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April 2007
Volume 10 / Number 4
Feature Articles

Border Codec Normalization is Table Stakes in VoIP Competition

By Matt McGinnis, Feature Articles

Since 1989, VoIP services have relied on one codec, or method of encoding and decoding voice into packets for transmission across IP networks. As VoIP usage increases with telco, cable, wireless, and Internet networks, there has been an explosion in the variety of codecs used to translate analog voice into packets. There are now more than 25 different codecs, each delivering optimum service for specific applications such as Voice over Broadband (VoBB), wireless, or cable. To remain competitive and ensure full connectivity for their customers, VoIP providers must support as many codecs as possible.

Codec transcoding solutions have become an important element to consider when choosing to support multiple codec types. With the wide selection of codecs between various networks, carriers must be able to “normalize” codecs at the network border to ensure compatibility with equipment in the backbone network. In this article, we’ll look at why transcoding is necessary, the requirements for implementing it in VoIP networks, and the options for where and how to deploy it.

Why Carriers Need Multiple Codec Support
Voice codecs are implemented within media gateways and customer premise equipment to convert analog and TDM voice to VoIP. The role of a codec within the VoIP network is to encode and decode voice for efficient travel on IP networks. Each voice channel (or call) requires network bandwidth, while the encoding/decoding process requires processing horsepower. Ideally, a codec would encode speech to perfectly represent the original analog waveform (as spoken by the individual) while using virtually no bandwidth and very little processing power. In reality, the choice of codec requires making a tradeoff between bandwidth and processing power to offer varying levels of voice quality.

Today, the G.711 codec is the most widely used codec in VoIP networks. Early VoIP providers realized that it was necessary to standardize on one codec to better facilitate transportation of calls across different networks. G.711 became the de facto standard because it represented, at the time, the best tradeoff between voice quality, bandwidth, and required processing power. G.711 offers a usable audible frequency spectrum from 300-3400Hz and has low processing overhead (0.2 MIPS per channel). On the other hand, G.711 requires a relatively high amount of bandwidth (64 Kbps per channel or up to 110 Kbps with the inclusion of IP headers). In many DSL deployments, total upstream bandwidth may be limited to 128 Kbps, so by using G.711, providers are limited to offering one VoIP line per customer.

As VoIP deployments expand, carriers have adopted alternative codecs to give them higher call density, lower bandwidth per channel, and higher quality voice (where quality is defined by a Mean Opinion Score or MOS). Figure 1 shows a list of codecs that are currently available. On one side of the spectrum, we can see that using the G.729 codec saves 88 percent of the bandwidth needed for a G.711 call (enabling the carrier to offer more lines per customer), but it requires 38 times the processing power and a slight reduction in voice quality. On the other side, new generations of wideband codecs are becoming adopted (iSAC, G.722, Speex WB, and others) and offer the promise of slightly reduced bandwidth with a much higher call quality utilizing the full audible frequency range of speech (50-8000 Hz). (See Figure 1.)

Figure 1

Figure 1: Codec names, target markets, and bandwidth.

The Role Of Codec Transcoding And Normalization
With carriers using a far wider variety of codecs than ever before, they must translate traffic from one codec to another as the traffic interconnects between two different codec types. Codec transcoding provides a means to convert traffic so that both sides can talk with each other without making codec format changes on the individual network devices themselves. The transcoding function is essentially a middleman that allows each side of a border (each provider) to use its codec of choice, but efficiently and effectively interconnect traffic at an appropriate point in the network. In order to reliably pass traffic among endpoints using different codecs, transcoding functions are essential.

For networks desiring support for a wide range of codecs from either the access network or from peering networks, a codec transcoding function can serve to normalize codecs, converting traffic encoding to a few selected codecs within a provider’s backbone. This provides numerous benefits ranging from simplified codec support requirements, decreased operational complexity, and quicker time to market. The transcoding location also becomes a central point where all codec treatment can be handled, allowing further flexibility with the addition of different codec types in the future.

Requirements For A Codec Normalization Solution
There are four key requirements for selecting a codec normalization solution.

Large list of supported codecs – Carriers need to support a broad range of codecs. With VoIP traffic from different wireline or wireless networks, carriers will need to offer support for over a dozen codec types to ensure proper call completion.

Lots of DSP (digital signal processing) power – The most important requirement for a codec transcoding solution is to have large amounts of DSP power. Because the carrier may not know which codecs will be processed at any given time, it must deploy codec transcoding capabilities that can easily scale and adapt to changing codec needs.

High density – Given the rapid growth in VoIP subscribers, carriers should be prepared to accommodate new users and processing demands, even with diverse codec support requirements. Ideally, a transcoding system should accommodate at least 40,000 calls per telco rack.

High availability – A transcoding system becomes a critical piece in the total end-to-end voice call, so it must offer at least telco-grade “five nines” reliability.

Transcoding Deployment Considerations
With these solution requirements, let’s look at the pros and cons of deploying transcoding in various elements within the network. Each variation offers a different mix of capabilities, with some offering stronger economic and operational benefits than others. There are four options for deploying transcoding in an IP voice network: media gateways, media servers, session border controllers, and media processing platforms.

Media Gateways – Media gateways must have enough MIPS to support the higher processing requirements of any new codec or codecs. However, since carriers have built networks to handle the G.711 codec only, it is often a rude awakening when they discover that processing performance in media gateways is somewhat limited, and that upgrading to support additional codecs often results in a decline in number of channels per system. To support diverse codecs from a media gateway, the carrier will also need to upgrade 100 percent of deployed ports, since each trunk group will require diverse codec treatment. This can be an expensive and cumbersome option.

Media Servers – Media servers typically reside in the backbone of the network and provide announcements, conferencing, and other special services to callers. While media servers have the advantage of being centrally located in a network, it is inefficient to have to backhaul each call requiring transcoding. In addition, most media servers are somewhat limited in the number of codecs they support, and are limited in the processing power available to support MIPS-intensive codecs.

Session Border Controllers – Session Border Controllers (SBCs) reside at carrier borders, most notably at the access or peering border. They are ideally located for codec normalization, but the key function of an SBC is to provide security and connectivity capabilities, and they are not optimized for scalable codec transcoding. These systems typically lack DSPs and are limited in their available MIPS resources for transcoding functions. It is possible to upgrade SBCs so that they can offer transcoding, but SBC architectures would then limit the density of SBC services that could be carried on the same system.

Media Processing Platforms – A media processing platform is a relatively new type of system designed specifically to handle transcoding and voice quality functions. It offers very high density and processing performance and can be deployed as a centralized resource for all existing media gateways, media servers, and SBCs in the network. Because the media processing platform can support an entire network, it is far less disruptive to deploy and less costly to install and support than any of the other options for transcoding (see Figure 2).

Figure 2: Border codec normalization using a media processing platform.

With growing acceptance of IP voice on wireless networks as well as LANs and WANs, the world is rapidly adopting a wide array of codecs to optimize bandwidth, density, voice quality, and costs in these network scenarios. To ensure reliable traffic exchanges around the world, carriers must adopt some form of transcoding solution. Media processing platforms represent the simplest, most powerful, and most cost-effective solution for adding robust and scalable transcoding functions into a network.

Matt McGinnis is Director of Product Marketing for Ditech Networks. McGinnis is responsible for defining exciting carrier applications and communicating their value to the market.

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