Research and trials for reliable VoIP applications

Computer Networks 52 (2008) 2447–2449

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Computer Networks journal homepage: www.elsevier.com/locate/comnet

Guest Editorial

Research and trials for reliable VoIP applications The Internet has burgeoned into a worldwide information infrastructure during the past few years, bringing about significant changes in the telecommunications arena. One of the new applications that has thrived is Voice over IP (VoIP), also known as Internet or IP telephony. Since its inception, huge strides have been made in quality and usability and now VoIP enjoys widespread popularity as an alternative to traditional telephony in homes and enterprises. Despite promises of advanced services, VoIP is still mainly a PSTN replacement service and is measured against the public switched telephone network (PSTN) in terms of quality-of-service and reliability. Unfortunately, despite advances, it still comes up short, typically because of challenging network environments. VoIP always uses internet protocols to transport voice signals between caller and callee and to establish and manage calls (‘‘signalling”). In some cases, the public Internet is used as a transport infrastructure. The combination of the use of widely deployed general-purpose protocols and the re-use of an existing infrastructure can offer significant functionality and cost advantages. For example, even for wired devices, phone numbers are no longer tied to a specific geographic location. Compared to the PSTN, it is fairly easy to add wideband audio, video, instant messaging and other media. Since end systems are built on general-purpose programmable processors and have access to other Internet services such as the Web, telecommunication providers and end users can more readily customize services to specific users and vertical markets. Sharing a single infrastructure can drastically reduce cost for wiring and switching bits, while providing more capacity for load spikes. Beyond those promises, however, the user experience and reliability of VoIP is still often inferior to that of classical circuit-switched PSTN services, particularly when the public Internet is part of the voice path. Varying degrees of packet loss, delay and jitter make for a challenging operating environment, in addition to problems related to network address translation (NAT), interoperability, and network stability that are beyond the scope of this special issue. In addition, IEEE 802.11 wireless networks were not designed with VoIP in mind, leading to additional problems for that class of access networks. Furthermore, the end-to-end security of VoIP is still an issue of concern. 1389-1286/$ - see front matter Ó 2008 Elsevier B.V. All rights reserved. doi:10.1016/j.comnet.2008.04.007

Motivated by the promise and challenges of VoIP, this special issue presents recent research activities addressing all aspects of VoIP communications. Of particular interest are studies that focus on the provision of VoIP over wireless links. We received 18 papers that were reviewed by experts in the field; six of them have been selected for publication. The first paper investigates the provision of QoS for VoIP applications over wireless networks by proposing multiple paths to deliver VoIP data destined for a particular receiver. CmpSCTP is introduced, a transport layer solution for concurrent multi-path transfer that modifies the standard stream control transmission protocol (SCTP). CmpSCTP exploits SCTP’s multi-homing capability by selecting several good paths among multiple available network interfaces to improve data transfer rate to another multi-homed device. Through the use of path monitoring and packet allotment techniques, cmpSCTP tries to transmit a number of packets corresponding to the ability of the path. At the same time, cmpSCTP updates the transmission strategy based on the real-time information of all the paths. Using cmpSCTP’s flexible path management capability, users can switch the flow between multiple paths automatically to realize seamless path handover. The theoretical analysis evaluated cmpSCTP and formulated optimal traffic fragmentation of VoIP data. Extensive simulations under different scenarios using OPNET verified that cmpSCTP can effectively enhance VoIP transmission efficiency and highlighted the superiority of cmpSCTP over the other SCTP’s extension implementations under performance indexes such as throughput, handover latency, packet delay, and packet loss. The second paper presents the results of VoIP trials with different commercial DVB-S/RCS satellite offers and popular Internet telephony and videoconferencing applications (Skype, MSN). These results reveal that packet delay and jitter are strongly affected by the satellite network component as well as the type of speech codec used. Accordingly, research presented in the second part of this paper is focused on a dynamic speech coding rate control adapted to the conditions of the underlying network, in which the satellite domain presents the most challenging portion of the end-to-end path. For this purpose, a novel cross-layer mechanism is proposed to facilitate and increase the accu-

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Guest Editorial / Computer Networks 52 (2008) 2447–2449

racy of the speech coding rate adaptation mechanism. Cross-layer design is a relatively new idea aiming to exploit information exchange among layers of the protocol stack. The simulation analysis shows that the proposed cross-layer mechanism can help to optimally adjust the speech coding rate to maintain the user-perceived quality in terms of mean opinion score (MOS) in the face of timevarying available satellite channel capacity. The third paper presents a scheduling algorithm for real-time packet-based phone conversations over wireless ad-hoc networks. Delay-sensitive, packet-based, applications require the reception of each media unit before its deadline in order to guarantee a synchronous playback process. In this paper, a playout algorithm tailored for real-time packet-based phone conversations transmitted over multi-hop wireless ad-hoc networks is proposed. The algorithm, denoted mobility aware playout algorithm (MAPA), adjusts the playout delay based on both node mobility, which characterizes mobile ad-hoc networks, and talk-spurt, which is an intrinsic feature of voice signals. The detection of mobility is done passively by the receiver agent using several metrics collected at the application layer. The perceptual quality is quantified by applying an augmented assessment method based on EModel specifications while including the variable nature of impairment incurred by users throughout a packetized phone conversation. Simulation results show that the proposed algorithm significantly outperforms conventional playout algorithms specifically over a mobile ad-hoc network (MANET) with a high degree of mobility. The fourth paper investigates the problem of enhancing VoIP service for ubiquitous communication in a WLAN over a university campus with spotty service area for supporting SIP-based VoIP service through either wired or wireless data networks. The campus WLAN service does not have 100% full coverage, and hence users cannot make untethered VoIP calls everywhere on campus. The goal of the paper is to overcome the limitations of such ‘‘dead spots” and motivate the use of campus IP telephony service. Two approaches are considered, called multi-hop relay and dual-mode communication. The first approach uses multi-hop relay to extend the WLAN coverage, while the second approach leverages the availability of dualmode handsets for ubiquitous voice communication. The performance evaluation of these approaches reveals that, while the two approaches can effectively allow voice communication in WLAN dead spots, they have one common problem, that is, the potential lack of support for voice call continuity, which can cause degradation of the speech quality to an active call. A cross-layer solution based on signal processing algorithms is proposed and tested to address the problem, thus achieving seamless voice call continuity while enabling ubiquitous voice communication on campus. The final testbed evaluation shows promising results for future research along the proposed direction. The fifth paper focuses on a special VoIP dimensioning issue. It is known that, current voice codecs like G.729, G.723.1 or AMR can generate traffic during voice inactivity periods for comfort noise generation (CGN). This feature

alters the classical on-off traffic pattern typically used to model the traffic generated by codecs with a silence suppression scheme. Therefore the frames generated to accomplish the CNG feature lead to severe inaccuracies in the dimensioning analysis done through traditional models based on multiplexing on-off sources like Markov-modulated poisson process (MMPP) or fluid model. In this paper the authors extend the traditional MMPP and fluid analytical models to include those sources which perform the CNG feature (Generalized VoIP sources). Then, a simple but efficient algorithm is proposed to find out the bandwidth reservation required to guarantee delay and loss in a packet-switch multiplexer node for VoIP traffic. Results are validated by simulations fed by VoIP traces and demonstrate a significant improvement in accuracy with respect to current on-off based modelling approaches. The sixth paper covers secure end-to-end information exchange over the Internet. The voice interactive personalized security (VIPSec) protocol is discussed, which is a protocol for media path key exchange to securely establish a session symmetric key for ensuring end-to-end secure communication, where it is possible to have biometric based authentication, exploiting the nature of the application. The method is appropriate for ensuring the security of an encrypted phone conversation, guaranteeing the integrity of the session key exchanged in the beginning of the conversation. VIPSec does not depend on a signalling protocol; it uses the same media path as the communication data. It requires minimal resources from the user handsets and no additional support from the network, so it is inherently scalable and readily deployable as it only needs an appropriately enhanced, secure handset. It does not rely on any public key infrastructure (PKI) infrastructure and it does not require permanent keys, enabling its use from any device, not necessarily owned by the communicating user.

Fotini-Niovi Pavlidou received the Diploma degree in Mechanical/Electrical Engineering and the Ph.D. degree from the Aristotle University of Thessaloniki (AUTh), Greece, in 1979 and 1988, respectively. She is now a full professor at the department of electrical and computer engineering (ECE) in AUTh where she is engaged in teaching and research in mobile communications, telecommunications networks and satellite communications. She is the author of more than 150 papers in international journals and conferences and serves as Division II editor for the JCN Journal and as an editor for IEEE transactions on wireless communications. She is permanently included in the technical program committee of many IEEE conferences, she has been the organizer and technical program chair of a number of events and has served as external Advisor for the promotion of many academics worldwide. She is involved in many European and National Projects, she was the delegate of Greece in the European COST program on telecommunications (1998–2004), served as Chairperson for the COST262 action ‘‘spread spectrum systems and techniques for wired and wireless systems” and she currently serves as the national delegate of Greece in the FP7 cooperation program in the ICT theme and the ARTEMIS JU. She is a Senior Member of IEEE (Communications and Vehicular Technology Society), currently chairing the Joint VTS & AES Greece Chapter in Greece.

Guest Editorial / Computer Networks 52 (2008) 2447–2449 Andreas Pitsillides is a Professor in the Department of Computer Science, University of Cyprus, and heads the Networks Research Laboratory (NetRL, http://www.NetRL.cs.ucy.ac.cy). Prof. Pitsillides is also a Founding member and Chairman and Scientific Director, of the Cyprus Academic and Research Network (CYNET) since its establishment in 2000. His research interests include fixed and wireless networks, flow and congestion control, resource allocation and radio resource management, and Internet technologies applications in mobile e-services. He has a particular interest in adapting tools from various fields of applied mathematics such as adaptive non-linear control theory, computational intelligence, and recently nature inspired techniques, to solve problems in computer networks. He has published over 190 research papers and book chapters, he is the co-editor of the book on modelling and control of complex systems (CRC, 2007), presented invited lectures at major research organisations, has given short courses at international conferences and short courses to industry. He serves/served on the executive committees of major conferences, as, e.g., INFOCOM 2002, 2003, 2004, WiOpt 2007, ISYC 2006, MCCS 2005, and ICT 1998. He is a member of the International Federation of Automatic Control (IFAC) technical committee TC 1.5 on networked systems and TC 7.3 on transportation systems, and of the international federation of information processing (IFIP) working group WG 6.3: Performance of communications systems. He is also a member of the editorial board of computer networks (COMNET) Journal. Recent European Commission funded research projects he is involved with include GINSEG, C-CAST, C-MOBILE, MOTIVE, BBONE, E-NEXT, SEACORN, M-POWER, GN2 and GEANT.

Henning Schulzrinne received his undergraduate degree in economics and electrical engineering from the Technische Hochschule in Darmstadt, Germany, in 1984, his MSEE degree as a Fulbright scholar from the University of Cincinnati, Ohio and his Ph.D. degree from the University of Massachusetts in Amherst, Massachusetts in 1987 and 1992, respectively. From 1992 to 1994, he was a member of technical staff at AT& T Bell Laboratories, Murray Hill. From 1994 to 1996, he was associate department head at GMD-Fokus (Berlin), before joining the Computer Science and Electrical Engineering Departments at Columbia University, New

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York. His research interests encompass real-time, multimedia network services in the Internet and modelling and performance evaluation. He is an editor of the Journal of communications and networks, the IEEE transactions on image processing and IEEE communications society editor of the IEEE internet computing magazine. He co-chairs the IEEE communications society internet technical committee and is chair of the IEEE communications society technical committee on computer communications. He was also technical co-chair of Infocom 2000. He is currently serving as a member of the Internet Architecture Board (IAB). Protocols co-developed by him are now Internet standards, used by almost all Internet telephony and multimedia applications. He is a fellow of the IEEE.

Dorgham Sisalem works as the Director for Strategic Architectures at Tekelec. Previously he was leading the VoIP, IMS, and security activities at the Fraunhofer Institute Fokus, Berlin, Germany. He has been in the VoIP arena since 1995, and has been involved in various research and development activities in the areas of QoS, security, and multimedia communication. He holds a Ph.D. in Engineering from the Technical University of Berlin, and has more than 40 papers in Conferences and Journals in the area of VoIP and QoS.

Fotini-Niovi Pavlidou Andreas Pitsillides Henning Schulzrinne Dorghan Sisalem Aristotle University of Thessaloniki, Department of Electrical and Computer Engineering, GR-54 124 Thessaloniki, Greece Tel.: +30 2310 996285; fax: +30 2310 996312 E-mail address: [email protected] (Fotini-Niovi Pavlidou) Available online 22 April 2008