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VoIP technology : applications and challenges
Chakraborty T., Misra I., Prasad R., Springer International Publishing, New York, NY, 2019. 239 pp. Type: Book (978-3-319955-93-3)
Date Reviewed: Feb 18 2020

Many of us are familiar with voice over Internet protocol (VoIP) technology, having replaced our traditional landline telephones with VoIP phones connected through a data network. Modern routers are often equipped with analog sockets so users can plug their existing analog telephones directly into them. What some of us may not realize, however, is that our long-term evolution (LTE) cellphones can also use a voice over LTE (VoLTE) system that transmits conversations in an arrangement that uses a similar packet-switched VoIP technology mechanism. VoIP traffic is also carried over wireless local area networks (WLANs) in campus environments, worldwide interoperability for microwave access (WiMAX)-based networks, and cognitive radio networks (CRNs).

This book provides an introduction to VoIP fundamentals and discusses the special needs for carrying VoIP traffic over each type of network.

Following an introductory chapter, chapter 2 introduces and compares the H.323 protocol and session initiation protocol (SIP) for message control signaling. Color call flow diagrams assist with the explanations. The chapter also discusses the Megaco protocol used in media gateway controllers. Readers may want a more detailed guide to basic call setup and control mechanisms (for example, [1]).

As discussed in chapter 3, VoIP traffic usually has to compete for bandwidth with other services, and a quality of service (QoS) mechanism can be employed to maintain a standard quality of voice service for users. Items of particular concern are packet loss, out-of-order arrival, and jitter. The use of playout buffers and packet loss concealment mechanisms can control these to some extent. It is noted that adaptive codec switching is now being used to maintain quality as congestion increases.

Chapter 4 discusses opportunities and challenges related to VoIP over WLANs. The authors observe that using Wi-Fi services when available and otherwise using cellular connections can realize considerable savings. This can reduce the need for multiple subscriber identification module (SIM) cards when roaming in other countries. Some smartphones are now able to switch VoIP traffic seamlessly between WLAN and Wi-Fi services. The authors also note that security and authentication are of major concern for wireless networks, and packet loss can occur as users move between Wi-Fi access points.

In chapter 5, the authors observe that IEEE 802.11e defines a set of QoS enhancements for WLAN applications. They suggest that there is scope for optimizing the performance of IEEE 802.11b VoIP access points through the analysis and adjustment of buffer sizes, threshold limits, and retransmission power. A NetSim test bed is used to explore the potential benefits for both voice and video calls.

Chapter 6, “Optimizing VoIP in WLANs Using State-Space Search,” examines how adaptive performance optimizations can be implemented to cope with time-variant network scenarios. A state-space approach is used in a dynamic search of current network attributes. The chapter includes some screen shots showing video images with various percentages of recorded packet loss.

Subsequent chapters discuss some other mechanisms for improving the mean opinion score (MOS) values for VoIP calls. For example, chapter 7 describes some commonly used codecs and introduces an algorithm for using random early detection (RED) monitoring to adaptively switch the bit rate of a Speex codec. Jitter buffers have been used to mask the effects of the packet arrival time variation that can occur in congested networks. A small buffer may discard too many late packets, and a large buffer may increase the end-to-end delay to an extent that causes some irritation.

Chapter 8 introduces an adaptive jitter buffer playout algorithm. In a similar vein, it is possible to vary voice packet payload sizes to accommodate the sort of congestion seen when a link becomes choked by a traffic level that exceeds the capacity of its underlying network. Chapter 9 introduces an algorithm for the adaptive variation of voice sample frames in a VoIP call using a G.726 codec.

The final chapter provides an introduction to cognitive radio networks. Applications using these networks are able to access idle frequency slots in different spectrum bands by ascertaining which channels are being used by licensed primary users and then seizing other channels for secondary users (SUs). VoIP communications by SUs must therefore accommodate channel switching operations at unpredictable times. The authors present some findings of simulations in this field.

The content of this book is somewhat dated. Much of it relates to IEEE 802.11b WLANs, which have largely been replaced by 802.11n and 802.11ac equivalents that have different modulation and performance characteristics. There is no mention of the web real-time communication (WebRTC) services like Facebook Messenger that have increasing usage, nor of the modern codecs that they use, for example, OPUS. However, those who work in VoIP-related fields will find the algorithm development details invaluable.

Reviewer:  G. K. Jenkins Review #: CR146894 (2007-0149)
1) Hartpence, B. Pocket guide to voice over IP. O’Reilly, Sebastopol, CA, 2013.
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