On 09/06/2015 02:10 PM, John Andersen wrote:
Voice is not that demanding. Buffering issue haven't been an issue between my UK customers and our offices.
That depends on whether it's interactive, such as a phone call, or not. With phone calls, delays of 400 mS or more can really interfere with speech. On the other hand, one way speech wouldn't be affected by several seconds of buffering. Years ago, when satellites were used for intercontinental phone calls, a call would go via the satellite in one direction and undersea cable in the other, in order to minimize delay. Of course, with VoIP and cell phones, there is already a built in delay and any due to buffering, latency, etc. simply adds to it. As for TCP, you can often have significant buffers, much larger than are typically used with SIP/RTP and if there's a lost packet, the flow to the receiver stops, until that missing block is replaced. With UDP, the missing data is simply ignored and there are some methods used to minimize the impact on call quality. Now, with TCP, the receiver typically sets the buffer or sliding window size, but the transmitter has to realize that some data is missing, which depends on flow control timers and then transmit again. How long will that take? Depending on the TCP implementation, that transmitter may retransmit all the data after the missing block too, which adds more delay for following data. I've experienced enough YouTube etc. videos that were starved to know you don't want that to happen in a phone call. -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org