Also remember that 1Mbps = 1024 Kbps.
The Codec type works by encoding and decoding audio signals. The most commonly used type of Codec is called G.711, and uses 64 kilobits per second plus additional overhead for security, which can be applied to an IP bandwidth of 80-90Kbps. But, G.711 does not use compression. You can use other codecs that compress the data to use less bandwidth, with lower quality, but still suitable for most conversations.
To get VoIP calls whose quality is comparable to PSTN, you will use a codec whose type and type is called G.729. G.729 is a codec favored by VoIP providers as a “bandwidth saver,” so they can connect slower connections. G.729 compresses 64Kbps to just 8Kbs, a compression ratio of 8 to 1, but in practice you get bandwidth savings of about 3 or 4 to 1. The G.729 codec uses around 24-30Kbps.
If you're willing to sacrifice call quality, your provider can use a codec called G.723.1. The compression ratio can be as high as 12 to 1. G.723.1 compresses the signal to 5,6 or 6,4 kilobits per second, resulting in a bandwidth requirement of 16,27Kbps or 17,07Kbps. In Megabits per second, it is as low as 0,0159.

All of this assumes that you will dedicate 100% of your resources to VoIP. Luckily, the FCC has table practical about the minimum recommended speed for other activities. To do basic email and web browsing, and even stream radio, you only need half a Megabit per second.
There's good news on the horizon for the bandwidth conscious. The Xiph.org Foundation has released a royalty-free codec known as Opus, based in part on Skype's video and audio codec type, SILK. OPUS has a variable bit rate, and can reach G.729 quality to stereo iTunes music quality, depending on available bandwidth.
VoIP is capable of making higher quality calls than PSTN, but it is a flexible enough technology that it can work in 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 the available bandwidth.
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