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We are currently troubleshooting a delay issue in our network and I would like to know what unit the jitter buffer value is in the Wireshark - RTP Player?


Kind regards, Nicklas.

asked 06 Apr '16, 07:57

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Nicklas Bargell
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That would be milliseconds, or something odd is going on.

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answered 07 Apr '16, 01:09

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Jaap ♦
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That's what I thought, but when I replay the same sound stream both incoming and outgoing and change the value from 50 where there is almost no delay down to 5 I get lots of delay. More then a few seconds. And that seems strange?

(07 Apr '16, 01:18) Nicklas Bargell

Not nessaserraly. You would have to delve into the internals of media play out to see what's going on. Effectively what you did is (depending on the specifics of the packet time stamp distribution) making it very hard for RTP packets to hit the window in which they can be accepted for play out. Usually a single packet represents 20 to 30 ms of speech, so 5 ms is very narrow in that respect, probably causing the play out mechanism to have to resync all the time, possibly causing these long mutes. As you see there are a lot of ifs, because it all depends on the specifics of your RTP stream. But you could start by looking at the packet time and use that as lower limit for jitter buffer.

(07 Apr '16, 04:50) Jaap ♦
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question asked: 06 Apr '16, 07:57

question was seen: 1,898 times

last updated: 07 Apr '16, 04:50

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