[VoiceOps] Easy ways to measure PDD

Calvin E. calvine at gmail.com
Tue Apr 21 15:45:37 EDT 2015

You're absolutely right, I had a brainfart while looking at the Asterisk
CDR details. You won't get PDD from this.

Regarding the other discussion of what PDD is in SIP signalling versus real
human experience, it all depends what you're looking for and how you're
willing to find it. Detecting non-silence in the 18x pre-session or post
200 OK session media is probably the best way of gauging the subjective
human experience, but requires considerably more effort than checking the
SIP signalling. Also, RTP analysis is not an option if you do not anchor
media between your customers and carriers (passing SDP connection
address/port from end to end).

"UAS (downstream carriers) can start sending a 183 in place of 100" If your
provider is doing this then you should probably consider changing service.
Defaulting to 18x can break route advancement if UAS does not ultimately
200 OK the call. In my environment a 503, 404, 603 etc. after 18x is
assumed to be a voice reject and ends the call, whereas a rejection without
18x would advance to the next carrier.

On Tue, Apr 21, 2015 at 9:08 AM, Jay Hennigan <jay at west.net> wrote:

> On 4/20/15 5:43 PM, Calvin E. wrote:
>> Asterisk does have logging and can write CDR with start time and answer
>> time. The exact details will depend on the version of Asterisk you are
>> using.
>> For example,
>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification
>> Watch for lines with "disposition" = "ANSWERED" and calculate the
>> difference of "start" and "answer".
> PDD isn't the time difference between start and answer. It's the time
> difference between invite and ringing.
> --
> Jay Hennigan - CCIE #7880 - Network Engineering - jay at impulse.net
> Impulse Internet Service  -  http://www.impulse.net/
> Your local telephone and internet company - 805 884-6323 - WB6RDV
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