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"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, however, 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.
VoIP Gateways:
VoIP technology allows voice calls originated and terminated at standard telephones supported by the PSTN to be conveyed 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 calling party 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 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.
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.
(A Representative VoIP Network)
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. 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 IP network used to support IP telephony can be a proprietary network, a network of leased
facilities, or even the 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, 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.
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.
Access charges are imposed by the local telephone company to allow long-distance carriers to
originate or terminate the local portion of each telephone call. 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 thanks to the reduced bandwidth required by the data packets associated with the call. And, finally, 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; i.e., the bandwidth required. 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 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.
Conclusion
Voice-over-IP technology allows the integration of voice and data communications, reducing costs and revealing new opportunities for both telecommunications service providers and corporate users. TecPhone, as a leader in
VoIP technology and IP telephony, is proud to offer a wide range of products and solutions to help its customers and users benefit from this new technology.
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