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Delay in inbound calls


Randall Garner
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We are noticing about a 3 second delay on intermittant inbound calls. We started logging the calls and reported the issue to our SIP Trunk provider.

 

Our Trunk provider, WideOpenWest (WOW) has ran an advanced trace on the calls we experienced the delay and told me the following -

 

"The SNOM SIP device introduces a delay of 100ms between consecutive packets when answering the call, without changing the SSRC, and thereby confuses our DSPs."

 

I've hear our provider refer to their SIP technology as MetaSwitch, but I'm not positive.

 

We are currently on version 4.5.0.1090 Epsilon Geminids (Win64)

Our main number goes to a Hunt Group with seven extension all in Stage 1 and Duration set to 15, with the final option going to an Auto Attendant.

 

Any advise on where to go from here?

 

Thanks,

Randall

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MetaSwitch is used in a lot of installations. That's not the problem.

 

I am sure when you measure the packets on the PBX server (e.g. Wireshark) you will see that the packets leave with 20 ms delay.

 

100 ms packet delay is a problem. It should be 20 ms, or maybe 30 ms (with 10 ms jitter). What could be is that if you have limited bandwidth, the SIP packets which can be easily 1 KB clog up the upstream. E.g. if you have 3 KB of upstream burst packets can occupy the line for 100 ms if you have 250 kBit/s. If you have hunt groups that send more calls out, e.g. forking to 3 hunt group members, then you end up with 100 ms even if you have 1 MBit/s upstream.

 

The solution is usually to introduce some kind of QoS, e.g. by giving RTP traffic a higher priority. There are routers available that do that for you; actually many modern router have settings where you can give UDP traffic from the PBX source address higher priority than e.g. TCP traffic.

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