Hi, On Android clients RTSP playback has a latency of around 5-6 seconds. The client buffers data for this duration and then starts playing. For live streaming this latency is very high.
Does the client use the bandwidth information (b=AS:<bandwidth>) to calculate the buffer size. e.g 5-6sec in this case? How is the bandwidth computed in the Live555 ? Did some search and few people suggested to do a buffer blasting initially when the session starts. i.e send data at a high rate at start so that the client buffer fills up fast and eventually reduce the startup latency. Can streaming framerate be controlled at runtime in Live555 ? Regards, Sambhav
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