Latency in RTP streams

[Patrice Bertin <patrice.bertin94 () gmail com>]

Hi everybody,

I have a capture example of a SIP call with G711 codec.
Looking at the RTP streams, the final field shows an X which normally
means that we have a VoIP related problem. 
By selecting the problematic stream and analysing it, wireshark provides
a packet-by-packet look at the stream. 
The thing I don't understand, is why does it give me the status "OK"
everywhere if I have the famous X and especially when I have a latency
of 5843.74 ms (cf the screencapture joined in the mail).
Other thing, which values Wireshark put for the latency and jitter? Is
it possible to modify it?

I hope this is understandable enough.

Thanks for your reply


Patrice

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