FAQ
A Guide to VoIP
"Voice-over-IP" (VoIP) technology enables the real-time transmission of voice signals as packetized data over "IP networks" that employ the Transmission Control Protocol (TCP), Real-Time Transport Protocol (RTP), User Datagram Protocol (UDP), and Internet Protocol (IP) suite.
In VoIP systems, analog voice signals are digitized and transmitted as a stream of packets over a digital data network. IP networks allow each packet to independently find the most efficient path to the intended destination, thereby best using the network resources at any given instant. The packets associated with a single source may thus take many different paths to the destination in traversing the network, arriving with different end-to-end delays, arriving out of sequence, or possibly not arriving at all. At the destination, the packets are re-assembled and converted back into the original voice signal. VoIP technology insures proper reconstruction of the voice signals, compensating for echoes made audible due to the end-to-end delay, for jitter, and for dropped packets.
While standards for VoIP technology are emerging, they are still in flux. Even VoIP implementations that are standards-compliant may not necessarily interoperate with the standards-compliant products of other vendors. The ITU-T H.323 standard, for example, does not encompass all aspects of VoIP communications, and each vendor of VoIP technology can have their own variations of the overall VoIP network architecture and algorithms. Variations among VoIP products include the algorithms and implementations used to support dynamic bandwidth allocation, packet loss recovery, adaptive echo cancellation, and speech processing to deliver voice quality as high as possible.
Digifonica has designed and is building a next-generation, converged voice and data network from the ground-up. This approach enabled us to build a fully inter-operational, standards-based IP network, which is build on the most up-to-date innovative IP technology and protocols. This gives Digifonica the ability to take advantage of any new developments in the field of VoIP at a speed, a cost and with flexibility not available to competitive service providers.
VoIP Gateways
VoIP technology allows voice calls originated and terminated at standard telephones supported by the PSTN to be transported over IP networks. VoIP "gateways" provide the bridge between the local PSTN and the IP network for both the originating and terminating sides of a call. To originate a call, the caller will access the nearest gateway either by a direct connection or by placing a call over the local PSTN and entering the desired destination phone number.
The VoIP technology translates the destination telephone number into the data network address ("IP address") associated with a corresponding terminating gateway nearest to the destination number. Using the appropriate protocol and packet transmission over the IP network, the terminating gateway will then initiate a call to the destination phone number over the local PSTN to completely establish end-to-end two-way communications. Despite the additional connections required, the overall call set-up time is not significantly longer than with a call fully supported by the PSTN.
The gateways must employ a common protocol -- for example, the SIP, H.323 or MGCP or a proprietary protocol -- to support standard telephony signaling. The gateways emulate the functions of the PSTN in responding to the telephone's on-hook or off-hook state, receiving or generating DTMF digits and receiving or generating call progress tones. Recognized signals are interpreted and mapped to the appropriate message for relay to the communicating gateway in order to support call set-up, maintenance, billing, and call tear-down.
VoIP Gatekeepers
The translation of a destination telephone number into the IP address of the correct terminating gateway is a primary VoIP "gatekeeper" function. The routing table maintained by the gatekeeper resolves which gateway corresponds to the destination telephone number in order to complete a call.
Gatekeeper functionality can be distributed among all the gateways of the VoIP network or can be centralized at one or several locations. When gatekeeper functions are embedded in each gateway, all gateways of the overall VoIP network act autonomously to coordinate their actions. With a centralized gatekeeper, all gateways of the network coordinate their actions with respect to the centralized gatekeeper rather than acting independently.
The Digifonica Network design builds in these gatekeeper functions into its Global network and is specifically designed to minimize the risk of congestion and service delay at a centralized system by distributing these functions throughout the network in a proprietary fashion.
VoIP Networks
Support of VoIP calls thus generally requires at least two VoIP gateways. Typically, a VoIP service provider would establish gateways (or relationships with other service providers and access to their gateways) in all countries or regions for which calls are to be originated and terminated. The resulting VoIP network is composed of the gateways, the local PSTN access to each gateway, and the IP network that links the gateways.
Access to the local VoIP gateway for originating calls can be supported in a variety of ways. For example, the PBX of a business can be configured so that all international direct dialed calls are transparently routed to the nearest gateway. In this way, high-tariff calls are automatically supported by VoIP to obtain the lowest cost. These implementations require expensive customer premise equipment and costly service set-up, therefore they are usually only adopted by large enterprise customers with a network of distributed branch offices across the country or world.
Alternatively, the calling party may be required to dial a local or toll-free number to access the nearest gateway and then enter a Personal Identification Number (PIN) and the desired destination phone number. This approach is particularly well suited for VoIP service providers marketing their service with prepaid calling cards. The drawback is that these access numbers are cumbersome to dial. Also, they can only be used for international calls within the country of purchase, therefore when traveling into other countries the caller has to purchase different calling cards for each country and remember a different access number for each country.
Another method of accessing VoIP gateways is through “edge” devices. These devices allow access to either the public Internet or to a managed network. The most commonly “edge” devices require a computer, a router and a microphone or IP phone to interface with the Internet. Digifonica has numerous versions of these gateways in a variety of port densities, allowing a call without a computer, router, dedicated microphone or phone.
The IP network used to support IP telephony can be a proprietary network, a network of leased facilities, or even the public Internet. The Internet is clearly the most inexpensive underlying IP network, but, because it lacks any central administrative or controlling entity, it can be subject to congestion, uncontrollable packet delays, and temporary outages. More reliable communications, albeit at higher cost, can be realized with dedicated networks, either proprietary or leased. With guaranteed bandwidth availability and manageable Quality of Service (QoS), a dedicated network provides a more stable and high-performance medium than the Internet.
A proprietary network can be simply established using leased lines and owner-operated networking equipment. Alternatively, bandwidth on frame relay or asynchronous transfer mode (ATM) facilities can be affordably obtained from such international carriers as WorldCom/MCI, AT&T, Cable & Wireless, Sprint, and others.
Finally, calls must be terminated at a corresponding VoIP gateway and completed to the destination phone number via the PSTN or, in the case of a call internal to a company's virtual private network, its dedicated lines. Depending on the location of the gateway and the destination phone number, long distance charges may apply. Typically, the terminating gateway will be in the same country as the destination phone number or in a country with competitive tariffs so that favorable long distance rates can be obtained. Implementation of least-cost routing algorithms insures that a given phone call is terminated at the gateway that realizes the lowest total end-to-end tariff.
Digifonica has developed the only managed IP network with gateways that connect to terminating carriers across the world.
VoIP Cost Structure
Long distance and especially international voice communications can be significantly less expensive when supported by an IP network rather than by the PSTN. Calls supported by VoIP technology are not subject to the same cost structure of access charges, transmission costs, and settlement charges.
The local telephone company to allow long-distance carriers to originate or terminate the local portion of each telephone call imposes access charges. In the United States, however, the Federal Communications Commission has ruled that such access charges are not applicable to VoIP calls.
Transmission costs associated with the actual long-distance transmission are typically much less due to the cost of the reduced bandwidth required by the data packets associated with the call. In addition, the settlement charges associated with international calls are not present when international transmission is carried by an IP network.
A call supported by the PSTN involves the establishment and cost of an end-to-end circuit that is maintained for the duration of the call. A call supported by VoIP technology, by contrast, involves the transmission of many individual packets over an IP network. The cost of a VoIP call thus depends in part on the number and size of the packets that must be transmitted. Fundamentally, transmission costs equal the cost of the bandwidth used. Use of speech compression algorithms can reduce the required bandwidth by a factor of 8 or more. Further, bandwidth reductions can be obtained by recognizing and not explicitly transmitting the silences that naturally occur in human speech. These reductions in bandwidth directly translate to a reduction in cost.
The total cost per VoIP call is thus due to the costs associated with access to the gateways at both ends and the cost of transmission over the IP network. If originating calls access the VoIP gateway through the PSTN, access costs at the originating end may include the costs of local or long-distance connections or the monthly cost of a toll-free access number. Access costs at the terminating end may include the costs of the local or long-distance connections associated with terminating a call from the nearest gateway to the destination number.
The cost of transmission over the IP network depends on what IP network is employed. If the public Internet is employed as the underlying IP network, then the only cost is the cost of Internet access at each gateway. Costs are higher if a proprietary or leased IP network is employed, but, in return, the network can provide enhanced reliability and assured Quality of Service.
VoIP Delivery Options
VoIP can be delivered in various different ways using any combination of PC and Phone as the delivery “vehicle”. The most rudimentary and cheapest options are PC-to-PC, PC-to-Phone or Phone-to-PC, and Calling Card Phone-to-Phone calling. These options can most easily be accessed by consumers.
More sophisticated and higher quality options include voice over broadband Phone-to-Phone and Enterprise Private Network Phone-to-Phone calling.
PC-to-PC
Allows Internet users to exclusively utilize PCs to place or receive a phone call through the use of a software or hardware application. This method of communication requires that the user's computer have a microphone, speakers and sound card. However, besides just making phone calls, many of the PC-to-PC software phone applications also enable users to hold group conferences (3 or more people), share the same PC screen, or surf the Internet as a group. In addition, some PC-to-PC solutions also support video transmissions through the use of a PC camera.
Generally, you will need at least a 75 MHz computer system, with a sound card, microphone, speakers, a connection to the Internet and the software or hardware application that will enable you to place and receive calls over the Internet. Most of the Internet telephony applications will only allow you to communicate with other users of the same application, so both you and your call's recipient must be using the same Internet telephone application.
PC-to-Phone or Phone-to-PC
Allows Internet users to place a call from their PC to a traditional phone, or receive a call on their PC from a traditional phone. These forms of communication utilize software or hardware applications that can connect a phone to a computer or that connect between the phone and the wall jack.
Calling Cards (Phone-to-Phone)
This option allows Internet users to place an Internet call directly, from any traditional phone to any other traditional phone. With some solutions, a computer is not even necessary. This type of communication generally requires the user to open an account with an Internet phone company and pay per minute charges. However, there are some client products that will enable you to use traditional phones without paying for the service.
VoIP over Broadband and Managed Voice Solutions
This is a variation of the above VoIP calling options. Instead of using dial-up Internet access, a broadband connection is used to allow for better sound quality and increased access to next-generation services. Broadband Voice Services are the type of VoIP services provided by Digifonica . In most cases it requires the company to buy specialized routers, servers and phone equipment that is set up on their premises. Digifonica can provide all of these types of services to Enterprises Globally and can also add additional feature sets to the services such as voice mail, unified messaging, voice encryption, and many others.
