Voice over IP-the winding road ahead

The level of excitement, coverage and investment in voice-over-IP services is growing daily.

>From telecommunications and IP equipment vendors and application developers to traditional telecom and non-traditional IP service providers, there appears to be almost universal interest in enabling voice transport on existing and new IP networks, and in internetworking IP voice with the public switched network.

Driven initially by the potential for call-savings, the long-term reasoning behind the voice-over-IP bandwagon is simple: Voice over IP opens the door for integration of voice, video and data on a single IP-based network and for producing significant cost reductions related to network devices, infrastructure, operations and network management. Even more important, it enables development of whole new categories of applications and services.

Voice over IP has additional advantages. Users connected to the public network can now make use of services residing in IP networks without changing the end user infrastructure. This increases the potential customer base far beyond the relatively small number of PC users connected to IP networks today.

Despite the current appeal of IP as a universal panacea, however, successful deployment of commercial voice-over-IP services will require that existing IP networks and network devices be significantly enhanced with new functionality. This includes mechanisms that support interworking with the public network on the transport and signaling level, ensure the quality of service required for real-time voice delivery, enable IP signaling and permit customer billing for service usage. Fortunately, voice over IP offers strong economic and competitive incentives to tackle these issues.

Building blocks Whether for real-time voice services or less time-sensitive voice mail and faxes, new gateways will be needed to interconnect IP networks and the public network as well as support communication between PCs and traditional telephone devices, including fax machines. These gateways will need to handle interworking between the IP network and the telephone network on the transport and signaling level. The ability to distinguish between real-time voice traffic, less time-sensitive faxes or non-real time voice mail will be critical to choose the most effective and economical route through the network.

The ITU-T has described how voice should be carried over packet networks in its H.323 recommendation. The Internet Engineering Task Force, Bellcore and the Technical Advisory Council have also defined how to carry voice over IP. Bellcore and the TAC have submitted their proposal to the IETF and ITU-T.

All are based on one common network architecture: a gateway that provides connectivity between the telephony and IP networks, translating audio and signaling, and a gatekeeper to provide essential services such as address translation between phone numbers or names and IP addresses, call authentication and other value-added services.

Gateway and gatekeeper functionality, complexity and design depend on whether they will be implemented on enterprise or public carrier networks. Public carrier gateways and gatekeepers must also handle complex tasks such as interworking with SS7, collecting billing data and integrating with other business and operations support systems. They must also meet stringent telco reliability and availability standards.

Carrier challenges For carriers seeking to offer "commercial-grade" voice over-IP-services, quality of service (QOS) issues represent one of the largest challenges. Delay and delay variation must be held at acceptable levels, even though today's IP networks were designed to deliver delay-insensitive data. It is also important to minimize packet loss. Today's IP networks must be enhanced to provide controlled and predictable classes of service that ensure that routers assign the network resources required to deliver acceptable voice quality.

Different router mechanisms can be used to meet this objective. These include various prioritization schemes, such as weighted fair queuing and router flow control mechanisms such as the IETF's multiprotocol label switching tagging scheme or use of type of service bits in the IP header. All these mechanisms require prior configuration by a network administrator, who must decide what priority and resources to provide for each specific service class.

A more dynamic alternative for assigning resources is the resource reservation protocol, which permits a voice terminal or voice gateway to request a specific IP QOS.

Regardless of which of these is used, a deeper problem remains. QOS is defined on an end-to-end basis, and therefore requires that sufficient network resources be provided throughout the entire network path. This is not an overwhelming issue for an enterprise network or single service provider environment where all resources can be administered through one network manager. However, it is almost impossible to administer today when multiple service providers are involved, as is the case with most national or international long-distance calls.

In addition, this fulfillment of QOS assumes that all routers in the network are equally capable of identifying voice traffic and providing the network resources required.

This is still the exception rather than the rule in today's IP networks because standards for many of these mechanisms have not been finalized and implemented by all router manufacturers.

Service providers will need to set up peering or service level agreements to deliver end-to-end QOS across multiple networks. These agreements will require constant monitoring and validation.

Signaling plays a critical role in establishing connections and providing enhanced services in today's voice networks. In circuit-switched long-distance networks, the call connection is only established end-to-end when the far end actually answers the call. Network resources are then assigned for the duration of the call. This is a new concept for connectionless IP networks, in which IP traffic is submitted into the network and transported, independently of whether the receiver is actually available or even reachable.

ITU-T recommendation H.225 defines the signaling protocol used to establish logical channels through the IP network for video, audio, data or control information between terminals and gateways. Signaling is used only between the H.323 terminal and the gateway or another terminal and is transparent to the IP routers involved.

Interworking with existing voice networks is the key to success for a commercial voice-over-IP service. Gateways must successfully translate IP signaling into the large variety of signaling protocols used in analog and digital wireless and wireline networks. In addition, different national variants of these signaling protocols are in use.

The most critical aspect of carrier-class signaling interworking is communication with SS7, which handles communications between nodes in the long-distance and international networks. Interworking with SS7 will be needed if voice-over-IP networks are to offer the same value-added services as today's telephone networks, such as 800 numbers and calling card services.

Carrier control and user expectations Beyond the obvious need to provide charge records, usage-based billing capabilities provide important control capabilities for both users and network owners. Billing details permit users to evaluate the value of their service-class selections, while pricing mechanisms allow service providers to control network traffic by providing pricing incentives and disincentives. Commercial voice-over-IP deployment, based on differentiated classes of service, will require comparable billing capabilities.

Gathering the individual call detail data needed to generate a bill is far more complex in IP networks than on the existing telephone network. Public network service quality does not, for example, vary during a call because a specific physical link is dedicated to each individual call for its entire duration.

IP networks, on the other hand, are connectionless-individual packets could potentially take a different route through the network to reach the destination. Moreover, different applications share the same physical connection. All this makes it difficult to predict the QOS each individual packet will experience. The gateway, therefore, must monitor quality throughout the call to ensure that users indeed receive the service quality they pay for.

In addition, billing data collected and generated must be compatible with existing public network billing system to allow consolidation and exchange of information between different service providers and vendor systems.

Reliability and availability are critical to business and consumer customers worldwide. Our society expects uninterrupted service. There is little reason to believe that voice-over-IP users will not have these same expectations. As almost any IP user can attest, today's IP connections have a long way to go to begin to even approach this level and gain customer acceptance.

Ostensibly, IP networks are designed to automatically detect links or routers that go down and to reroute the traffic accordingly. However, during the process of rerouting, packets often get lost or links become so overloaded that traffic delivery slows noticeably. This is unacceptable for voice traffic, which requires uninterrupted service with a guaranteed service quality.

To achieve its goal, heavily used voice-over-IP links will have to be constantly monitored to switch traffic over to links that can handle traffic overflow loads in less than a few milliseconds. IP switches or routers will need redundant components or alternate systems that can provide backup in the event of failures.

Daunting as these challenges may seem, the economic rewards are compelling. Commercial voice-over-IP services, linked to the public network, open the door to more than 650 million telephone users and more than 200 million mobile phone users, compared with the 100 million PC users that currently have access to IP networks.

Interim steps that validate this concept and improve acceptance are already being taken. New and existing long-distance and international service providers are using IP networks today to provide IP telephony services at attractive rates.

Over-provisioning or the use of vendor-specific prioritization schemes will ensure acceptable QOS for early adopters. Enterprises are investigating and even deploying IP voice into their data networks, sometimes starting with less delay-sensitive fax and voice mail traffic. Software developers are creating new reference applications for businesses that demonstrate how integration of IP voice and data can provide powerful business tools for call centers and marketing and customer service departments.

Voice-over-IP networks will not replace the existing telephone network. If they can provide improved functionality and reliability approaching the public network, however, they will create new opportunities to provide enhanced services to PC and traditional telephone users, incorporating not just voice, but true multimedia.