Every industry has its specific terms and jargon and this can be confusing for the newcomer and even more so for people researching new products or solutions.

The telephony and communications industry is no different. With the migration from Digital PBXs to new Server-based, IP networking based systems, those industry terms expanded dramatically. This list of terms is something we compiled to help go through our site, documents, blog, etc. We hope it helps.

Analog or Analogue Telephony

When telephones were first invented all the voice communication and the signalling, to instruct the telephone system who to call, were Analog. Rotary dial phones sent pulses down the line to indicate the number being dialled – 1 pulse for 1, to 9 pulses for 9 (and 10 pulses for zero). The telephone system counted these pulses and used them to direct the call to the final destination. (If someone turned the dial when you were talking, you would be interrupted by a series of clicks. The voice and signalling connection were the same.)

The handset was limited to only sending a portion of the bandwidth that someone could hear (topped out at 3.3khz or under 4khz) – this allowed the telephone wires to be adjusted to send the voice signal further down the wire without being amplified. This was less than the full range we could hear but was considered enough to determine what someone was saying. In fact, it has been a sufficient range even to today and we can definitely make out who is calling – although you will now see some systems offering ‘wide-band’ capability.

This was all done over a single pair of wires and later this would all be referred to as Plain Old Telephone Service or POTS. While it might seem basic today, it was used to create national phone networks.

Later advances added touch-tones as a way to communicate. Now the telephone system would listen for a series of tones to determine how to route the call. And this would also allow features to be added like Caller ID, Call Waiting, Speed Calling and more. These came out in the 1980’s just as the advent of digital telephony was being brought to the market. However, these systems were highly reliable and available and brought with it the notion of the highly reliable telephone system with a greater than 99.999% up time or five nines.

PBX or Private Branch Exchange

The telephone companies used the term trunk to indicate the connection between a central office switch and a telephone system being used by businesses. And as businesses were buying their own telephone systems to mainly connect their internal employees but also allow people to call in and out, the term Private Branch Exchange was used. Linking the tree theme of trunk and branch – which still continues to be used.

So rather than buy 100 lines if you had 100 employees, a business would buy a PBX and connect say, 10 trunk lines from the telephone companies Central Office. This was far more cost effective and the PBX provided all sorts of useful features like call forwarding, paging, meet-me paging, conferencing and more.

Digital Telephony

With the development of the transistor, microchips and digital memory, PBXs could now be made more cost effectively, be smaller in size and provide more advanced capabilities.
Now your voice would be digitized at the handset. It would be sampled and converted to a series of ‘0s’ and ‘1s’. Because the cost of memory was still high, it was limited to 8 bits per sample. And to ensure the highest frequency would be captured was sampled at 2 x the highest frequency which was 4khz. So 8 bits x 4Khz or 4,000 cycles per second became 64,000 bits per second or the standard 64Kbps we see a lot in communications today.

No longer limited by touch tones to initiate a feature, the Digital PBX could provide separate signalling between the core PBX ‘server’ and the telephones as 4 wires were used to connect the phones, separating the voice from the signals.

Buttons could be used for a feature – one button per feature – lights could indicate which line was calling, a phone could conference multiple lines and more. Even calling features became more intricate including advanced inbound call routing or call centres. Sophisticated messaging applications, displays with messages and more.

PBXs became the workhorse of the communications world which still primarily relied on voice connectivity for immediate service.

Trunk

Trunks are the connections from your Service Provider or Telco to your Telephone system. There are several ways that can happen. Analog, Digital and now SIP.

Many people still have Analog trunks that can be purchased in units of one. They offer limited capabilities although they can provide the calling line ID. If you have a very small system and need only a few trunks, this may still be a good option. Tend to be highly reliable.

Digital trunks bring a single 4-wire connection into your telephone system but can provide the equivalent of 24 analog trunks. Moreover, these trunks can be used for different purposes. They can be for DID (Direct Inward Dialling) or outbound calls or 800 calls. They are much more flexible. Service Providers or Telcos will offer different ‘bundles’ some will provide as few as 8 channel or trunk equivalents while others may start at 12. Businesses tend to have a ratio of 4:1 to 5:1 for Telephones to Trunks for a typical business. So if you have from 30 to 40 telephones a digital trunk will start to make sense.

Services (or features) can be acquired on top of the physical trunk connection. For example, 800 service in which you pay the Telco for each call made to you over the ‘toll free’ line you are offering your customer. Or DID for Direct Inward Dialling, in which you can have the caller go directly to your telephone and not the reception or attendant console. The last few digits of the number dialled are passed to your PBX that converts the call to your desk phone. You buy these DID numbers on a per month basis in blocks. These can be as few as 10 or 50 depending on the Telco. But could be much less expensive than hiring a receptionist/attendant and allows some of the great new features like twinning mobile phones to your desk phone providing faster access.

VoIP

VoIP stands for Voice over IP or more completely Voice over Internet Protocol. In telecommunications a protocol means the rules that govern how devices communicate with each other. Internet Protocol is the primary protocol for managing and routing data over the Internet and, indeed on your internal data network.

Unlike a Digital voice system that utilises a ‘fixed’ connection between the two endpoints – telephones – and sends a continuous stream of ‘0s’ and ‘1s’ – even when there is silence or only one way when one person is talking – data networks are shared.

All data are put into packets – think envelopes – with an address on the outside of where to send the packet. Imagine someone sending a 10 chapter book where one chapter at a time is put into an envelope. Each envelope is numbered from 1 to 10. Some chapters are bigger than others so take longer to deliver. Chapter 3 is caught in traffic and doesn’t arrive until the end. The person receiving the envelopes waits until they all arrive, puts the envelopes back into sequence, opens them and recreates the book to read.

The envelopes don’t have to be of the same size, so large files can be chopped into larger blocks. And a larger block may take longer to send. Also, as it is a shared network any envelopes caught behind the larger one will have to wait. This is not ideal for telephony which is a real time application, the two parties cannot afford to wait, re-order and piece together bits of the conversation. You can imagine how garbled that would sound.

In the late 90’s people were experimenting with new ways to digitise and compress video, including the voice component to send over data networks. Ultimately this led to standards for how that was done. At that time international telephone calls were very, very expensive. Using these standards and stripping off the voice component, more calls could be squeezed onto the same circuits being use to carry just one call, and offering lower cost international calls. Thus Voice over IP was created.

IP Telephony

Leveraging on the work to develop VoIP, telephone system vendors starting adding support for IP Telephones to their existing systems. These IP Telephones would be connected to the data network and the telephone control unit would also be connected to the data network. For calls between IP Telephone and IP Telephone, the packets or envelopes would be passed directly between each other. Requests for features, like conference or call forward, etc, would be sent to the Telephone system via the data network. If calls were required outside of the business or to other non-IP Telephones, the packets would be sent to special cards in the Telephone system to ‘translate’ between Digital telephones and IP Telephones or Digital trunks and IP Telephones.

Today, most telephone systems are IP based – some only support IP Telephones. This eliminates many of the special cards and proprietary hardware of digital PBXs. The software is now delivered on standard computer servers connected to the data network.

They still support all the same features as Digital telephones and many manufacturers even create Digital and IP Telephones that look exactly the same on the outside – just the electronics differ.

If you have CAT5 or better cabling and a quality data network that supports PoE, then it probably makes sense to use IP Telephones. Most IP Telephones have a data jack on the back that computers can be plugged into, so only one data port is required. With high speed data networks, the bandwidth is sufficient to support both. And they can also be configured to virtually ‘separate’ the voice and data traffic (VLANs), apply a higher QoS to the voice traffic packets, prioritizing them, and put in place other mechanisms to ensure that voice traffic is not negatively affected by other traffic.

With SIP Trunks being readily available and less expensive than traditional trunks, you can create an ALL IP based system.

With advances and cost reduction in Internet connections and computing power you can also use a cloud based telephone service, delivered to your site, home, mobile phone, anywhere in fact, without needing to buy or house equipment in your office or premises.

SIP

SIP stands for Session Initiation Protocol. SIP is a form of signalling that allows greater capabilities than what was possible with older analog and digital systems; it allows much more advanced capabilities – like instant messaging and presence. Many IP Telephones are, in fact, SIP Telephones.

Because basic SIP definitions should be standard, many SIP telephones are interchangeable or work on different systems, whereas all digital telephones were always proprietary. For example, our Flexfone Essential cloud based telephone service uses SIP phones from Aastra, Panasonic and Polycom.

SIP Trunk

Instead of bringing in dedicated physical trunk connections to bring in incoming calls as with Analog and Digital trunks, SIP Trunks leverage your standard Internet connection. Data networks or Internet connections are inherently a ‘shared’ connection. One internet connection can have a significant bandwidth today. Remember an Analog or Digital trunk is the equivalent of a 64Kbp/s connection and yet you can buy 10Mbp/s, 20Mb/s, 100Mb/s or even higher Internet connection speeds. With appropriate configuration your Internet connection could now be used for your incoming telephone calls at a significant reduction over traditional trunks – as much as 30% to 60% less.

Unified Communications

UC or Unified Communications is another of those terms that is confusing as so many vendors have defined in their own way to highlight some feature or capability of their product. Most communication vendors make a variety of ‘applications’ like voice messaging and IVR as well as integrating their telephone server with the corporate directory service, email systems and more.

As PBXs migrated to IP Telephone systems and the proprietary CPU or Controller migrated to an off the shelf server, often a virtualized server, it was possible to load all the applications onto a single server. So Telephony, Messaging, Conferencing, Call Centre all loaded on to a single server. This reduced or eliminated integration between these applications and meant a single Active Directory or Email integration meant all these applications were now ‘Unified’ onto a single server.

It has also often meant the (simpler) integration of collaboration applications like instant messaging, presence , data sharing (white-boarding), speech recognition and so much more.

Depending on what manufacturer you look at, you will get a different definition. But ultimately it does represent the broader communications capabilities that you can now use within your business.

Collaboration

A difficult term to narrowly define but in communications terms we are talking about conferencing, presence, instant messaging, video and more. Capabilities that let 2 or more people work together. In the traditional telephony world collaboration usually meant a one-to-one telephone call or a multi-party conference call – still very useful. Today that is no longer sufficient. People are on the road, at home, working from different cities even and want to leverage all the tools available to connect. And to leverage the power of smart devices such as today’s laptops, tablets, smart phones as well.

So collaboration has come to include capabilities that may require the ‘connection’ or integration of telephone systems or PBXs to other systems – email servers, Microsoft Office, and more. But now doing it simply without users having to go through awkward processes to connect. So click-to-call from within any office application or when on the web; Seamlessly setting up a video conference from your laptop, tablet or phone. Joining in calls and being able to share data, files and perhaps editing them. There are so many ways to collaborate now.

Hosted

The term hosted is troubling as it can denote either a cloud delivered application, it can mean a Private Cloud solution, where you own your own equipment but are ‘hosting’ it outside your own premises or office or a combination of those, often and perhaps incorrectly, referred to as Hybrid.

So we have further defined Cloud, Private Cloud and Hybrid below to make this simpler.

Cloud

When related to telephony, cloud telephony refers to offering or delivering telephone service from the Service Providers data centre. All the common equipment resides in the data centre and only the telephones or other end user device would be on your premises or in your office. The service would be delivered over the Internet and your internal data network.

The service is a monthly fee service or OPEX (OPerational EXpense.) All the ‘trunk’ connections are at the Data Centre, so you don’t have to pay a monthly fee for trunks coming into your building.

The major advantage of this kind of service is in eliminating all the support, maintenance of the controller or server, upgrades, patches, etc. This is ALL handled seamlessly by the Service Provider. It also tends to be more flexible, especially for smaller businesses or small multi-site businesses or those with seasonal variations in employee numbers. See the Flexfone Essential and Flexfone Advanced pages for more information.

Private Cloud

Some businesses want to own their entire telephone system but would like to host the server somewhere else. Perhaps for risk mitigation (for example their location is not ideal – in a flood plain for example) or to have someone else worry about the day to day housing of the communications server(s) in a true Tier 2 or 3 data centre with power back up, cooling and dual egress to service providers.

It may even simplify connectivity to multiple sites and perhaps allow or reduce the cost of network connectivity.

There are many great reasons for considering this as communications systems become the key element of not just telephony but overall collaboration and where an ‘always on’ 24x7x365 requirement is essential for the business.

Hybrid

This term is often used to indicate Private Cloud. But there are now vendors offering the ability to have a true Hybrid solution in which Some of your sites may have a premise based solution – with servers dedicated to a location, perhaps your Head Office. Perhaps the server isn’t on your premises but you have acquired all the technology and hosted some of it at a Data Centre for any number of reasons. However, your company also has a number of smaller sites. You don’t want to place smaller systems at these locations, bring in trunks, etc. But you can acquire a Cloud based service that uses the same technology you have acquired for your main site. And these have been ‘connected’ so that you can provide seamless capability across all your locations, even though some are completely owned (CAPEX) and some are Cloud based (OPEX).

This would be a true Hybrid solution.

QoS – Quality of Service

Quality of Service is a term used in data networks concerning communications and collaboration, specifically with IP Telephony. IP Telephony places your telephone calls over a data network and not a dedicated voice network as with Digital telephony. But a data network is a shared network which means your voice call competes with files being printed, files being shared, emails, yes, and even people looking at pictures of cats over the internet. All that data is travelling over a single shared network.

To ensure that real-time business applications like telephone calls are not going to be disrupted we need a way to manage and prioritize that telephone traffic. This is often referred to as QoS or Quality of Service. Some tools have been created to try and quantify and measure QoS, such as a MOS or Mean Opinion Score.

Ultimately there are several factors that can impact the quality of voice calls from delay, jitter (variation in delay), prioritization, error rate and so on. By configuring your data network you can reduce or eliminate many of these factors.

In fact, it is imperative that you configure your network to do this as IP Telephony will be a constant challenge.

Newer data network devices have the tools built in to allow your network to be optimally configured, while older ones may not. Furthermore, the newer systems may even provide tools to simplify this process by automatically recognizing an IP Telephone has been attached and configuring the connection based on a set of predefined rules.

A good IP Telephony VAR will ensure this is done correctly.

PoE – Power over Ethernet

Another term newly introduced into telephony with the advent of VoIP and IP Telephony. PoE stands for Power over Ethernet. With Analog and Digital telephone systems the telephones were powered directly by the telephone system. If local power was lost the telephones would still work (if the PBX was on a UPS) as power was delivered over the dedicated CAT3 (or better) wiring between each telephone and the PBX.

Early data networking switches didn’t offer this capability as the devices they connected were powered locally; mostly desktop PCs, servers and Printers. Most IP Telephone manufacturers made a local power adaptor for their phones but this was messy compared to what was needed for Digital telephones. Standards for PoE were developed and Data Network Switch providers starting adding PoE capability to their switches. Now power could be passed over the CAT5 (or better) data network connections to the phones (and wireless Access Points which also needed powering).

Now there are many more devices that need power with the evolution to the Internet of Things or IoT, such as lights, security cameras and door controllers.

But care must still be taken as not all PoE switches are the same; do not deliver the same amount of total power. With the many devices that need powering we incrementally increased the power output through the new standards that have come to market. PoE comes in different ‘classes’ meaning different power supplied – usually indicated in watts. These go from 4W up to 30W.

Class Usage Power Level Output at the Power Sourcing Equipment (PSE) Maximum Power Levels at the Powered Device (PD)
0 Default 15.4 W 0.44 – 12.95 W
1 Optional 4.0 W 0.44 – 3.84 W
2 Optional 7.0 W 3.84 – 6.49 W
3 Optional 15.4 W 6.49 – 12.95 W
4 Valid for 802.3at High PoE 30 W 12.95 – 25.5 W

When checking a data network switch make sure to calculate the maximum power draw required. Some switches will offer up to Class 4, but can’t provide that for ALL the ports.

CAT3/5/6 Cabling

The cabling used today to support telephony – Analog, Digital or IP Telephony – is based on twisted copper pairs. The term CAT for Category is used to precede a version of cable. Over time, as the need to support higher data rates has been required newer versions have come to market.

CAT3 was the most commonly used to support Digital telephone systems. But is not suitable for data networks.

CAT5 supported a bandwidth of 100 Mhz which meant 100BaseT and 1000BaseT Ethernet.

CAT5e with the e for enhanced and provided improved quality and superseded CAT5

CAT6 is supports even higher LAN speeds up to 10GBaseT

There are other versions but these are the most common.

In some cases, only the Server connections to a Server Rack data switch might be at CAT6 while the remaining cabling can still be at CAT5. Each iteration increases in price and moving to CAT6 is much more expensive. Also, as business install higher speed business Wi-Fi there may not be a need for CAT6 to the desk.

Finally, and probably of greater importance may be the need to use fire rated or fire retardant ‘plenum’ cable (plenum being the space in the ceiling cables and conduit are run) depending on your local regulations. If cable is run in conduit, this may not be required but if run open in the ceiling, it may be a requirement and is more expensive to install.

Presence

Many collaboration applications depend on ‘knowing’ a person’s status or availability. For example, an instant messaging client will only show those people you want to be connected to who are currently available or who may be online but are in, say, Do Not Disturb mode or a meeting. The idea is that by knowing a person’s ‘presence’ or availability you can make a better decision on how to communicate with them.
Depending on the system the person can choose how to display or ‘publish’ their status such as available, away, in a meeting, do not disturb, etc. Some systems will dynamically update that status. For example, if on the phone will show on a call, if in a video conference will say that too. It will vary dramatically between systems and what the person who is ‘publishing’ wants to share and even, if they want to share with you.
Ultimately, many people are using these types of applications outside of work, but they are insecure. You risk important and confidential information getting into the public domain. By implementing a business collaboration solution, you can reduce the risk of that information becoming public.

IM – Instant Messaging

Instant Messaging or chat is one of the most common collaboration applications today. Often used with Presence, you can see the status and availability of a colleague and, if free, engage in a real-time chat communication. For most business systems today this can be done from any location and any device, meaning that it improves communication and connectivity between members of your staff and does so securely.

Jitter

A funny name for a technical term but it has serious, negative consequences when making telephone calls over a data network. When you use an IP Telephone it turns your voice into hundreds or thousands of small data packets. In an ideal world all the packets would be delivered in a smooth and seamless fashion, one after the other. In the real world, other ‘traffic’ on your data network will cause those packets to be disrupted. Just as you cannot always drive you car on the highway at a constant speed because of the amount of traffic varying along the route. This variation in delay is called Jitter.

So to help control the disruption caused by this varying delay or jitter, systems incorporate jitter buffers. Small data storage areas to collect the packets as they arrive and provide an opportunity to smooth out the arrival rate to other person’s ear. Too short and echo and variation in speech may happen. Too long and the delay between speaker and the other person becomes too disruptive as people don’t know when the other person has started or finished speaking.

Help on these topics?

At Unity Connected Solutions we work with customers on their communications systems everyday. We are IP communications specialists. We have provided on premise, hosted, private cloud and cloud solutions. If you have any questions on these topics or have other terms you need help with, please reach out by clicking the button below and emailing us.

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