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List:       asterisk-dev
Subject:    Re: [asterisk-dev] Extensions to RTPAUDIOQOS for SIP/other channels
From:       Don Morrison <dam () us ! ibm ! com>
Date:       2007-04-27 18:22:48
Message-ID: OF7E277521.D288B2FD-ON862572CA.0063BFA7-862572CA.0064F715 () us ! ibm ! com
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> Also to note is that RTPAUDIOQOS variable doesn't really work correctly,
and
> thus you should pull these stats from the CHANNEL() function. Just in
case
> you weren't aware of stats being there.

Leif, can you please expand a bit on this? I see where there was a bug
fixed recently where the lost packet count was wrong. And from what I
gather,  RTPAUDIOQOS only contains statistics between the caller and the
Asterisk server. Are there other problems?

I've been desperately trying to find a way to measure the audio quality for
those calls that traverse the Asterisk server. Dumping RTPAUDIOQOS to the
CDR file is the best I've been able to come with. I'll take a look at
CHANNEL(). Let me know if there are any other ways to get what I'm after -
accurate jitter and lost packet counts for the entire audio path would be a
good start. Thanks.


Don Morrison

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