VoIP: A Technology Primer

VoIP (Voice-over-IP) technology enables the real-time transmission of voice signals as packetized data over IP networks. 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.

Galaxy Telecom made it its mission to build a next-generation, converged voice and data network from the ground-up. This approach enabled us to build a fully interoperational, standards-based IP network, which is build on the most up-to-date innovative IP technology and protocols. This gives Galaxy Telecom 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.

Making a VoIP call

Support of VoIP calls 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 travelling 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 or 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. Galaxy Telecom’s VoX service is the most simplified and intelligent of these solutions, allowing a call without a computer, router, dedicated microphone or phone. The VoX has built in intelligence, therefore only requiring a standard telephone to access the network.

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.

Galaxy Telecom has developed a fully managed multi-pop IP network which connects to numerous terminating carriers across the world.