Why cells?
Consider a speech signal reduced to packets, and forced to share a link with bursty data traffic (traffic with some large data packets). No matter how small the speech packets could be made, they would always encounter full-size data packets, and under normal queuing conditions, might experience maximum queuing delays. That is why all packets, or "cells," should have the same small size. In addition the fixed cell structure means that ATM can be readily switched by hardware without the inherent delays introduced by software switched and routed frames.Thus, the designers of ATM utilized small data cells to reduce jitter (delay variance, in this case) in the multiplexing of data streams. Reduction of jitter (and also end-to-end round-trip delays) is particularly important when carrying voice traffic, because the conversion of digitized voice into an analogue audio signal is an inherently real-time process, and to do a good job, the decoder (codec) that does this needs an evenly spaced (in time) stream of data items. If the next data item is not available when it is needed, the codec has no choice but to produce silence or guess — and if the data is late, it is useless, because the time period when it should have been converted to a signal has already passed.
At the time of the design of ATM, 155 Mbit/s Synchronous Digital Hierarchy (SDH) with 135 Mbit/s payload was considered a fast optical network link, and many Plesiochronous Digital Hierarchy (PDH) links in the digital network were considerably slower, ranging from 1.544 to 45 Mbit/s in the USA, and 2 to 34 Mbit/s in Europe.
At this rate, a typical full-length 1500 byte (12000-bit) data packet would take 77.42 µs to transmit. In a lower-speed link, such as a 1.544 Mbit/s T1 line, a 1500 byte packet would take up to 7.8 milliseconds.
A queuing delay induced by several such data packets might exceed the figure of 7.8 ms several times over, in addition to any packet generation delay in the shorter speech packet. This was clearly unacceptable for speech traffic, which needs to have low jitter in the data stream being fed into the codec if it is to produce good-quality sound. A packet voice system can produce this low jitter in a number of ways:
- Have a playback buffer between the network and the codec, one large enough to tide the codec over almost all the jitter in the data. This allows smoothing out the jitter, but the delay introduced by passage through the buffer would require echo cancellers even in local networks; this was considered too expensive at the time. Also, it would have increased the delay across the channel, and conversation is difficult over high-delay channels.
- Build a system that can inherently provide low jitter (and minimal overall delay) to traffic that needs it.
- Operate on a 1:1 user basis (i.e., a dedicated pipe).
没有评论:
发表评论