[asterisk-users] Measuring total end-to-end latency

[Olivier]

Hi,

I don't have a direct answer, but I've read several times about purposely
customized system over the PSTN, echoing incoming incoming audio to produce
metrics
when troubleshooting call quality.

I alse remember a thing called Recqual targeting the same goal.

I hope this helps

2017-10-31 11:53 GMT+01:00 Antony Stone <
Antony.Stone at asterisk.open.source.it>:

> Hi.
>
> Does anyone have some recommendations for measuring total end-to-end
> latency
> (by which I mean: the time between person A saying something and person B
> hearing it) when there are both SIP and PSTN/analogue/mobile legs in the
> call
> path?
>
> Examples:
>
> Person A has a SIP phone registered to Asterisk, which has a SIP trunk to a
> connectivity provider, who has connections to PSTN (analogue landline)
> connectivity providers and to mobile network (Vodafone, Orange, etc)
> providers.
>
> Person B might answer the call on an analogue landline telephone.
>
> Person C might answer the call on a mobile phone (perhaps on its home
> network,
> perhaps roaming on a foreign network).
>
>
> Is there any way to measure total latency of calls between A and B or A
> and C?
>
>
> Thanks in advance for any ideas / suggestions.
>
>
> Antony.
>
> --
> "Remember: the S in IoT stands for Security."
>
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