Having an issue with a new implementation of cloud hosted VOIP via Fuze and I was curious if anyone else has struggled with Fuze (or any other vendor) over the internet.
The problem we are having is after a SIP INVITE has reached a ring group and one phone picks up, the server is sending SIP CANCEL to all phones that did not answer in that ring group, but we are getting SIP CANCEL packets lost in transmission to phones occasionally. The result is the phone that did not rx the SIP CANCEL rings until it's ring time has been reached or manual intervention.
Understanding that via RFC 2543 a phones response should be 200 OK or 487 REQUEST TERMINATED if a CANCEL is rx'd, are there any vendors re-transmitting these SIP CANCELs for phones without a response returned?
Follow up, to those of you who use consumer grade circuits at your locations and hosted VOIP, do you have any issues with running SIP over TCP and standard internet packet loss? We are QoS'd to our edge honoring EF, CS3, AF41 - so the issue seems to truly reside on the internet with the loss.
Anyway, curious to hear about all the struggles of your hosted VOIP implementations so we can vent together.
Cheers! (is it okay that it's ONLY my 4th glass of the fire water today so far?)
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