As shown below, each technology allocated around the voice codec solves latency, echoing, jittering and packet loss that are the main causes of voice-quality deterioration in VoIP. Other than issues concerning conventional narrowband voice, these technologies solve issues related to deterioration of sound quality when using broadband voice.
Figure: High-quality VoIP technologies allocated around the voice codec
First you will hear the voice of a conventional telephone. Next, you will hear the voice using wideband voice codec, rich in expression. Experience the difference!
Optimal adjustment of packet size according the transmission bandwidth
Much like long-distance calls years ago, voice latency throws off the tempo and deteriorates the quality of communication.
Since voice information is set by converting data to packets (by means of buffering), voice latency is unavoidable in VoIP communication.
Latency caused by converting voice into packets is determined by how much information is contained in a single packet.
Although smaller packets of voice information are more desirable from the viewpoint of shortening this latency, smaller packets consume more transmission bandwidth as each packet contains header information.
Since there is a trade-off between voice latency and consumed transmission bandwidth, the packet size is optimally adjusted according to the transmission bandwidth in the actual usage environment.
Know-how in actual real world usage environments is crucial for corresponding to the infinite number of operational conditions
Due to the above voice latency, echoing is noticeable and deteriorates the communication quality in VoIP communication.
Eradicating echoes is an important issue for maintaining voice quality in VoIP communication. OKI uses an echo canceller capable of eradicating only echoes and has improved and expanded other technologies developed by the company.
Standards as indicators of performance exist for echo cancellers such as ITU0T G.165 and G.168. However, compliance to these standards do not necessarily fulfill performance in the actual usage environment.
This is due to the fact that there are infinite number of operational conditions in real communication, depending on who is talking to whom, what the conversation is about and the tempo of conversation.
Examining all of these on a desk (as a standard) is simply impossible.
OKI has accumulated the know-how in the actual usage environment by early implementation of products in the market.
Natural reduction of unnecessary latency with unique latency recovery process
Real time transmission of packets is not guaranteed in VoIP communication. For this reason, jittering occurs between packets received by the receiving terminal.
This jittering disrupts continuity of voice signals received by terminals.
As a result, communication is deteriorated by such phenomenon as voice interruptions and voice skipping.
Although it is possible to solve this jittering by employing a buffer in the receiving terminal, voice latency occurs when the volume of data accumulates in this buffer.
OKI has developed a unique latency recovery process to solve this problem by reducing unnecessary latency without the loss of natural communication even in the event of major jittering.
Loss of voice information is unnoticeable by the human ears
Conversation in real time is one of the most important factors in bidirectional communication.
For this reason, UDP/IP that does not resend packets is normally used in VoIP communication.
In the case of UDP/IP, however, communication quality is deteriorated since packet loss occurring during transmission directly leads to loss of voice on the receiving terminal.
Although packet loss handling is normally a part of codec processing on the receiving terminal side, there are some codecs that have been standardized long ago that are not supported.
OKI's unique handling reduces deterioration of voice quality caused by packet loss even for such old codecs.
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