|Topic:||VoIP over WiMAX and wireless broadband|
Tzvika Friedman is the CEO of Alvarion, a major provider of WiMAX and wireless broadband network equipment for carriers, ISPs and private network operators. Mr Friedman previously served as Alvarion’s President and Chief Operating Officer. Prior to joining Alvarion he held senior roles at ECI Telecom. Mr Friedman earned a Bachelor’s and then a Master of Science degree in Electrical Engineering, Summa Cum Laude, from Tel Aviv University, and graduated with distinction with a Master’s in Science from the Sloan Program of Management, at the London Business School.
Today, with the advent of Internet Protocol, IP, networks and Voice over IP, VoIP, competitive carriers and service providers are looking to standards-based WiMAX networks to offer economically voice and data in the same package. Operators in Europe have already started these services and those in the Middle East and Africa are just waiting for regulators to establish the guidelines and license the service. Operators, however, need to plan carefully their systems to ensure the quality of service users demand.
Voice over Internet Protocol, VoIP, technology enables packet-based IP networks to carry digitized voice. With VoIP, competitive carriers and service providers can offer telephony for voice and fax services together with traditional data services over the same IP infrastructure and, by doing so, increase revenue and improve business models. VoIP telephony is an excellent technology for operators who have already, or are planning to build, wireless access data (WiMAX) networks. Operators already providing this service include Irish Broadband in Ireland and Iberbanda in Spain. Given the still large amounts of unsatisfied demand for basic voice services in Africa and the Middle East, VoIP over wireless is of great interest to both new and incumbent operators, who are looking into WiMAX as a way to accelerate broadband penetration in an environment where wireline technologies are limited by poor infrastructure and distance. Thanks to WiMAX and VoIP, people and areas with no existing telecom infrastructure will receive both voice and data services for the first time – and at the same time. No longer will operators have to decide between providing voice or data – both can be delivered over the same wireless broadband network. According to Adlane Fellah, of Maravedis: “As the interest level in the Middle East and Africa continues to grow for WiMAX-related products and services, regulators in Africa and the Middle East are nearing the final phases of what technical guidelines should be implemented in their countries, as well as which operators will be granted licences to deliver WiMAX products and services.” Constructing a VoIP telephony service over a wireless IP network, however, requires an understanding of VoIP technology, and the unique characteristics of the wireless medium, to maximize call quality and capacity. Of course, the all but ubiquitous mobile cellular networks have transmitted voice over wireless for many years, but in the beginning, these were analogue voice transmissions and even the current digital networks do not make voice connections using IP. Wireless Local Loop, WLL, systems – sometimes also called radio in the loop, RITL, or fixed-radio access, FRA, are examples of wireless access networks commonly used for voice. These fixed networks connect subscribers to the public switched telephone network, PSTN, using radio signals as a substitute for copper for all, or part, of the connection between the subscriber and the switch. WLL networks have been an efficient way to deploy basic plain old telephone service, POTS, for millions of subscribers – without the expense of burying, or stringing overhead, tons of copper wire over very long distances. WLL networks continue to be deployed and operate today in developing countries around the world, however, many are now being replaced by proprietary wireless broadband or standardised WiMAX networks. The reasons carriers give for building wireless networks in preference to wired infrastructure are numerous and have been pretty consistent over the years: 4 Low initial capital expenditures and very low network operating costs; 4 Operating costs proportional to customer base and associated capacity needs; 4 Ability to reach areas not serviced by existing infrastructure; 4 Low costs and fast speed to connect new customers to the network; and 4 Ease to upgrade and expand the network, including the ability to add network capacity through a simple upgrade to the backhaul (transmission between remote facilities to the central network) bandwidth. To this list, we can now add the ability to offer VoIP services for increased revenues as a simple increment to carriers’ and wireless Internet service providers’, WISPs, current wireless data networks, which are usually built for standard Internet access applications. VoIP goes wireless Because of the unique physical and operating characteristics of wireless networks, carriers wanting good-quality VoIP services over a wireless network need to do proper planning and choose the right VoIP equipment. With their own advantages and limitations, some of which can affect VoIP service performance, wireless IP networks have unique phenomena that may affect overall voice quality. To combat these issues, the choice of CODEC (which originally stood for ‘coder decoder’ but no one uses that term anymore), packet loss, latency, and jitter must all be examined closely and considered when designing a wireless network to handle VoIP traffic. CODECs, or coder decoders, are the devices necessary at each end of the voice connection to ‘code’ the caller’s voice into packets at the sending end and to ‘decode’ the caller’s voice at the receiving end. With the great advances in CODEC technology over the past decade, high-quality CODECs contained on a single chip are widely available and affordable. The selection of a high-quality CODEC – especially one optimized for the high performance necessary in wireless networks – is critical. High-quality audio CODECs are engineered to overcome packet losses, and most CODECs perform one or more functions to make a packet loss unnoticeable to the user. For example, a CODEC may choose to substitute the packet last received for the lost one or even perform sophisticated interpolations of the entire packet stream to “guess” at the contents of the lost packet in an attempt to eliminate interruptions in the audio stream – which manifest themselves as ‘clicks’ during the user’s call. Packet loss occurs when packets are not properly received at their destination and so are not available to the receiver. Packet loss can be caused by many different factors, including overloaded links, excessive collisions during transmission, physical media errors due to interference or low link quality, and others. Substantial packet loss degrades voice quality. Packet loss starts to be a real problem when the percentage of the lost packets exceeds a certain threshold, usually around five per cent of the total packets when randomly distributed, or even less when packet losses are grouped together. In those situations, even the best CODECs are unable to hide the packet loss from the user resulting in degraded voice quality. With wireless networks transmitting in so called ‘free space’, interference and other wireless phenomena cause lost packets to be much more common than in wireline networks. So, wireless networks extensively use mechanisms such as retransmissions or automatic repeat-request to minimize this phenomenon. However, this comes at the potential expense of increasing latency and jitter in the call. Jitter, historically a pejorative term for random delays in voice transmission time-slots on a wireline network, can be used in a positive way to increase VoIP performance in both wireline and wireless networks. In voice communications, voice packets are generated at a constant rate, so the voice decompression algorithm at the receiving end expects them to arrive at a constant rate. Delays inflicted by the network, however, may be different for each packet and that is jitter. To restore the fixed spacing between the packets, the typical solution is to implement a jitter buffer within the VoIP gateway to compensate for any jitter, or random packet delays, which may occur as the packets traverse the network. By deliberately delaying incoming packets in order to present them to the decompression algorithm with a fixed spacing, the jitter buffer also provides the time to process and fix out-of-order errors by examining the sequence number in the Real Time Protocol, RTP, frames. This has the effect of smoothing the packet flow and increasing the resiliency of the CODEC to packet loss, delayed packets, and other transmission effects. The downside of the jitter buffer, however, is that it adds delay into the network, so it must be designed carefully. The key is to obtain the benefits of the jitter buffer without introducing enough additional latency into the system to disrupt calls. Latency is a term that refers to the time a packet takes to cross the network, from its origin to destination including traversing the various network devices in between such as gateways, switches, loop carrier systems, etc. Being synchronous and real time, phone conversations are quite sensitive to latency with most callers noticing round-trip packet delays over 250 ms in the form of clicks, pops, and dropped calls. Given the significantly lower design criteria for one-way latency, the main sources of latency must be understood and examined closely. Beside the size of the jitter buffer discussed above, other network elements that introduce the most latency are: 4 CODECs: the coding and decoding of every packet, including compression, increases latency. For example, the G.723 protocol used for VoIP adds a fixed 30 ms delay. Different CODECs introduce different amounts of latency depending on their quality; 4 Network Latency: overall network latency can be controlled somewhat through packet and link prioritization. For example, voice traffic can get a higher priority through the network to ensure minimum latency is introduced. 4 Network Design: correctly configuring and building the wireless IP network, good telephony quality can be achieved and overall voice and data capacity can be increased. The WiMAX Forum The WiMAX Forum has created service quality types that are relatively similar to existing quality types for wireline packet service, quality types such as best effort, available bit rate, variable bit rate and constant bit rate. They have defined how much latency or jitter is acceptable. QoS was not part of the very first wave of WiMAX forum certification testing, but a number of vendors have been adhering to these quality expectations for a while. Choosing VoIP equipment that can support all the right functionality can have a critical impact on the capacity and performance of the telephony application over the wireless network. By optimizing both the VoIP equipment and the wireless infrastructure, voice capacity can be increased significantly and make a difference that can absolutely improve the business model and turn the application into a very worthwhile revenue generator. Since WiMAX and other broadband wireless networks will be built for multiple services, VoIP applications will be designed from the beginning to make sure that they can coexist with the much bigger bandwidth needed for data and video services. Thanks to the evolution and development of VoIP over WiMAX, people throughout Africa and the Middle East will soon begin to receive voice and data services at the same time – and for the first time.