IP telephony, or voice over IP, is a technology that allows oral conversations through Internet data packets. This type of technology works through IP telephony codecs. We explain everything about these codecs and how they work.   

 

What are IP telephony codecs?

 

IP telephony codecs are part of the “standards” established in IP telephony to optimize this technology. There are three basic standards; video or audio codecs, transport protocols and directory services. 

 This type of telephony is supported in a digital network, so there will necessarily be a digitalisation of the audio signal to transport oral messages. IP telephony codecs do this. 

In addition, codecs are divided into two categories; they can be with or without loss. Loss-free codecs retain all the information of the original transmission and therefore maintain the quality of the audio/video signal. However, lossy codecs, in order to achieve compactness, reduce the quality and consumption of broadband.  

Codecs are the basis of IP telephony ; they mean the switching of audio and video signals between the analogue and the digital.

 

How do they work?

 

To explain how IP telephony codecs work, we must first understand two concepts. Digitalisation and codification. Digitalisation converts analogue signals into digital signals. Codification is the key step, where we find the function of the voice codec. It translates the values for the next transmission.

Codecs use algorithms, and one of the most advanced is the “CS-ACELP” algorithm that allows you to organize the available broadband. 

 

How do they perform data conversion?

 

IP telephony codecs perform the conversion of analogue to digital information in the following manner. First, they take samples of the audio signal (a few thousand times per second). Then they convert it into digital information and decode it. The samples are later reconstructed and the information is then transmitted. This is why we say that codecs are dedicated to the encoding and decoding of information. 

 

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Types of IP telephony codecs

 

As we have already said, there are different types of codecs, as they depend on what type of algorithm they use. To know which type of codec suits you best, you must know the difference between them, as it can mean a major change in how the voice quality or the broadband you need will be. Some types are:

 

  • G.711 codec: Also known as alaw/ulaw, as these are the name of the two possible compression laws it can use. It is a simple codec that requires little computational overhead. 

 

  • G.729 codec : the most recommended and used for IP technology, this codec requires a low broadband and compresses the audio in pieces in ten milliseconds. 

 

  • G.723.1 codec : a codec that was originally developed for video conferencing but is now used for IP technology. It is basic and is used for low-broadband applications such as multimedia services. 

 

  • G.726 codec: based on ADPCM technology. It has one advantage; it can reduce the necessary broadband without increasing the computer load. 

 

  • G.729A codec : is an annex of the G.729 algorithm with a lower complexity that uses less computational capacity. It is also compatible with the G.729 codec and has good audio quality. 

 

Thanks to the IP telephony codecs, you can enjoy a company with internet telephony  You just need to choose well which codecs to use to get the maximum performance from your digital communications. 

In addition, WebRTC technology uses open, low-latency codecs. These codecs can provide a real-time communications protocol for a web page. WebRTC uses audio and video codecs. Within the audio codecs, we have for example :

  • OPUS Codec : an open source audio codec. It is very versatile and supports up to 255 audio channels. It is an ideal option for audio transmission over the internet and can also store audio files.

 

  • iSAC Codec : is a broadband audio codec. It was developed by Global IP Solutions but since 2011 it is part of the WebRTC open source project. There is also a less complex version of this codec for mobile phones and PDA’s, with an average bit rate of 40 Kbps.

 

  • Codec iLBC : is a narrowband codec, which was also developed by Global IP Solutions. It is currently part of the WebRTC open source project. It is a codec designed mainly for voice, although it is also useful as a codec for streaming audio. Its main feature is that it has a great capacity to preserve quality even if samples are lost.

 

Some WebRTC video codecs are :

  • H.264 Codec : is a high compression video codec, developed by Video CodingExpertsGroup (VCEG) and Moving Picture ExpertsGroup (MPEG). Its use was initially focused on low quality video for conferences or mobile applications. However, it was not useful for professional scenarios so extensions were programmed for this purpose. These extensions are called high profile and use a different coding principle (prediction, transformation, quantification, etc.)

 

  • Codec VP8: is a codec that Google released in 2009 as open source. It has a series of basic characteristics; for example, it requires low broadband. It also has a hardware that allows a wide variety of devices connected to the internet, since the client can have a powerful computer or a low-powered mobile phone; both devices can work.