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IP telephony vs data transmission.


ducatirider

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It's all about latency. A conversation becomes difficult if the end to end latency exceeds 100 ms or so, and the network itself uses up about half of that. This means there's no time for buffering and retransmission of lost packets. QoS can reserve bandwidth and prioritise voice traffic in order to reduce latency and minimise packet loss.

 

Audio streaming, on the other hand, is a one-way affair, so there's no problem adding a large buffer to handle retransmissions and network jitter, even if the latency becomes several seconds. The music was probably recorded years or even decades ago anyway.

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44 minutes ago, mansr said:

so there's no problem adding a large buffer to handle retransmissions and network jitter, even if the latency becomes several seconds.

 

Yes, though these days as we become more accustomed to instant response from anything to do with our computers, a latency of several seconds might feel quite slow.  Even a second is probably an eon in terms of what's required for an audio buffer, I would imagine.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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