When you establishes a VoIP system in your home or office, the codec used to transmit the voice signals will have a significant bearing on the quality of the call you can make. The codec determine the quality of the voice signal and the bandwidth require for the calls. If a codec requires too much bandwidth, the call can suffer from distortion.
Additionally, if the codec uses too little bandwidth for the call, the calls will sound low quality. Although many people prefers to use the default codec that comes with the phone system, it may not be the best codec if the organizations adds many lines to the system or if the internet connection dont have much bandwidth. The chart provided allow people to compare codecs side by side in making a decision for there organization.
How codecs and the network affect VoIP call quality
For example, narrowband codecs, such as G.711, produce calls that sound like a standard telephone but require a high amount of bandwidth. Newer codecs, such as Opus and EVS, scales the bitrate to accommodate the available connection to provide near-CD quality sound. Additionally, the Mean Opinion Score help people compare codecs from other manufacturer.
The Mean Opinion Score rates the quality of the audio at a scale between one and five. Once people understands this rating, they can make a decision on the best codec to use in their organization. Bandwidth is also a concern when establishing a VoIP system.
Each packet contain extra header information that take up bandwidth. This takes up approximately sixteen kilobits per second of bandwidth. This amount of data are constant, regardless of the codec used for the calls.
As such, a codec with a lower bitrate will use less bandwidth than a codec that requires more bandwidth. This difference in the amount of bandwidth becomes even more importently for organizations that make many calls at the same time. If the number of calls strain the available bandwidth from the internet connection, the VoIP system will lose audio quality or drop calls.
The differences in audio quality between the codecs can be seen in the different tiers of audio quality offer by the codecs on the chart. Narrowband codecs limit the frequency range of the voice signals. This can make the voice on the call sound thinly.
Wideband codecs, such as G.722, provide a wider frequency range to the callers, giving the voice calls more clarity. Full-band codecs provides the highest quality of audio because they can capture the entire range of human hearing. However, high quality of audio come with high data rates that can strain the network if the network cannot support such high data rates.
The quality of the network connection will have as significant an effect on the quality of the VoIP system as the codec will have on the quality of the calls. People must monitor the latency, jitter, and packet loss on the network. If the one-way delay in the network is above one hundred fifty milliseconds, people will talk over each other on the calls.
If the jitter in the network is above thirty milliseconds, the receiving device will hold packets in a buffer, adding more delay to the calls. If the packet loss on the network is above one percent, there will be gaps in the audio that people will hear on the calls. These network problem will occur regardless of the codec.
Thus, people can configure the quality of service settings on the router that connects the network to the internet. To protect the voice traffic on the network, there is some steps that can be taken. For example, the system can mark the packets with a high priority so that they is processed before other packets.
Additionally, the bandwidth for uploading files can be reserved for the voice traffic. The voice devices can also be located on a separate network segment from the computers so that large files are not download when voice traffic is active. Using wired connection for the voice devices will eliminate the jitter that comes from wireless connections.
Finally, a user can disable the SIP ALG setting on the router so that problems such as registration failure and one-way audio problems are prevented. The codec can be chosen according to the needs of the organization. If the available bandwidth is limited and there will be many calls at one time, a low-bitrate codec will be chosen, even if the audio quality are compromised.
Alternatively, if the organization want to provide callers with a high level of clarity in their conversations, then the person will choose a codec with a wideband or full-band. However, this require a higher bandwidth. The need for bandwidth can depend on how many people will be on the calls at one time, as well as how much of that bandwidth is already taken up by the rest of the network.