Voice over Internet Protocol (VoIP) is an efficient and cost-effective solution which allows individuals to make calls over an internet connection. It is a flexible system that converts your data, voice or video into digital signals. VoIP allows for clear communications, without reliance on large, expensive and often complex telecommunications equipment. It bypasses the traditional PSTN data transfer by converting the audio information into data packets.
VoIP systems utilize codecs to determine the compression that will take place based on available bandwidth and desired audio quality, among other factors.
VoIP codecs are essential to allow for the quick transfer of data, sound and video across an internet connection. They are what allows us to stream programs in real-time, and allows us to interact live with colleagues on a conference call. These codecs work by converting analog voice packets into digital packets and subsequently back to uncompressed audio.
There can be occasions when the audio quality of a VoIP call significantly degrades, making it difficult to understand the other party. The below table highlights some of the issues which may occur:
|Poor audio quality||Audio quality may be reduced if there is low availability of bandwidth or there is network instability. This is done to prevent the call from dropping.|
|Shaky audio||This issue occurs due to issues with the capacity of bandwidth. Small packets of data may fail to reach the intended end destination.|
|Latency||Delays are typically caused by the network becoming congested. Therefore Quality of Service (QoS) may be needed to improve the prioritization on a network.|
|Echo||Echoes may be a result of device or equipment problems. However, they can also be caused by latency on the network.|
Codecs are ultimately responsible for the audio quality of the call and a myriad of issues such as latency. Often there is a correlation between the codec’s performance and its compatibility with the devices utilized.
To improve the quality of your audio, it is possible to enable Quality of Service (QoS) to help with the prioritization of traffic over your connections. QoS allows your VoIP data to supersede other traffic of lesser prioritization.
Generally, bandwidth is the main constraint when it comes to providing the clear and crisp audio quality that callers expect today. The job of the codecs is to conserve bandwidth where possible while ensuring optimal sound quality and customer satisfaction.
Selecting the correct codec
When selecting which codec to use for your VoIP connection, it is essential to understand that, at their most basic level, codecs are a dual compressor and decompressor of data. They reduce the size of the data to quickly transfer it through the available bandwidth, and a codec on the receiving end then decompresses this to return it to a usable state. In fact, the word codec is a portmanteau of the words ‘compression’ and ‘decompression’.
Often, small pieces of data can get lost during this process and this is where the issues discussed emerge. If the chosen codec discards too great a level of audio, voice calls can quickly become distorted. Such issues can have a serious negative impact on business interactions, particularly in the case of negotiations or customer support calls. Codecs should be chosen based on the available bandwidth and the organization’s needs; this will help avoid the audio becoming overly compressed when it’s unnecessary.
Examples of codecs
Audio is often described as being in two bands. Narrowband includes audio frequencies ranging from 300 Hz to 3400Hz. Wideband covers a much broader range of frequencies (50 Hz – 7000 Hz), allowing for improved clarity and understanding.
VoIP codecs operate in a similar manner to help achieve the best quality conversations and experience for users. The bitrate is the volume of data transferred into audio; the higher the bitrate, the better the audio quality. The following table will detail a number of codecs that VoIP users may wish to utilize.
|G711||Many consider this to be the optimal codec to interface with PSTN as there is no digital compression. The compression ratio is 1:2 with a bitrate of 64kbit per second each way. Unlike many other codecs, there are also no licensing fees.|
|G722||This is a wideband codec that has come off patent and is now free for use. As above the transmission rate is 64kbit per second. It has been reported that there is little to no perceivable latency on this codec.|
|G729||G729 has a much lower bandwidth requirement, but generally provides acceptable audio quality. The audio is encoded into 10 millisecond frames with 80 audio samples in each. This means that the transmission rate is 8kbit per second. G729 would not be suitable for the transmission of music and other non-verbal audio.|
|Opus||The supported bitrate for Opus is between 6kbit and 510kbit per second. It is generally unmatched as a codec for music transmission, as well as its capabilities for speech recognition.|
All VoIP codecs operate in the same manner, compressing and decompressing audio. However, it is important that a business choose the appropriate codec for their requirements to provide customers with the best possible service. The wrong codec could result in the network facing bandwidth issues and problems with the transmitted audio, such as quality, latency, and echo.
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