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Before the Dawn of IP Telephony - Part 6
Originality and ingenuity for establishing voice quality
These contents translated a serialization article carried by ITPro IP telephony ONLINE published by Nikkei Business Publications, Inc. Jump to the original (Japanese).
(continued from previous page)
Basic concept of establishing voice quality
In order to establish voice quality, we assumed the three items below to be necessary.
(1) No voice intermittence
The main reason why sending voice using IP never became practical is due to intermittence at the beginning or in the middle of a voice communication. Such intermittence is caused by jittering, or the variation of time taken for data to reach the other party. This problem is unavoidable with IP. Obviously, voice is extremely difficult to hear with such intermittence and communication is virtually impossible. Hence, we first placed maximum weight on the development that guarantees continuity of communication.
(2) Suppress unnecessary latency
Latency causes difficulty in communication as much time is taken for voice to reach the other party. Hence, latency should be kept as short as possible. However, shortening latency causes the intermittence described in (1) to occur more often. There is the need to keep latency to a minimum under the favorable condition of no jittering, and guarantee continuance of communication even in the case jittering occurs.
(3) Completely eradicate echoes
Echoes seriously deteriorate the quality of communication. The need for completely suppressing echoes goes without saying. The difficult task is to realize this while maintaining (1) and (2).
Since (1) ~ (3) are closely related as shown above, they must be treated as a set. Realizing these three items as a single entity was our basic concept during the development of our first product, BS1100-VOICEHUB, and remains so today in our latest products.
Continued to next page: Buffer control...
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