Digital voice market needs novel apps

Meanwhile some operators, especially the large U.S. incumbent and international carriers, face a rapidly growing problem of equipment obsolescence. Some of the "modern" digital switching systems and access concentrators are more than 20 years old and approaching the end of their design life. The problems this brings for the carriers range from structural failure of aged printed-circuit boards to withdrawal of technical support for some older switch designs-even by the large equipment vendors. The last thing these carriers want to do right now is to replace infrastructure with yet more narrowband equipment.

Although VoIP has already been deployed successfully in the carrier backbone and enterprise networks, there is much yet to do in order to implement VoIP in the access network. New chip sets and soft switch/media gateway standards now allow VoIP to be used to provide service to narrowband voice customers. This enables carriers to deploy VoIP-based solutions to resolve the end-of-life problems that will begin to bite at service providers in the next few years. It also allows carriers to migrate large portions of the PSTN over to a true packet-based infrastructure.

Challenging environment

The access network is a much more challenging environment for VoIP than the backbone for many reasons. Bandwidth in the access link is limited, even with digital subscriber line (DSL) technology. Equipment is distributed over many thousands of sites, which makes incremental updates requiring multiple engineer visits prohibitively expensive. Converged VoIP and data services potentially expose the voice network to much greater risk of malicious attack than when the networks are disjointed. Finally, there are widely different scalability requirements on access concentrators-for example, demand in central London is far greater than on a Scottish island.

These issues give rise to a variety of technical challenges, including ensuring that voice takes precedence over data, jitter and packet delays requiring echo cancellation, the security of the voice infrastructure and interoperability of equipment.

VoIP has been around for a number of years now, especially in enterprise networks. Many carriers are already familiar with issues such as quality of service and bandwidth utilization. In the past year a number of technological and standardization developments have rendered VoIP much more applicable to access networks.

One of the most significant changes is the recent development of chip sets that are capable of driving both the broadband and baseband frequencies on a copper loop. Such chip sets enable a single line card to support narrowband voice or xDSL on any line without the need for splitters. The baseband voice traffic can be encoded to a VoIP stream for the same packet-based access network that is used for the data traffic, eliminating the need for a parallel time-division multiplexed (TDM) link into the traditional PSTN infrastructure. This provides obvious savings in terms of the single converged network, and simplifies stock and maintenance procedures by providing a universal line card for POTS, ISDN or xDSL services.

Similar advances in DSP technology are dramatically reducing the power and pc-board real estate that trunk gateways require to provide echo cancellation, transcoding between G.729AB and G.711 codecs and silence suppression. This means it will soon be possible to build CompactPCI form-factor IP-to-TDM gateway cards with all the digital signal processing needed to support more than 2,000 voice channels, each at maximal compression.

All these silicon advances would be of little use without standardization of the protocols used in the network, particularly in relation to signaling between customer premises equipment and the operator's network. Here, too, significant progress has been made, with standards bodies like the Multiservice Switching Forum, International Softswitch Consortium, OpenVoB and CableLabs now promoting variants of the following architectures:

• DiffServ and 802.1p for prioritization of voice within the core network.

• H.248 (or its predecessor, MGCP) for control of integrated access devices, access concentrators and trunk gateways.

• SIP-T or BICC for intercontroller signaling.

• SIP signaling for soft clients and IP phones.

• IPSec to provide authentication between network elements.

By combining the above technologies and protocols, it is possible to deliver a truly converged access network infrastructure that is, in effect, a new broadband-based paradigm for the PSTN. Such solutions rely on two main components: voice-capable DSL access multiplexers (DSLAMs) that support broadband and narrowband services simultaneously on each line, and flexible next-generation voice switching platforms that provide existing PSTN services for VoIP subscribers. These new voice switches incorporate a call agent that supports H.248 control of gateways in the IP network, SS7 protocols to deliver voice traffic into the existing PSTN and SIP-T for signaling between call agents.

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With this equipment, end users with nothing more sophisticated in hand than a normal analog telephone can be part of an end-to-end VoIP network-without even knowing it. For the carrier, there's the added benefit that enabling DSL access requires only a configuration change to give that customer full broadband VoIP.