Synchronous Versus Asynchronous Networks
Distributed systems would be a lot simpler if we could rely on the network to deliver packets with some fixed maximum delay, and not to drop packets. Why can’t we solve this at the hardware level and make the network reliable so that the software doesn’t need to worry about it?
To answer this question, it’s interesting to compare datacenter networks to the traditional fixed-line telephone network (non-cellular, non-VoIP), which is extremely reliable: delayed audio frames and dropped calls are very rare. A phone call requires a constantly low end-to-end latency and enough bandwidth to transfer the audio samples of your voice. Wouldn’t it be nice to have similar reliability and predictability in computer networks?
When you make a call over the telephone network, it establishes a circuit: a fixed, guaranteed amount of bandwidth is allocated for the call, along the entire route between the two callers. This circuit remains in place until the call ends . For example, an ISDN network runs at a fixed rate of 4,000 frames per second. When a call is established, it is allocated 16 bits of space within each frame (in each direction). Thus, for the duration of the call, each side is guaranteed to be able to send exactly 16 bits of audio data every 250 microseconds [33, 34].
This kind of network is synchronous: even as data passes through several routers, it does not suffer from queueing, because the 16 bits of space for the call have already been reserved in the next hop of the network. And because there is no queueing, the maximum end-to-end latency of the network is fixed. We call this a bounded delay.