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I am going through some documentation of a voip software that uses Live555 as the underlying network layer. As per RFC for RTSP - live555 seems to have implemented it. But the output is not clear to me. From archives of Live555 here question it seems that to get jitters in terms of mirco or milli seconds, I have to divide the jitter value by sampling frequency. But what about the network bit-rate? Should I use it to divide the jitter value to derive jitter in terms of micro/milliseconds?

Any help is appreciated

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  • Output is not clear in what sense ? Please be elaborate. Is it getting stuck? Is there a visible frame loss? – Shehryar Jan 27 '15 at 06:11
  • The voip application displays jitter values but does not provide units, So I checked the code and found out that I need to divide the jitter values by sampling rates. Jitter values are for Video and Audio streams separately, I want to know if I divide the jitter value for each stream by network-bitrate or sampling rate of audio and video. – Wajih Jan 27 '15 at 06:25
  • I think you should be dividing it by the network-bitrate because that is the final link in the chain. Your app basically displays the jitter you get whence the RTSP packets have traversed the network and will, in-effect also contain pre-packet encapsulation jitter values. I hope this helps. – Shehryar Jan 27 '15 at 06:29
  • Thanks - I was thinking of the same but still unsure about this. May be I will dig it out somehow. – Wajih Jan 27 '15 at 10:36

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