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Are you ready to ditch the switchboard and move to IP telephony?
The future is most certainly voice
When I was growing up it was something of a novelty to have a phone in the house. And when I started working in IT in 1989 my company was still encouraging staff to wait to make calls until after 1pm, when peak time ended.
There was something unusual about the new phone system we deployed that year: the Philips Sopho-S was a digital PBX (private branch exchange) with digital handsets, an ISDN PSTN (public switched telephone network) lines and a Ferranti voicemail system – though we never quite got that to work properly.
Wind on 20-plus years and many are still using proprietary digital phone systems. It is only in the last few years that standards-based IP telephony has started to infiltrate the market.
Given that modern IP standards are well understood and widely implemented, why do we still have proprietary protocols? Should we be sticking with traditional technologies or should we be looking to the future?
The first digital phone systems used point-to-point digital connectivity between the PBX and the handset. The two had to be physically patched together end to end – they could use the company's structured cabling network but they were not Ethernet-based and didn't go through the LAN switches.
The telecoms manager at one of the famous public schools once told me that he liked this approach because it meant he could spread phones around the campus just using data cabling, without having to distribute bazillions of Ethernet switches. In this day and age, though, you would go for something IP-based which you could plumb into your organisation's network.
There are two choices with internal IP telephony: proprietary and standards-based. The main reason for a proprietary approach is that it tends to provide a more feature-rich experience than SIP (Session Initiation Protocol). With a PBX full of whizzy features only a proprietary phone – or softphone PC application – can make the most of them all.
It is common these days for vendors' handsets to be multi-protocol: a Cisco phone generally lets you choose either SCCP (the Cisco-specific protocol) or SIP. Similarly Mitel's handsets can talk MiNet (proprietary) or SIP. So you are not cutting off your options by choosing vendor-specific hardware.
Sometimes you have to go to the lowest common denominator that is SIP, most commonly where you want to hook in your smartphones. No problem here, though. All the modern phone systems let you combine vendor-specific handsets and SIP-connected applications, on the understanding that some of the funky features might not be available on these.
There is one other internal presentation requirement in many businesses, of course – analogue lines (what the Americans call POTS, or plain old telephone service) for modems and fax machines. There are many ways to stun this particular cat:
- On-board analogue: most PBXs have some kind of analogue option because most companies still need the odd fax machine, though you need to be able to string a point-to-point copper pair between socket and PBX.
- IP-to-POTS gateways: if your particular PBX doesn't have analogue ports, or you can't string a point-to-point connection, put an Ethernet-connected IP gateway near the fax machines and let it talk SIP back to the PBX.
- Don't even bother: either get your telco to pull in dedicated analogue services or outsource your fax requirements to an IP-connected fax gateway, or even a bureau service. There's no shame in saying “I can't be bothered with the hassle”.
For calling out to the big wide world, you have two options: PSTN has been around since the Ark and is known to be rock solid. IP trunks are a newer concept and are generally cheaper, but this can have a non-financial cost. We will come to that.
First, the PSTN approach. ISDN is the way to go. You really don't want to be mucking about with analogue lines as the call setup time is hideous. ISDN2 has two channels (hence two concurrent calls) and you can multiply them – rent two ISDN2s and you can support up to four calls.
If you want to do more than six concurrent calls go for ISDN30. This will support up to 30 channels but you can start small – the minimum in BT's case is eight channels. You benefit from scalability as you can get extra channels turned on quickly with no need to change the physical presentation.
IP is becoming increasingly common as a means of making calls that would previously have gone via the PSTN. A quick Google throws up a range of provider offers, many of them irresistibly cheap.
Is there a catch? Of course there is: it's called voice quality.
If you buy a SIP service from an arbitrary provider, you connect your calls to the provider via the internet. This means, of course, that you must have enough bandwidth to cope with the traffic load.
Although SIP calls take only a few kbps, if your link, or a link somewhere upstream on the internet connection, becomes congested for even a few seconds that's your voice quality down the drain.
Not great, particularly if you are communicating with customers and your staff all sound like old-time British comedian Norman Collier, famed for his “faulty microphone” routine.
We are starting to see more local telcos providing SIP connectivity through the internet connections they provide, and then either running their own SIP-to-PSTN services or hooking into similar services from others.
This is a Really Good Thing because your service provider is able to control the traffic streams through your local connection and over its network, thus guaranteeing the necessary bandwidth for the voice traffic and giving the rest to the less time-critical data streams.
If your provider is any good it will also have similar bandwidth controls termed class of service (CoS) across its backbone network and even with many of its peer providers. This is a more costly option but it means voice quality has an excellent chance of being as good as on an ISDN circuit, and it still costs less than the PSTN alternative.
Connecting sites together
This brings us to the final piece of the puzzle. If you are a multi-site organisation, how do you connect your telephony systems so that you can make calls not only internally within a site but also externally to customers and suppliers between sites?
In the basic form, each site could be a separate PSTN-connected entity. It is generally a simple matter to define plans that let you do funky stuff such as shorthand dialling – dialling a three- or four-digit number causes the phone system to expand this and dial a full PSTN number to connect to the remote office via the public phone network.
In its most advanced form you could build a WAN between all your sites to carry data and voice traffic, but this option is not for the fainthearted or those with small wallets.
The happy medium, then, is to ask yourself what you need in terms of inter-site voice connectivity. The answer is generally a compromise: good quality communication most of the time for a minimal cost.
As we mentioned earlier, if the local telco/ISP at any of your offices provides a SIP service over its network you can at least guarantee voice quality over part of the call path. But even if you are using crappy old ADSL connections at each location you will find that most of the time you can make decent quality calls between sites.
From time to time you may have periods of high-bandwidth usage but you can easily mitigate this. One option is to have small-scale PSTN links at each site and configure a prefix into your phone system that you can dial to bypass the IP connection and force the call to use the PSTN.
The other is simply to have an extra connection. ADSL links cost next to nothing – certainly far less than ISDN. Why not just have a 2Mbps ADSL set aside at your busier sites and dedicate it to the voice traffic?
At least you then know that someone emailing a socking great attachment out of the office is not going to trample on your phone calls, and even the puniest half-meg upstream ADSL can carry more than enough voice calls.
Five steps to voice
Points to remember when bringing voice to the enterprise:
- Don't be scared to use proprietary phone handsets, but try to choose a brand that supports SIP so they are future-proof, and ensure the PBX supports SIP so you can hook in smartphones and other SIP-only devices.
- Analogue is a pain in the butt, but you can support it very easily either by using a SIP-to-POTS gateway or by outsourcing functions such as fax that would otherwise use analogue.
- ISDN is flexible and rock solid, and although there is a cost you can scale it sensibly. If you absolutely need call quality, it is your friend.
- If you want to use IP voice for external calls, explore what your ISP can give you in terms of guaranteed performance. The greater the section of the call path you can guarantee, the better the quality.
- If you can live with the occasional drop-out or colleagues sometimes sounding like Daleks, you can do IP voice very acceptably over cheap and cheerful ADSL or cable modem connections. These can be cheap enough to make it a no-brainer simply to buy another link and dedicate it to the phone system. ®