VoIP

Voice over Internet Protocol (VoIP) is a general term for a family of technologies to deliver voice communications over IP networks (like the Internet). Terms synonymous with VoIP include IP telephony, Internet telephony, SIP, Skype, voice over broadband (VoBB), broadband telephony, and broadband phone.
VoIP systems use protocols to control the set-up and tear-down of calls as well as audio codecs which encode speech over an IP network (and often a range of codecs are used). Routing VoIP phone calls over data networks (to avoid the need for separate voice and data networks) can reduce communication and infrastructure costs.
Over the last decade, a variety of business-drivers have created a need for businesses to impliment VoIP into their organisation in one way or another. VoIP can facilitate tasks and provide services impossible using traditional methods. You can transmit more than one telephone call over the same broadband connection can reduce costs. Secure calls can be made using standardized protocols (such as Secure Real-time Transport Protocol.) Location independence and mobility can be easily achieved using a varety of business VoIP-related solutions – with only a non-business grade Internet connection is needed to get a connection to a VoIP provider. For instance, call center agents using VoIP phones can work from anywhere with a sufficiently fast and stable Internet connection. Integration with other systems and services including video conversation, message or data file exchange in parallel with the conversation, audio Business Conferencing, managing address books, and passing information about whether others (e.g., friends or business colleagues) are available to interested parties.
SIP – Session Initiation Protocol

The Session Initiation Protocol (SIP) is a signalling protocol, widely used for setting up and tearing down multimedia communication sessions such as voice and video calls over the Internet. One common misconception is that SIP is only used by telephones or for telephone calls.
Other feasible application examples include video Business Conferencing, streaming multimedia distribution, instant messaging, presence information, systems monitoring and online games. Officially speaking, SIP is a 3GPP signaling protocol and permanent element of the IMS architecture for IP-based streaming multimedia services in cellular systems.
The SIP protocol is situated at the session layer in the OSI model, and at the application layer in the TCP/IP model. SIP is designed to be independent of the underlying transport layer; it can run on TCP, UDP, or SCTP.
In our experience, the biggest problem when implimenting SIP in the business environment across Wide Area Networks is NAT (Network Address Translation). This is where a device – usually a firewall or router will manage IP addresses and translate one IP into another – usually to conserve addresses, or add a layer of security.
SIP as an open technology which has successfully been adopted by a wide range of business telephone system manufacturers, enabling 3rd parties to manufacture devices which are much more feature rich than the only other industry standard – analogue devices. A SIP phone can conference, transfer, hold and provide potentially hundreds of business telephony features.
Quality Of Service

Because an IP network is unreliable (in contrast to the circuit-switched public telephone network) and does not inherently provide a mechanism to ensure that data packets are delivered in sequential order, VoIP implementations face problems mitigating latency and jitter.
Voice travels over IP networks in packets in the same manner as data, so when you talk over an IP network your conversation is broken up into small packets. These voice and data packets travel over the same network with a fixed bandwidth. This system is more prone to congestion and DoS attacks than traditional circuit switched systems.
Some delays can be minimized by marking voice packets as being delay-sensitive.
A cause of packet loss and delay is congestion, which can avoided by means of QoS. The receiving end must restructure IP packets that may be out of order, delayed or missing, while ensuring that the audio stream maintains a proper time consistency. Variation in delay is called jitter. The effects of jitter can be mitigated by storing voice packets in a jitter buffer upon arrival and before producing analog audio, although this further increases delay.
When IP packets are lost or delayed at any point in the network between VoIP users there will be a momentary dropout of voice if all packet delay and loss mechanisms cannot compensate. It has been suggested to rely on the packetized nature of media in VoIP communications and transmit the stream of packets from the source phone to the destination phone simultaneously across different routes.
A number of protocols have been defined to support the reporting of QoS/QoE for VoIP calls. These include RTCP XR (RFC3611), SIP RTCP Summary Reports, H.460.9 Annex B (for H.323), H.248.30 and MGCP extensions.
21CN

The 21st Century Network (21CN) programme is the network transformation project of the UK telecommunications company BT Group plc.
It will see the UK incumbent’s telephone network move from the present AXE/System X Public Switched Telephone Network (PSTN) to an Internet Protocol (IP) system. As well as switching over the PSTN, BT has revealed plans to deliver many additional services over their new network, such as on-demand interactive TV services.
21CN is BT’s global, software driven customer network that introduces a new, simpler portfolio of next generation services. It is the foundation for BT’s transformation into a global, software-driven communications business.
Fundamentally 21CN is about customer choice – it’s a platform for innovation that truly puts flexibility and choice in the hands of our customers – both in the UK and across the world. To meet the needs of all our customers – both today and in the future – the scope of 21CN has been expanded and now includes:
- The introduction of software-driven innovation capabilities
- Elements that better meet the needs of BT’s enterprise customers and global reach
- New technology to improve mobile access
- Consistency between 21CN and any potential BT next generation access strategy.
Phone System

A phone or telephone system encompasses the general use of equipment to provide voice communication over distances, specifically by connecting telephonesto each other. A telephone system may also be known traditionally as a PBX (Private Branch Exchange) or PABX.
Many people’s experience of a telephone system is a strange box on the wall, or a fridge-sized box in the server room, which remains on, but nobody knows how to administer it. Times are changing however, and most modern systems are small units, aesthetically lookling like a small network appliance, or a 19″ rackmounted unit, more akin to a computer server.
Digital Telephony

Digital telephony is the use of digital technology in the provision of telephone services and systems. Almost all telephone calls are provided this way, but sometimes the term is restricted to cases in which the last mile is digital, or where the conversion between digital and analog signals takes place inside the telephone. Telephony was digitized to cut the cost and improve the quality of voice services, but digital telephony was then found useful for new network services to transfer data speedily over telephone lines. Sometimes people refer to Digital Telephony as TDM, although this actually refers to “Time Division Multiplexing”.
IP Telephony

IP Telephony is a modern form of telephony which uses the TCP/IP protocol popularized by the internet to transmit digitized voice data. Contrast this with the operation of POTS (an acronym for “Plain Old Telephone Service”).
IP Telephony has a wide range of advantages over the ‘old school’ telephone systems. It is much more closely aligned with the data network, which virtually any modern business or organisation already has in place. It uses the same network equipment, the same physical infrastructure, and will scale to the size of the ‘network’ accessible to it (once this includes the ‘internet’, the possiblities are endless).
IP Telephony has been proposition for nearly two decades now, but is gaining traction year on year. Whereas 5 years ago an IP phone or IP line between telephone systems was an interesting addition, it is now part of the de-facto standard system.
Computer Telephony Integation

Computer Telephony Integration (CTI) enables computers to know about and control phone functions such as making and receiving voice, fax, and data calls with telephone directory services and caller identification. The integration of telephone software and computer systems is a major development in the evolution of the automated office.
CTI is not a new concept. Such links have been used in the past in large telephone networks but only dedicated call centers could justify the costs of the required equipment installation. Primary telephone service providers are offering information services such as Automatic Number Identification and Dialed Number Identification Service on a scale wide enough for its implementation to bring real value to business or residential telephone usage. A new generation of applications (middleware) is being developed as a result of standardization and availability of low cost computer telephony links.
IP Phone

An IP phone uses Voice over IP (VoIP) technologies allowing telephone calls to be made over an IP network such as the internet instead of the ordinary PSTN system. Calls can traverse the Internet, or a private IP Network such as that of a company.
IP phones use control protocols such as Session Initiation Protocol, Skinny Client Control Protocol or one of various proprietary protocols such as that used by Skype.
IP phones can be simple software-based Softphones or purpose-built hardware devices that appear much like an ordinary telephone or a cordless phone.
There also exist the possibility to reuse ordinary PSTN phones as IP phones, with analog telephony adapters (ATA). It may have many features an analog phone doesn’t support, such as e-mail-like IDs for contacts that may be easier to remember than names or phone numbers.
Softphone

In computing, a softphone is a software program for making telephone calls over a data network using a general purpose computer, rather than using dedicated hardware. Often a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a phone, with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC, or with a USB phone.
Softphones were traditionally proprietary – meaning you had to purchase software and licensing specific to your own business telephone system’s manufacturer. However more so in 2009 you can buy, or even download for free a softphone from any number of 3rd party companies, which will work and provide many common advanced telephone features within your larger telephony solution.
Softphones are available for Linux, Windows, Apple OSX, Windows Mobile, Symbian and other operating systems – providing a wide range of possiblities within mobility, cost and integration to business systems.
A Modern Contact Centre

Contact Centres for Businesses within the 21st Century incorporate technologies such as speech recognition systems and speech synthesis software to allow computers to handle first level of customer support, text mining and natural language processing to allow better customer handling, agent training by automatic mining of best practices from past business or interactions, and many other systems and technologies to improve agent productivity and customer satisfaction.
Automatic business leads selection or lead stearing is also intended to improve business efficiencies, both for inbound and outbound campaigns, whereby inbound calls are intended to quickly land with the appropriate agent to handle the task, whilst minimising wait times and long lists of irrelevant options for people calling in, as well as for outbound calls, where lead selection allows management to designate what type of leads go to which agent based on factors including skill, socio-economic factors and past performance and percentage likelihood of closing a sale per lead. The concept of the Universal Queue standardises the processing of communications across multiple technologies such as fax, phone, and email.
A Unified Contact Center provides your agents with a smart multimedia solution that can cope efficiently with large numbers of inbound inquiries and outbound contacts – by email, fax, IM, web chat, SMS or phone. Agents can deliver a superior performance that has a positive impact on customer satisfaction, your callers’ brand experience and revenues.
The right solution lets you improve contact center management. Reports help to finetune performance. More importantly though, you can empower your agents by introducing real-time self-monitoring functionality with Agent Desktop – so they can better self manage their workloads.
PBX

A private branch exchange (PBX) is a business telephone exchange that serves a particular business, office or organisation, as opposed to one that a common carrier or telephone company operates for many businesses or for the general public. PBX is also referred to as:
PABX – private automatic branch exchange
EPABX – electronic private automatic branch exchange
Historically, the expense of full-fledged PBX systems has put them out of reach of small business. More recently, however there has been a large set of small, consumer-grade and consumer-size PBXs available. These systems are not comparable in size, robustness or flexibility to commercial-grade PBX, but still provide a surprising set of PBX features. These systems are more attractive to the small business because they’re available to those on a tighter budget, and they can enable the business to perform much more like a larger company.
The first small business PBX systems were for the analog telephone systems, typically supporting four private analog and one public analog line.
Particularly in Europe these systems for analog phones were followed by small business PBXs for ISDN. Using a small PBX for ISDN is a logical step, since already the basic rate interface of ISDN (which is the phone interface individuals and small businesses typically get) provides two logical phone lines (two B channels) which can be used in parallel. Small, entry-level systems are also extremely cheap.
With the pickup of VoIP PBX, PBX functions have become simple additional features of consumer-grade or small business routers and switches.
Open source PABX systems provide flexibility and features, plus the means to actually inspect and change the inner working of a PBX. They have also opened business opportunities for newcomers to the market of mid-size PBXs.
IP PBX

An IP PBX handles voice signals under Internet protocol, bringing business benefits for computer telephony integration (CTI). An IP-PBX can exist as physical hardware, or can carry out it functions virtually, performing the call-routing activities of the traditional business telephone system , in the form of PBX or key system as a software system. The virtual version is also called a “Soft PBX”.
DECT

DECT or Digital Enhanced Cordless Telecommunications (formerly Digital European Cordless Telephone) is an ETSI standard for digital portable phones, commonly used in businesses or organisations. DECT can also be used for wireless broadband data transfers. DECT is recognised by the ITU as fulfilling the IMT-2000 requirements and thus qualifies as a 3G system. Within the IMT-2000 group of technologies, DECT is referred to as IMT-FT (Frequency Time). DECT was developed by ETSI but has since been adopted by many countries all over the world.
Unified Messaging

Unified Messaging (or UM) is the integration of different streams of communication (e-mail, SMS, Fax, voice, video, etc.)enabling businesses to bring all of these mediums into a single unified message store, accessible from a variety of different devices.
Unified messaging is a subset of a fully integrated business Unified communications system. While traditional business telephone systems delivered messages into several different types of stores—voicemail systems, e-mail servers, and stand-alone fax machines—with Unified Messaging all types of messages are stored in one unified system. Voicemail messages, for example, are delivered directly into your inbox. You see them right beside your e-mail when you open up Outlook, offering powerful new ways to collaborate more effectively. For example, you can forward a voicemail or fax. You can even take notes in your voicemail message or search for old voicemail messages. No more notes stuck to your monitor!
Today, UM solutions are increasingly accepted in the corporate business environment. The aim of deploying UM solutions generally is to enhance and improve business processes as well as services. UM solutions targeting professional end-user customers integrate communications processes into the existing IT infrastructure, i. e. into CRM, ERP and mail systems (e. g. Phoenixnet PH, Microsoft Exchange, Lotus Notes, SAP, etc.).
Unified Billing System

New England Telephone Company and unify all your billing into one system – to your customers, or from your service providers – bringing it all into one easy-to-administer portal. This can bring a far greater efficiency to your business and reduce the amount of personnel hours used to preform your business or organisation’s backoffice tasks.
Our user portal is accessible from any machine with an internet browser, is fully secure, and available 365 days a year.


