Measuring Total End-to-end Latency

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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?


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)

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.


One thought on - Measuring Total End-to-end Latency

  • 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 <>: