ducatirider Posted January 31, 2019 Share Posted January 31, 2019 I was curious how IP telephony and data networks differ. Why does UC require QoS prioritization? How is this different from computer audio? Link to comment
0 mansr Posted January 31, 2019 Share Posted January 31, 2019 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. Jud 1 Link to comment
0 Jud Posted January 31, 2019 Share Posted January 31, 2019 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. Link to comment
0 Popular Post mansr Posted January 31, 2019 Popular Post Share Posted January 31, 2019 1 hour ago, Jud said: 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. There's an easy way to have both fast start and a large buffer. Since the source is a stored file, and typical network bandwidth is much higher than required, the first second or two can be downloaded very quickly allowing playback to commence almost instantly. Then, while playing, the buffer can be filled at a higher rate until it holds any desired amount, potentially the entire track. This behaviour is readily observed on YouTube, where the red progress bar indicates the playback position and a grey bar shows how much has been downloaded. Jud and tmtomh 1 1 Link to comment
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ducatirider
I was curious how IP telephony and data networks differ. Why does UC require QoS prioritization?
How is this different from computer audio?
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