VoIP FAQs

Find out more by clicking on one of the headings below

VoIP for outbound and inbound call centres

Q.  Can call centre VoIP systems really match traditional telephony?

A.    Yes. You are probably making VoIP calls every time you place a long distance call.  

Telephone companies use VoIP to streamline their networks. By routing thousands of phone calls through a circuit switch and into an IP gateway, the bandwidth they use for the long distance is much reduced. Once the call is received by a gateway on the other side of the call, it is decompressed, reassembled and routed to a local circuit switch. In time, all the current circuit switched networks will be replaced with packet switching technology.  

Many companies have already switched entirely to VoIP. However, a VoIP system will only deliver its best performance and results if it is installed and configured correctly. Once the system is installed to the correct tolerances you will benefit from the reduced costs and improved flexibility.

Q.  Aren't call centre VoIP systems just the same as Skype?

A.    No. Skype and its competitors use VoIP to relay voice conversations across the internet. However, products such as Skype are designed as peer-to-peer (P2P) systems and, despite newer modifications to the software allowing them to connect to standard telephone lines, they are not as flexible as VoIP.

P2P software uses a large chunk of your available bandwidth to transfer data. VoIP systems are designed to use less, because such heavy bandwidth requirements would quickly overwhelm most business networks.

Q.  What about the quality of the call?

A.    Sufficient bandwidth will result in good call quality. Because the call is digitally encoded between transmission and reception (or as far as the PTSN relay) you will find that the call will be indistinguishable from a traditional phone call. There are two different codecs. One codec gives better quality but needs more than twice the bandwidth. This flexibility means that VoIP can be used by anyone, regardless of connection quality.

VoIP is the answer for any call centre seeking to reduce costs and maximise productivity without compromising service and quality.

Q.  How reliable are call centre VoIP systems?

A.    We install every VoIP system with layered redundancy, giving you a service which is as reliable as any traditional system. There will be the need for scheduled maintenance to keep the system running properly, but this can be carried out after hours.

Q.  How important is good bandwidth?

A.    Your available bandwidth is an important factor. As broadband (ADSL) offers asymmetric speeds, the speeds available to upload and download information can be very different. Typically, the upstream (from your PC to the internet) connection can be as little as one eighth of the speed of the downstream (from the internet to your PC).

A typical ADSL line will suffice for an office with up to four lines. However, a high quality ADSL+ line would be sufficient for more than 30 phone calls to be carried out simultaneously. Of course, you can have more than one line, but you can get far more use out of each line.

Q.  Are the telephone units reliable?

A.    Yes. BlackBIT only supplies proven products which are supplied by reputable manufacturers.

Q.  Does the computer network need to be turned on to use VoIP?

A.    Not necessarily, as long as your internet connection is working. New VoIP phones are self contained.  

When a call centre is fitted out, a computer network is not necessarily needed for the system to run efficiently. The system can be configured so that each member of staff can report on the outcome of the call through the telephone itself. Messages can be recorded regarding the more complex aspects of the call and these are then tagged to the relevant data. This keeps all the activity together, even if your staff are spread across a wide geographical area.

Q.  How fast does the internet connection have to be?

A.    For high quality VoIP you need a reliable broadband internet connection such as dsl or cable.

The minimum bandwidth required for VoIP is 88 kbps in both the upload and download directions: BlackBIT can test this for you. It is best to limit VoIP usage to no more than 50% of your maximum internet bandwidth in order to ensure that you are always getting the best quality signal.

Q.  How does the voice clarity of VoIP compare to traditional telephony?

A.    The quality of VoIP meets or exceeds the quality of a traditional landline. The quality of your internet connection and your VoIP provider determine voice clarity.

Q.  What about VoIP security?

A.    VoIP is actually more secure than normal email or online shopping. It is important to remember that VoIP is built on already established equipment and applications.

Q.  What if the power fails?

A.    If the power fails, your internet connection, and therefore VoIP, will fail, unless you have a UPS (uninterruptible power supply) or generator backup facility because VoIP equipment is dependent on electricity. 

However, unless your business is located in an area where the electrical supply is unstable and prone to blackout, the known benefits of flexibility, functionality and low cost will far outweigh the unlikely downside of power failure.

Q.  How can our staff get trained in using VoIP?

A.    BlackBIT can provide training either on site or over the phone.

Q.  Is it possible to customise our specification?

A.    Yes, after carrying out a thorough survey of your needs, BlackBIT will establish the scope of the system and service. We can provide bespoke configuration as required.

IP telephony makes sense, in terms of economics and infrastructure. If you have any more questions, please call us on 01392 279999

VoIP for businesses

Q.  Why is VoIP so important?

A.    VoIP, internet telephony and converged networks bring voice technologies, telephone and data technologies and the internet together. In so doing they provide great benefits to users. Although cost savings are a big factor in switching to VoIP, the real benefits of VoIP will come from the convergence of voice and data thereby introducing new features and capabilities such as the ability for automatic forwarding of voicemail to email.

Q.  What do IP telephony, packet telephony, internet telephony and converged networks mean?

A.    These are different ways of describing what is essentially the same thing – VoIP. They all mean something slightly different to people working in the industry, but for the average end user they mean the same thing – bringing the telephone and the internet together.

Q.  How fast does my internet connection have to be?

A.    For high quality VoIP, you need a reliable broadband internet connection such as dsl or cable. 

The minimum bandwidth required for VoIP is 88 kbps in both the upload and download directions: BlackBIT can test this for you. It is best to limit VoIP usage to no more than 50% of your maximum internet bandwidth in order to ensure that you are always getting the best quality signal.

Q.  How does the voice clarity of VoIP compare to traditional telephony?

A.    The quality of VoIP meets or exceeds the quality of a traditional landline. The quality of your internet connection and your VoIP provider determine voice clarity.

Q.  What is a virtual PBX?

A.    One of the latest outcomes of VoIP technology is the virtual PBX or hosted PBX. Small and medium sized businesses can now have a sophisticated telephone system without significant investment in telephony equipment. In actual fact, the entire telephone system is operated and maintained by your VoIP service provider. 

A virtual PBX allows staff to work from their home, hotel or on their mobile phone – but they are still connected to the same office telephone system. Callers can be transferred or put on hold with music. Call conferencing can be set up or the phone answered by an automated answering facility which can direct callers to different departments.

Q.  Can our company get local telephone numbers in area codes other than our own?

A.    One of the great benefits of VoIP is that your company can have a telephone number assigned to it in any area code: your office does not have to be located there. Businesses often want to have a local presence in different cities with a local phone number for customers to call. 

Business VoIP allows a customer to call that local number which can nevertheless be answered by the company anywhere in the country.

Q.  Can our staff take their office phone number with them when they travel on business?

A.    Yes, provided they have access to a high speed internet connection. If they take an ATA (analogue telephone adapter) with them they can always be reached. 

Many VoIP business clients install a soft phone on their laptops so that they can make and receive calls no matter where they are.

Q.  Can emergency phone calls be made from a VoIP system?

A.    Yes. In the UK all VoIP systems have to have the capability to provide information to the emergency organisations in the event of an emergency call.

Q.  Can VoIP handle faxing?

A.    Not by itself. As fax was designed for analogue networks, it does not move well over a VoIP network. Fax communication uses the signal in a different way from regular voice communication.

When VoIP technologies digitise and compress analogue voice communication it is optimised for voice and not for fax. 

Businesses can always keep a dedicated PSTN fax line if they wish or use any of the available efax software clients that send faxes as e-mail attachments.

Q.  How can our staff get trained in using VoIP?

A.    BlackBIT can provide training either on site or over the phone.

Q.  What if the power fails?

A.    If the power fails, your internet connection, and therefore VoIP, will fail, unless you have a UPS (uninterruptible power supply) or generator backup facility because VoIP equipment is dependent on electricity. 

However, unless your business is located in an area where the electrical supply is unstable and prone to blackout, the known benefits of flexibility, functionality and the low cost of VoIP will far outweigh the unlikely downside of power failure.

Q.  What is Skype? Is this VoIP?

A.    No, Skype is a free software program that utilises a peer to peer Voice over IP application. Skype is an application driven by a Windows operating system. Skype may take over some of the resources of a more powerful computer so that it becomes a "node" and supports the calls of other Skype users. Skype does not use pure SIP and so it will not allow users to choose universal providers or equipment.

Q.  Can I call any phone number using VoIP?

A.    Yes. VoIP allows you to make and receive calls exactly in the same way you would using a landline.  You can call any PSTN number or mobile number using a VoIP system.

Q.  Can anyone call in to a VoIP phone?

A.    Yes. You will be able to accept calls from both PSTN (traditional landlines) and other VoIP users. 

Further, you can purchase a traditionally geographic number such as 0161 or 0113 which can be registered/directed to any IP PBX, IP address or hosted account in any geographic location worldwide.

Q.  Can we keep our existing telephone number(s)?

A.    Yes, existing numbers are ported from your current telephony supplier, a process which BlackBIT manages on your behalf as part of our service.

Q.  Which calls are free and which calls do we pay for?

A.    Any calls within your sites which are connected to your own network do not cost you anything. Calls that break out of your own network are charged at very competitive agreed rates.

Q.  Does our company still need a landline?

A.    Some VoIP users like to keep a landline in case of emergencies. Since VoIP uses an internet connection to make calls, power outages or surges could interrupt service.

Q.  Can international calls be made?

A.    Yes, a VoIP phone can place a call to any telephone with a dial tone, regardless of where it is located. 

VoIP calls are less expensive than international calls made via traditional phones.

IP telephony just makes sense, in terms of both economics and infrastructure requirements. If you have any more questions, please call us on 01392 279999

BlackBIT VoIP functionality

We have listed below some of the standard as well as optional features available with a BlackBIT VoIP telephony system

Block anonymous call (optional)

Your phone will not ring if the call does not show Caller ID. With this feature enabled, callers who withhold their Caller ID will not be able to call you. Instead, they will hear a recorded message suggesting that they remove the block on their Caller ID.

Block caller ID

Your phone number will not appear to the person you have called. 

Call conferencing

You can talk with two other callers at the same time – there is no need to end the first call, dial another number, speak to that person, end that call and then dial the original caller back. Call conferencing for up to ten people is available as an optional add on. 

Call forwarding

You can forward:

  • all your calls to another landline phone or mobile; or
  • calls when the phone is busy; or
  • calls when there is no answer; or
  • calls when there is no answer after a specified time delay.

Call history

Via the phone user interface, you can look up the last 60 calls you have answered, the last 60 calls you have dialled and the last 60 calls you have missed. We provide fully itemised billing each month, so you can see exactly who has been called, for how long and how much each call has cost.

Caller ID

This feature displays the identity of the person calling – you can then decide if you want to answer the call or let it go to voicemail.

Call transfer

You can transfer the call you are on to any other number (eg, mobile) by calling the new number and introducing the caller. If the intended recipient does not wish to take the call, you can speak to the first person again and take a message.

Call waiting

The system lets you know when someone is trying to get through. You can see their name and number enabling you to prioritise answering, or leaving the call to go to voicemail.

CRM system integration

Integrate customer relationship management systems for caller name population. Select a name in your address book and dial them with the click of a mouse. The VoIP phone will then place the call to the selected individual. You will always know which customer/supplier is calling you. You can highlight a name in Outlook, Act, etc and click to dial – no more need to type numbers into the telephone.

Call Recording

Calls can be recorded for future playback. Benefits include future reference, dispute resolution, training purposes. A message should always be played to callers informing them that their call is being recorded. Legal advice should be taken before implementing this feature.

Day/night mode

The systems can be programmed to operate differently according to time of day, day of week, date and day of month. This provides endless ways to ensure that the system operates according to your requirements.

Do not disturb

If you need some quiet time, you can turn on Do Not Disturb and all incoming calls will be blocked. Instead calls will go either to voicemail or to a message that you are not accepting calls at this time. Outgoing calls are not affected.

Find me/follow me

The follow me service is similar to a call diversion service. However, it has some big improvements over other call diversion services. When a call is placed on the follow me service, the recipient of the call has to press button 1 to accept the incoming call. The voicemail on the mobile will not make the sound associated with pressing 1, so the caller will never be put through to the voicemail on the mobile telephone. The original call can be sent to multiple telephone numbers in sequence, or all at the same time. The first to answer and press 1 takes the call. If nobody accepts the call, it falls back through to the BlackBIT VoIP voicemail system.

Interactive Voice Response (IVR) system

The system offers menu options in order that callers can be put through to individuals or departments. Many large organisations use menu systems. Such systems are guaranteed to give the impression of size, as well as delivering the caller to the appropriate department, and indicating the nature of the call.

Monitoring for training purposes

Calls can be listened in to by authorised parties. This feature has many benefits. It allows managers to listen in to their staff to promote good customer service. It allows new operators to be trained by hearing experienced operators dealing with calls. It allows a manager to assist an operator who is dealing with a troublesome caller. Legal advice should be taken before implementing this feature.

Personal address book

Up to 100 telephone numbers can be stored in the directory of each telephone handset.

Queue management

The system places callers in a queue. The next available operator then answers the next queued call. This system integrates fully with the IVR option. Call traffic is sometimes unpredictable. Sometimes the number of callers is greater than a finite number of operators can handle. This system will place callers into a queue which enables the operators to deal with the current call without being distracted by call volumes. Callers will be delivered to operators in order. Callers can be informed of their position in the queue.

Speed dial

For quick dialling to your most used number, up to nine numbers can be stored as speed dial numbers.

Voicemail with voicemail to email

Voicemail comes as standard, but it is not standard voicemail. BlackBIT voicemail is available to all at no extra charge. It is highly configurable and each voicemail message that you receive can be sent to your email as an audio file attachment meaning that you can keep the message for as long as you deem necessary as well as share it with colleagues for their input.

For general information on VoIP, go to VoIP glossary of terms and definitionsIf you need further technical information, please call us on 01392 279999

VoIP glossary

VoIP glossary of terms and definitions

ADSL

An Asymmetric Digital Subscriber Line is a form of DSL, a data communications technology that enables faster data transmission over copper telephone lines than a conventional voice band modem. ADSL is different from other forms of DSL in that the volume of data flow is greater in one direction than the other, ie, it is asymmetric. ADSL is marketed as suitable for “passive” consumers to connect to the internet because they usually want to use the higher speed direction for the "download" from the internet but do not need high speed in the other direction. Limiting upload speeds limits the attractiveness of this service to business customers.

Analogue

An analogue signal is a variable signal which is continuous in time and amplitude as opposed to a digital signal which is discrete.

ATA

An Analogue Telephone Adapter connects one or more standard analogue telephones to a digital and/or non-standard telephone system such as a VoIP based network. An ATA usually takes the form of a small box with a power adapter, one ethernet port and one or more telephone ports. By using an ATA, it is possible to connect a conventional telephone to a remote VoIP server. Since the ATA communicates directly with the VoIP server, it does not require a personal computer or any software such as a softphone.

Broadband  

Broadband in telecommunications refers to a signalling method that includes or handles a relatively wide range (or band) of frequencies.  Broadband is always a relative term, understood according to its context. The wider the bandwidth, the greater is the information-carrying capacity. In data communications an analogue modem will transmit a bandwidth of 56 kilobits per seconds (kbit/s) over a telephone line; over the same telephone line a bandwidth of several megabits per second can be handled by ADSL.

CLI

Caller Line Identification is a service whereby the caller’s phone number is transmitted when a phone call is made. With a CLI equipped phone, the number is displayed so that the person being called can see who is calling before answering.

CNG

Comfort Noise Generation is artificial background noise used in radio and wireless communications to fill the silence in a transmission resulting from voice activity detection or from the audio clarity of modern digital lines.

Codec

A codec is a device or computer program capable of encoding and/or decoding a digital data stream or signal. The word “codec” stands for 'compressor-decompressor' or, more accurately, 'coder-decoder'. A VoIP codec is used to encode voice for transmission access IP networks. Codecs generally provide a compression capability to save network bandwidth. Some codecs also support silence suppression, where silence is not encoded or transmitted.

DHCP 

The Dynamic Host Configuration Protocol is a network application protocol used by networked computers (clients) to obtain configuration information for operation in an IP network. This protocol reduces system administration workload, allowing networks to add devices with little or no manual intervention. The DHCP server ensures that all IP addresses are unique.

DDI

Direct Dial-In is a feature offered by telephony providers for use with their customers' private branch exchange (PBX) systems. The provider supplies one or more trunk lines to the customer for connection to the customer's PBX and allocates a range of telephone numbers to this line (or group of lines) and forwards all calls to such numbers via the trunk. As calls are presented to the PBX, the dialled destination number is transmitted, usually partially (eg, last four digits), so that the PBX can route the call directly to the desired telephone extension within the firm without the need for an operator. The service allows DDI routing to each extension while maintaining only a limited number of subscriber lines to satisfy the average concurrent usage of the customer.

DSL

Digital Subscriber Line is a family of technologies that provides digital data transmission over the wires of a local telephone network. DSL can be used at the same time and on the same telephone line as a POTS telephone, as it uses high frequency bands, while a POTS telephone uses low frequency.

DTMF

Dual Tone Multi-Frequency is a mode of telephone dialling in which each digit has a different sound when numbers on the telephone are pressed. Interactive telephone menus are able to work because the DTMF frequencies are standardised - a pair of frequencies is uniquely linked to a number (or # or *) on the telephone keypad. Each key-press produces two frequencies depending on the row and column of the key, hence ”dual tone”.

HTTP

HyperText Transfer Protocol is the protocol that powers the World Wide Web.

H.323

H.323 is a VoIP protocol that preceded SIP. Unlike SIP, the H.323 standard specifies the complete VoIP protocol and not just the signalling methods.

IP address

An Internet Protocol (IP) address is a numerical label that is assigned to devices participating in a computer network utilising the internet protocol for communication between its nodes, ie, connecting points at which several lines come together. An IP address serves two principal functions in networking: host identification and location addressing.  Any participating network device can have its own unique address.

IP phone

An IP phone or VoIP phone is an entity used to make telephone calls over the internet. An IP phone normally looks identical to a regular telephone but instead of connecting to the normal POTS telephone line jack on the wall, it connects into a router or wall jack using an RJ-45 ethernet connector.  It then becomes a fully operational phone with software onboard, provided by the switch or system.

ISDN

Integrated Services Digital Network is a circuit-switched telephone network system that also provides access to packet switched networks, designed to allow digital transmission of voice and data over ordinary telephone copper wires. The result is better quality and higher speeds than that available with analogue systems. Prior to ISDN, the phone system was seen as a way to transport voice only. The key feature of ISDN is the integration of speech and data on the same lines, adding features such as fax and video that were not available in the classic telephone system.

IVR

Interactive Voice Response is a phone technology that allows a computer to detect voice and touch tones using a normal phone call. The IVR system can respond with pre-recorded or dynamically generated audio to direct callers on how to proceed.  IVR systems can be used to control almost any function where the interface can be broken down into a series of simple menu choices.

Key system

A key system or key telephone system is a multiline telephone system typically used in small office environments. Key systems are noted for their expandability and having individual line selection buttons for each connected phone line. Some features of a private branch exchange such as diallable intercoms may also be present.

LAN

A Local Area Network is a computer network covering a small physical area, like a home, office, or small group of buildings.

Layered redundancy    

Layered redundancy is a hierarchical system which provides fault tolerance in the case of hardware or software failure or upgrade.

Leased line   

A leased line is a symmetric telecommunications line connecting two locations. Unlike traditional PSTN lines it does not have a telephone number, each side of the line being permanently connected to the other. Leased lines can be used for telephone, data or internet services.

LNP

Local Number Portability is the ability to transfer either an existing fixed-line or mobile telephone number to another carrier.

PBX also known as a PABX

A Private (Automated) Branch eXchange is a telephone exchange owned and/or operated by a private business and it operates as a connection between that business and the public switched telephone network (PSTN). It connects the outside telephone network to the internal telephones, fax machines and extensions within the business. Advanced features such as voicemail, hold, transfer, least cost routing, etc are also provided

PoE

Power over Ethernet is a technology used to transmit electrical power along with data to remote devices over standard ethernet cables in a network. It is useful for powering IP telephones where it would be costly to run power separately.

POTS

A Plain Old Telephone System means the voice grade telephone service that is still the basic form of residential and business telephone service in most parts of the world. It has been available almost since the introduction of the public telephone system.  In recent times more advanced forms of telephone service such as ISDN, mobile phones and VoIP have been introduced.

Protocols

In computing, a protocol is a set of rules used by computers to communicate with each other across a network. A protocol is a convention or standard that controls or enables the connection, communication, and data transfer between computing endpoints.  Protocols may be implemented by hardware, software, or a combination of the two. At the lowest level, a protocol defines the behaviour of a hardware connection.

PSTN

The Public Switched Telephone Network is the network of the world's public circuit switched telephone networks. Originally the PSTN was a network of analogue telephone systems but now the PSTN is entirely digital. The PSTN is largely governed by technical standards created by the ITU-T and uses E.163/E.164 addresses (commonly known as telephone numbers) for addressing.

Pulse dialling

Pulse dialling is the sequences of clicks still found on some very old telephones. It originated when, for the purpose of signalling, a rotary dial was integrated into telephone instruments.

Rate centre

A rate centre is a geographical area used to determine the boundaries for local calling, billing and assigning phone numbers. Typically, a call within a rate centre is local, while a call from one rate centre to another is a long distance call.

Router

A router determines the proper path for data to travel between different networks and forwards data packets to the next device along this path. Routers connect networks together to access the internet. They are available in both wireless and wired versions.

SIP

Session Initiation Protocol is used for establishing a communication pathway on an IP network. A SIP line supports communication as simple as a two-way telephone call or a multi-media rich exchange including web video conference, voice-enriched e-commerce, web page click-to-dial, instant messaging, virtual reality or online video games. SIP’s simple protocol and method of establishing and terminating communication over an IP creates scalability, extendibility, and reliability in multiple coding languages and has become the protocol of choice for signalling communication via VoIP.

SDP

The Session Description Protocol is used to define SIP message bodies for phones calls. SDP does not deliver media itself but is used for negotiation between end points of media type, format, and all associated properties. The set of properties and parameters are often called a session profile.

Skype

Skype (sky peer-to-peer) is a proprietary software application that allows users to make voice calls over the internet. Calls to other users of the service and, in some countries, to free-of-charge numbers, are free, while calls to other landlines and mobile phones can be made for a fee.  Skype provides an uncontrolled registration system for users with no proof of identity. This permits users to use the system without revealing their identity to other users: the displayed caller's name is no guarantee of authenticity. Skype does not provide the ability to call emergency numbers because it is not an "interconnected VoIP provider”.  

SMTP

Simple Mail Transfer Protocol is the most common protocol for electronic mail (e-mail) transmission across internet protocol (IP) networks.

Softphone

A softphone is a software program for making telephone calls over the Internet using a computer rather than a dedicated telephone. A softphone is designed to mimic the functions of a real telephone and often appears to look like a regular telephone. A user will normally connect a headset to their computer via their soundcard or usb port.

Switch

A network switch is a computer networking device that connects network segments. Network switches are capable of inspecting data packets as they are received, determining the source and destination of that packet, and forwarding it appropriately. By delivering each message only to the connected device it was intended for, a network switch conserves network bandwidth and offers generally better performance than a hub.

TCP/IP

The Transmission Control Protocol and the Internet Protocol were the first two defined networking protocols. Today's IP networking represents a synthesis of two developments that began in the 1970's, namely LANs and the internet, both of which have revolutionised computing.

UPS

An Uninterruptible Power Supply can power equipment when a location loses its power supply. The system has a built in battery which keeps network devices operational when the power goes off.

Virtual telephone number

A virtual telephone number is a telephone number without an associated phone line. Usually these numbers are programmed to be forwarded to either a VoIP service or to a different phone line, fixed or mobile. Virtual numbers are sometimes used in conjunction with mail forwarding services to create a virtual office in a remote place. Virtual numbers are especially appealing to technology companies (for technical support), exporters (to give the impression that the company is local) and service companies such as call centres.

VoIP

VoIP (also known as Voice over IP or Voice over Internet Protocol) is the routing of voice conversations over the internet or through any other IP based network. Voice over IP protocols carry telephony signals as digital audio, typically reduced in data rate using speech data compression techniques. SIP and H.323 are the two main standards for VoIP, although SIP is becoming the more popular due to its generic nature, and, unlike H.323, it was designed specifically for wide area internet telephony. There are also some custom protocols such as that used by Skype.

WAN

A Wide Area Network is a computer network that covers a broad area, ie, any network whose communications links cross metropolitan, regional, or national boundaries. The largest and most well-known example of a WAN is the internet. WANs are used to connect LANs and other types of networks together, so that users and computers in one location can communicate with users and computers in other locations.

For information on the features of BlackBIT VoIP, go to BlackBIT VoIP functionalityIf you need further technical information, please call us on 01392 279999