VoIP (Voice Over Internet Protocol) or Internet Telephony are general terms meaning to send voice as digital packets over the Internet (IP).

For enterprises, leveraging VoIP means folding internal voice communications onto the LAN/WAN data network thus resulting in a single network to manage. Furthermore, inasmuch as voice is turned into another applications “running” on the data network, new productivity-enhancing features can be devised. Also, with the use of a softphone or a WiFi phone, incoming calls can be forwarded (through the Internet) to the called party, wherever he is, as long as he is connected to the Internet (mobility).

Because Internet voice is digital, it may offer features and services that are not available with a traditional phone. If you have a broadband internet connection, you need not maintain and pay the additional cost for a line just to make telephone calls.

With many Internet voice plans, onecan talk for as long as one wants with any person in the world (the requirement is that the other person has an Internet connection). You can also talk with many people at the same time without any additional cost.

There are numerous benefits that are:

Economic: Decreased capital infrastructure and staff to support a single network (e.g. one network of cabling, wiring closets), Simplification of, and reduced cost of moves/adds/changes (MACs) of personnel, reduced long distance/toll charges (especially with multi-branch locations).

Increased Communication Functionality: More options available (e.g. unified messaging), Cost to implement options are low to nil.

Increased Productivity: Scalability allows the network to grow more easily, the flexibility allows for a dynamic changing user community.

Yes. It also depends on the agreement you have with your VoIP service provider. For instance: your company is located in San Francisco and calls New York a lot. You can have a New York area code and phone number which would incur no long distance charges calling New York.

Typically there is no limit to where you can call, just as a traditional phone. Though some service providers may only allow calls to other subscribers of their service. The call can be made to a local number, a mobile phone, to a long distance number, or an international number. You may even utilize the service to speak with more than one person at a time. The person you are calling does not need any special equipment, just a traditional phone will do.

Yes. VoIP users can definitely make and receive calls to/from PSTN lines. Any type of call (e.g. local, long distance, international, etc.) is possible.

For this to be true though, however, VoIP user need to register with a Service Provider(SP). The SP provides the bridge between PSTN and VoIP. See "What is a VoIP Service Provider?" FAQ below.

No. A traditional phone system operating through an office PBX would also be inoperative. In both cases, you require a power back-up system, like a UPS (uninterrupted power supply) which would provide power without any interruption when the power goes out.

No. You may still use your fax machine in the same way. Depending on the agreement that you have with your service provider, you might have many new fax features available through a software application running on your desktop.

Yes. either ATA (Analog Telephone Adapter), an IP adapter for traditional phone devices, and IP phones can be used instead of regular analog phones.

Yes, however, we suggest that your organization also keeps one analog or PSTN line to maximize your employee access to the 911 service.

Yes. It is possible for you to use your VoIP service wherever you travel as long as you have access to a broadband connection.

No. If your enterprise elects to use “soft phones”, which is a software application running on your pc or laptop, then your phone device would be replaced by the pc or laptop and would need to be on to communicate through it.

Yes. VoIP allows users to surf the web or send/receive emails while making and receiving VoIP calls. VoIP shares the broadband bandwidth with other computers and devices on your local area network (LAN) and prioritizes voice calls.

In a small office environment (where there are up to 15 extensions), a single high-speed DSL or cable modem connection to the Internet should be fine. Approximately 800Kbps/3Mbps upload/download. However, for larger offices, it is suggested that a T1 line or better be used for the internet connection.

The following equipment and services are required for VoIP deployment: High-Speed Broadband connection, IP Phones or softphone, or Analog Telephone Adapters (ATAs) and a VoIP Service Provider.

VoIP Service Providers (VSPs) are the next generation telecom service providers that provide interconnection between IP and PSTN networks. They allow call origination and termination between these two networks.

It depends on your preference and budget. An ATA allows you to use analog phones for VoIP. While this might save money, they are missing some of the one touch feature keys (e.g. transfer, hold, etc) and several other features.

The auto-attendant is the automated system that guides incoming callers to an extension without human intervention. It is possible for an organization to have multiple recordings and auto-attendants.

An IP-PBX (Private Branch Exchange) is a server-based application providing, in a VoIP environment, the common telephony features and services supported by traditional PBXs: call transfers, conference calls, voicemail, music-on-hold, auto call routing, and auto-attendant.

VoIP gateways are devices that take analog voice signals and convert them to IP for transport over a LAN (Local Area Network) or WAN (Wide Area Network).

Gateways are bridges between a VoIP network and the PSTN. While a company would leverage its LAN/WAN network to support internal voice communication with VoIP gateways are required to place/receive calls to/from the PSTN.

The term codec is a contraption of COder and DECoder. The codec is where the analog to digital transformation is performed. In the PSTN, the Pulse Code Modulation (PCM) scheme is used to generate a 64kbps stream. More “economical” coding schemes have been designed to reduce this stream to 32, 16, and even 8 kbps. The main ITU codec standards are G.711, G.722, G.726, G.728, G.729, G.723. This reduction in the number of kbps requires comes at the price of reduced quality of the voice signal, thus a required trade-off between bandwidth and quality of service.

In order to transfer voice (analog signal) over IP (internet), the voice message must first be digitized. Codec basically transfers voice into data in order to travel through the public internet network. Presently codec algorithms are designed to reduce the size of their voice packets to increase speeds of transmissions and reduce bandwidth loads on the network.

VoIP signaling protocols are used to begin and end calls. They carry the required information to locate end users, and negotiate assigned actions depending on the end user’s device capabilities. The following are common VoIP signaling protocols: SIP (Session Initiation Protocol), H.323, Cisco SCCP (Skinny Client Control Protocol), IAX (Inter-Asterisk Exchange), and MGCP (Media Gateway Control Protocol).

Bandwidth is related to the codec being used. The G.711 codec uses 64kbps of bandwidth. The higher transmission rate in the G.711 codec is a common misconception about VoIP. The G.729 Codec compresses voice to 8Kbps, while the G723.1 codec has the ability to compress voice at two separate rates i.e. 6.3kbps and 5.3kbps. It should be noted that packet overhead (typically 46 octets) will increase these respective bandwidths, so it is best to choose a compressed voice Codec to minimize this effect. For example, the total required bandwidth including the packet header is about 9.33kbps for G723.1 (5.3kbps) and about 10.40kbps for G723.1 (6.3kbps).

H.323 is a complex protocol and does not go through a firewall unless special software has been written for the firewall. SIP is a simpler protocol and can be routed through firewalls by opening specific port numbers in advance.

FXS and FXO stand for Foreign Exchange Subscriber and Foreign Exchange Office respectively. The FXS interface delivers POTS service from the local phone company (from the Central Office). The FXS interface provides dial tone, battery current, and ring voltage to the subscriber device. The FXO interface receives POTS service from the local phone company (from the switch at the CO). The FXO interface provides on-hook and off-hook indication.

Gatekeepers and registrars are gateways whhich provide authentication, authorization, call control, call routing, and session invites.

The Federal Communications Commission (FCC) has worked to create an environment to promote competition and innovation that will benefit consumers. Historically, the FCC has not regulated the internet or the services provided over it. The FCC decided on February 12, 2004, that a voice over internet service will be an unregulated information service. The FCC also started proceeds to examine the FCC’s role in this new environment of increased consumer choice. The FCC organized an Internet Policy Working Group to identify, evaluate and address policy and issues that will arise as telecommunications services move to Internet-based platforms. For more information on the FCC visit www.fcc.gov/ipwg.

No, they do not but they have taken a preliminary position to be finalized in the third to fourth quarter of 2005.

In April 2004, the CRTC launched a hearing process towards a regulatory framework for VoIP communications in Canada. In the abstract of its Telecom Public Notice 2004-2 (http://www.crtc.gc.ca/archive/ENG/Notices/2004/pt2004-2.htm) , the CRTC states “The Commission is of the preliminary view that voice communication services using IP that utilize telephone numbers based on the North American Numbering Plan and provide universal access to and/or from the Public Switched Telephone Network (PSTN) (referred to in this public notice as "VoIP" services) have functional characteristics that are the same as circuit-switched voice telecommunications services. In the Commission's preliminary view, its existing regulatory framework should apply to VoIP services, including its determinations related to forbearance.” Currently, the CRTC is expected to issue its ruling by mid-may 2005.

Voice quality is most commonly rated through a voice quality metric called the Mean Opinion Score (MOS) The MOS is a 5 point scale where 5 represents excellent voice quality and 1 represents bad voice quality. A score of 4.0 or greater is considered suitable for a voice call. The name MOS comes from the fact that voice quality is measured by using people listening to sounds in headphones and rating the quality of these sounds from 1 to 5.

The Internet was not designed to carry voice traffic. Indeed, the Internet was designed to carry traffic between computers (and PCs). This data traffic has attributes that are opposite to voice traffic. Data is mostly connectionless while voice is connection-oriented; data traffic is insensitive to delay while voice is very sensitive to delay as well as to delay variation (jitter). When voice and data traffic share the same IP network, control is exercised to ensure voice quality, and provide priority to the voice packets.

Several QoS mechanisms have been designed to manage traffic classes with different service requirements. These mechanisms are used currently in enterprise networks as well as at the boundary with the Internet.