(15-May-2016, 16:59)arcam Wrote:(15-May-2016, 13:36)iamwappie Wrote: But..... If the TCP protocol is used it is assured that all network packages are received correctly. Put them in a buffer in the receiving device and once full enough stream and play them. So with that reasoning I though bit perfect is bit perfect....
You don't want to use TCP for streaming (due to latency), but RTP/UDP. Retransmission is typically handled in the RTP stack. Different retransmission strategies can be put in place. For example, for video, you don't need to retransmit past frames since they would arrive past their display time. For audio, you don't want to miss any packet or else you will hear glitches.
Latency is a non issue for non-broadcast music replay (in the home setting, not studio's) and asynchronous protocols that typically perform buffering. The only exception I can think of is when a single source is streaming to multiple and/or different receivers/endpoints.
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