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Voice, the network, and the human

Editorial Type: Opinion     Date: 09-2014    Views: 2323   





Patrick Hubbard, Head Geek at SolarWinds reflects on the glorious past of the telephone - with a hard stop in the future...

Our expectation of telephone call quality was mostly influenced when using PTSN (Public Telephone Switched Network) which was, in all its bundled copper analogue glory, amazing. It's no surprise really because when a switched network dedicates its entire bandwidth to a single function, it should be.

VoIP runs on heavily converged networks trading lower quality for convenience and lower cost - and completely overlooking the user. Using a PTSN bias, all but the best VoIP deployments seem counter intuitive, leaving IT administrators stuck in the middle to create happy users and prevent a help desk deluge of poor VoIP quality reports.

WE'RE NOT TELCO LINEMEN
Sadly, the FeatureLine and private branch exchange (PBX) solutions never really ended for many enterprises who implemented IP-PBX internally, using their PTSN trunks. The Post Office managed to innovate, moving from TXK crossbar switches to TXE reeds and eventually real live digital switching with System X.

However, enterprises stuck with PBXs hanging off increasingly sophisticated PTSN foreign exchange subscriber (FXS) ports, as their internal foreign exchange office (FXO) ports remained mired in the past. This changed with VoIP.

When network service delivery is poor we consult our network performance monitoring tools. When call quality is poor, it's unclear where to start. The BT line engineer would clip a Lineman's handset onto punch-down posts, looking for an engaged line to listen to for a minute or so, then declaring a MOS (Mean Opinion Score) value using their ear and brain to form a subjective opinion of call quality.

Listening for hum, clipping, echo and other areas of call path weakness is not an option for IT administrators. There are VoIP call paths but they are router to router and we don't have packet-based Lineman's handsets to reassemble flows and listen to calls. We rely on simulated call metrics like IP SLA, or dive deep and try to parse our VoIP controller's call data record files into something useful. Today's calculated MOS, though detailed, is frankly not as useful for troubleshooting.

IMPLEMENTING QUALITY GATEWAYS
Generally speaking the gateway or network can cause problems for VoIP, one of which at least is straightforward to diagnose and correct. If it's the network, dropped packets or jitter are top suspects. This can be addressed like any other traffic issue by locating the overloaded network device, congestion, miss-configured QoS/ToS or route flapping. If it's the gateway, it's a different story.

The gateway, and specifically the interface between FXS ports and the VoIP infrastructure on the inside, is a rich source of issues. Identifying a mismatched hybrid analogue connector or IP delay as the source of call echo takes time. The PTSN network expects near speed of light transport while your LAN may be in no hurry, causing a loopback echo at the analogue/digital interface.

Is it crosstalk (current leakage) on the outside or 25 ms latency on the inside? If it's delay, you'll find yourself first calculating optic delay (delay = transmitter delay + receiver delay + (0.005 ms/km * distance in km)), circuit by circuit. Crosstalk echo, hiss or asymmetric volume may have you searching for little used configuration commands to adjust impedance in high-density FXO/FSX cards.

YOU ARE NOT ALONE
The great news for network administrators managing VoIP is while you may think your environment is unique the basics of call quality are not. A user’s tendency to complain about call quality is a natural human reaction. Fortunately we don't have to provide universally perfect, HD calling. We do however need monitoring tools combined with knowledge to keep MOS above 4.0 and our users away from the help desk.

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