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Media/WebRTC/WebRTC Debugging: Difference between revisions
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Delay remedy
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====Audio/Video Delay==== | ====Audio/Video Delay==== | ||
Delayed media is commonly caused by long physical paths between endpoints, though anything that slows down inter-hop delivery of packets can be at fault. Note that this is different than the bandwidth of the network path, and a high latency | Delayed media is commonly caused by long physical paths between endpoints, though anything that slows down inter-hop delivery of packets can be at fault. Note that this is different than the bandwidth of the network path, and a high latency will not be fixed by reducing the video resolution or audio sample rate. Round trip time, or RTT, is the time it takes for a packet to get from the sender to the receiver and then for a packet to get from the receiver back to the sender. If the path is symmetric between the two endpoints one can assume that the one way delay is half the delay of the round trip. | ||
The second common cause of A/V delay is jitter, the magnitude of variability in packet inter-arrival times. In order to smoothly play out of the incoming stream a receiver experiencing jitter will have to buffer (delay) incoming packets. | The second common cause of A/V delay is jitter, the magnitude of variability in packet inter-arrival times. In order to smoothly play out of the incoming stream a receiver experiencing jitter will have to buffer (delay) incoming packets. | ||