Monday, March 15, 2010

Going Green the Telecoms way

Last year, the 2009 United Nations Climate Change Conference (COP15) was held at the Bella Center in Copenhagen, Denmark, between 7 December and 18 December. The aim of the conference was to review the Kyoto protocol and develop a framework for climate change mitigation. Delegates, Ministers and Heads of State from all over the world attended the conference. The most paramount issue was on reduction of carbon dioxide emissions by industrialized countries, which are the major contributors. There has been a sharp acceleration in carbon dioxide emissions since 2000 to more than a 3% increase per year from 1.1% per year during the 1990s is attributable to the lapse of formerly declining trends in carbon intensity of both developing and developed nations. The direct emissions from industry have declined due to a constant improvement in energy efficiency, but also to a high penetration of electricity. In concluding the conference, a Copenhagen Accord was drafted by the US, China, India, Brazil and South Africa on December 18, detailing the actions to be taken in order keep any temperature increases to below 2°C. One aspect that may have not been captured in the COP15 conference was the role of the Telecommunications industry in mitigating the effects of climate change.

Travel industry – air travel, marine travel and car travel - is one of the sectors that has high rate of carbon dioxide emissions. According to British Airways, a 747-400 plane cruises at 576 mph (927km/h), burns 12,788 liters of fuel per hour, and carries 409 passengers when full. This means that on average each passenger burns approximately 31 liters of fuel per hour. What if there was a way of averting the burning of thousands of liters of fuel by travelers? For instance, in business traveling, traders need to travel to source for commodities, meet potential clients, close deals and attend trade fairs. With modern telecommunications technology business persons can replace the hassle of travel and carry out their duties without causing damage to the environment. One such technology is telepresence which integrates life-size, High Definition (HD) video with high-quality sound in a room setup, creating the feel of actually being in the same room as participants at other locations. The technology can be implemented anywhere in the world utilizing the global Internet Protocol (IP) network, as simple as walking into the conference room next door. Telepresence is also employed in other sectors such as education, health, military and government.

Telepresence is similar to video conferencing, only that it gives an immersive experience. It provides stimuli such that the user perceives no differences from actual presence of the counterpart. As the screen size increases, so does the sense of immersion, as well as the range of subjective mental experiences available to viewers. The stimuli depends on the application and bandwidth used. Consider a person watching television, for example, the primary senses of vision and hearing are stimulated giving the impression that the watcher is no longer at home. Similar capabilities give telepresence a level of visual and acoustic realism that encourages active usage. The quality of experience also may be influenced by room customization. While high-end telepresence users might have many of these added services, other users may have simple rooms outfitted with plug-and- play, high definition technology. Indeed, any room can be a telepresence room.

The fundamental methodology used in a telepresence system is digital compression of audio and video streams in real time. The audio and video signal is sampled and quantized, a process called encoding. This process results in a digital stream of 1s and 0s is subdivided into labeled packets, which are then transmitted over the global IP network. The receiving telepresence system decodes the digital stream to display the visual and generate the audio. The hardware or software that performs this compression is called a codec (coder/decoder). To create a vivid, compelling and natural experience a resolution of between 720p and 1080p at 30 frames per second is deployed, giving a crystal clear video stream. For audio standards-based wideband codecs are implemented to improve the voice quality carried over IP networks. Wideband codecs provide clearer, more lifelike voice communications and markedly improved intelligibility because of the additional voice data included in the audio stream. They also double the voice signal range, in the range of 30 hertz to 7000 hertz or higher, while using the same network bandwidth as narrowband codecs.

In designing a telepresence system, first decision is to determine if the telepresence traffic will be carried on an overlay network or a converged network. An overlay network is a new set of connections that parallel the current network, whose purpose to provide links that are dedicated to the telepresence application. However, a converged network utilizes one network for voice, data and video applications. A network engineer needs to evaluate his requirements and decide whether to use an overlay or converged network. This decision (overlay versus converged) is driven by the sophistication of Quality of Service (QoS) in the current network and often by the deployment schedule. It is much faster to get an overlay network running correctly than to get all the details of additional bandwidth and QoS working on the converged network. I would advise enterprises to start with this approach and work their way back to a converged network when they are ready.

The next step is to determine bandwidth requirements. How much bandwidth will be required to support the proposed telepresence suites? Telepresence systems have a range of bandwidths over which they will operate, with varying degrees of quality as a result. Run some tests with the vendor to determine what quality level you really need. Then lay out the network map and determine which Local Area Network (LAN) and Wide Area Network (WAN) links will need to support that bandwidth. Telepresence systems usually consume about 5 Mbps per screen for today’s technology. So a 3-screen system requires 15 Mbps of continuous network bandwidth when in use. Next steps are to ensure that QoS is properly deployed to guarantee high-quality transport for the telepresence video and audio streams. Interactive video conferencing is a real-time application, so it uses UDP and requires low loss, low latency and low jitter. Getting this wrong means displaying your network loss on 60” plasma screens to your top level executives.

This technology is however not popular in Kenya because of a number of challenges. Namely; high cost of equipment, high cost of bandwidth and lack of expertise in this area. The key requirement of setting up a telepresence system is to put in place sufficient bandwidth for the telepresence traffic, above and beyond what was required by the business before telepresence was installed. First, Wide Area Network service providers need to give clear Service Level Agreements (SLAs) that address the requirement for video to have very low loss and jitter, and latency that is reasonable given the geographic distances involved. Secondly, the cost of this bandwidth can vary widely across geographic regions around the world. The first logical approach is to find a WAN service provider with a sufficiently large footprint to be able to supply service to all the enterprise locations of interest. Such service providers may offer partnering agreements with additional service providers, crossing from network to network, and getting a real guarantee on the traffic quality and QoS parameters. Additionally, young Kenyan engineers and technicians must venture into this area and learn how to design and implement telepresence solutions. Foreign expertise will always be more expensive, as it is now, but not to discourage our brethren from other parts of the globe from practicing in Kenya.