Lowest bandwidth voip codec

Codecs make large chunks of data smaller to reduce the internet bandwidth required to transmit that data. That compressed data then gets decompressed by a codec on the receiving end to make that data usable again. Lossy compression discards some audio data to compress the data as much as possible. Discarding a little bit of audio data enables a lossy compression codec to reduce audio data to one eighth or one tenth of the original size. However, even with lossy compression, you can still get very high quality VoIP call audio.

Those will get covered a little further on. For now, just understand that you want a VoIP codec that reduces your bandwidth requirements as much as possible, while retaining clear voice quality. A codec that reduces audio data to one fourteenth of the original size will sacrifice more audio quality than a codec that reduces the data to one eighth of the original size. This can have a huge negative impact on any business, where negotiation, sales, and customer service are handled over the phone.

Therefore, a weak VoIP codec could cost you big bucks. That way you can avoid using a codec with more compression than you actually need. Yes, you can estimate how much bandwidth you need, based on how many VoIP lines you have. You can use Kbps per VoIP line. Broadly, there are three common VoIP codecs.

The G. However, it also requires the most bandwidth. This codec requires at least 96Kbps of bandwidth per line. Because you can mistake silence for a disconnected call, CNG provides locally generated white noise so the call appears normally connected to both parties.

This does not have an effect on H. VAD on H. Although the voice samples are compressed by the Digital Signal Processor DSP and can vary in size based on the codec used, these headers are a constant 40 bytes in length.

When compared to the 20 bytes of voice samples in a default G. With cRTP, these headers can be compressed to two or four bytes. This compression offers significant VoIP bandwidth savings.

For example, a default G. This is a summary of the history. The exact heuristics used at present in order to detect RTP packets for compression are:. Skip to content Skip to search Skip to footer. Available Languages.

Download Options. Updated: April 13, Contents Introduction. Introduction This document explains voice codec bandwidth calculations and features to modify or conserve bandwidth when Voice over IP VoIP is used. For example, the G. Codec Sample Interval ms This is the sample interval at which the codec operates. With MOS, a wide range of listeners judge the quality of a voice sample on a scale of one bad to five excellent. The scores are averaged in order to provide the MOS for the codec.

Voice Payload Size Bytes The voice payload size represents the number of bytes or bits that are filled into a packet. The voice payload size must be a multiple of the codec sample size. For example, G. Voice Payload Size ms The voice payload size can also be represented in terms of the codec samples. Each frame contains eighty audio samples. When used for the purposes of VoIP frames can be sent in a packet. This is a licensed product, so the simplest way to leverage it is buying hardware that uses it, thus the fee will already be paid.

A common variant of it is G. On the MOS scale it just hits the 4. The patent for the G. There are 2 variants of this and both operate on 30 millisecond audio frames, but they operate of different algorithms. The first variant has a bitrate of 6. The second variant has a bitrate of 5. As far as encoded frames go, the first is 24 bytes long, and the latter This variation can be used freely, so it tends to be an incredibly common option in open source VoIP apps.

It operates on audio frames of 20 milliseconds, which is samples. This open source codec is patent free. It was designed to work with 8kHz, 16kHz, and 32kHz sampling rates.