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Full Version: [Q] SIP Qualify latency
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I am testing the W52P with a local Asterisk 1.8.23 server. All extensions and trunks have the SIP setting qualify set to yes to monitor their availability.
For some reason, the W52P consistently shows a latency of of ~130 ms, increasing to ~160 ms when on a call.
Everything is on the same subnet and same switch and no NAT. Other endpoints have a single-digit latency, including another DECT offering from a competitor.
This 130 ms latency is actually worse than my slowest trunk.
While I am not finding any issue that can be linked with this latency, I have had occasional dropped call and some voice quality issues, nothing that I could reproduce. It leaves me to wonder if this could be causing some problems somewhere...
Any opinion on this?
Hi lleo_,

Can you supply more infos to me?
1. Which sip message do you count about the latency? Is it an INVITE or NOTIFY message?
2. Do you have other Yealink phones like T4X, T3X,T2X? What is the account latency of them?
3. Abou the droped line issue and voice issue, please check the distance between base and handsets. Make sure the network is stable.
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