An Enhanced SIP Proxy Server for Wireless VoIP in Wireless Mesh Networks

ADVANCES IN WIRELESS VOIP An Enhanced SIP Proxy Server for Wireless VoIP in Wireless Mesh Networks Bo Rong, International Institute of Telecommunications Yi Qian, National Institute of Standards and Technology Hsiao-Hwa Chen, National Sun Yat-Sen University ABSTRACT The wireless mesh (WMN) has emerged recently as a promising technology for next-generation wireless ing. In WMNs, it is important to provide high quality multimedia service in a flexible and intelligent manner. To address this issue in this article, we study the Session Initiation Protocol (SIP) for wireless voice over IP (VoIP) applications. Especially, we investigate the technical challenges in WMN VoIP systems and propose a design of an enhanced SIP proxy server to overcome them. An analysis of the signaling process and a study of simulation results have shown the advantages of our proposed approach. INTRODUCTION In the last few years, the wireless mesh (WMN) has drawn significant attention in the research community as a fast, easy, and inexpensive solution for broadband wireless access [1 3]. However, there are still many challenges in the design of WMNs. One of the most important issues is how to efficiently support a wireless voice over IP (VoIP) application, which is expected to be one of the killer applications for future wireless s. Many researchers advocate Session Initiation Protocol (SIP) as a feasible signaling solution for VoIP applications, because it is simpler and more efficient than H.323 [4, 5]. For example, SIP has been selected as the call control protocol for third generation (3G) IP-based mobile s [6]. In this article, we address how to deploy SIP in WMNs to support quality of service (QoS), guaranteed multimedia communication. Particularly, we assume that a WMN is connected through a gateway to the IP core, which employs multi-protocol label switching (MPLS) technology [7]. To deploy SIP in such a architecture, we must deal with many new technical challenges that have never been faced in wired s. These new challenges are raised by the inherent combination of wireless infrastructure, user mobility, and heterogeneous computing [1 3]. In this article, we mainly address the following three challenging issues. First, the interaction between a WMN and the IP core can increase the signaling complexity and cause long, call set up delays. Second, the requirements of access bandwidth change from time to time in WMNs due to the mobility of users and the variation of wireless channel conditions. Therefore, it is necessary to design a dynamic access bandwidth prediction and reservation scheme. Third, a call admission control (CAC) mechanism should be implemented in case there is a distinction between the actual access bandwidth requirements and the predicted/reserved access bandwidth condition. To overcome these challenges, we further propose to build an enhanced SIP proxy server, which we describe in this article. The enhanced SIP proxy server employs common open policy service (COPS) to dynamically reserve the access bandwidth in the IP core for all SIP s in a WMN. Moreover, the enhanced SIP proxy server contains two special modules to deal with traffic prediction and call admission control problems. The rest of this article is organized as follows. We first introduce the background of WMNs and SIP-based VoIP. We then discuss the challenges of deploying SIP in a WMN and develop an enhanced SIP proxy server to deal with these challenges. Finally, we evaluate the performance of our proposed approach and conclude the article. SIP-BASED VOIP IN WMNS WIRELESS MESH NETWORKS As shown in Fig. 1, a WMN consists of two types of nodes: mesh routers and mesh clients. The mesh routers form the infrastructure of a mesh backbone for mesh clients. In general, mesh routers have minimal mobility and operate just like a of fixed routers, except that they are connected by wireless links through wireless technologies such as IEEE 82.11. We can see 18 163-684/8/$25. 28 IEEE IEEE Communications Magazine January 28

from Fig. 1 that the WMN can access the Internet through a gateway mesh router that is connected to the IP core with physical wires. In this study, we assume that the IP core employs MPLS technology. MPLS operates at an open systems interconnection (OSI) model layer that lies between traditional definitions of layer 2 (data link layer) and layer 3 ( layer). It was designed to provide a unified data-carrying service for both circuit-based clients and packet-switching clients. Many researchers recommended MPLS as a reliable way to provide QoS guaranteed services [7]. In WMNs, every mesh router is equipped with a traffic aggregation device (similar to an 82.11 access point) that interacts with individual mesh clients. The mesh router relays the aggregated data traffic of mesh clients to and from the IP core. Typically, a mesh router has multiple wireless interfaces to communicate with other mesh routers, and each wireless interface works, corresponding to one wireless channel. These channels have different characteristics due to the inherent features of a wireless environment. In practice, wireless interfaces are usually running on different frequencies and built on either the same or different wireless access technologies, such as IEEE 82.11a/b/g/n. It is also possible that directional antennas are employed on some interfaces to establish wireless channels over long distance. SIP-BASED VOIP SIP is defined in RFC 2543 as an Internet Engineering Task Force (IETF) standard for multimedia conferencing over IP. It is an ASCII-based, application-layer control protocol that can be used to establish, maintain, and terminate calls between two or more end points. Just like other VoIP protocols, SIP is designed to provide the functions of signaling and session management within a packet telephony. Signaling enables call information to be carried across boundaries. Session management provides the ability to control the attributes of an end-to-end call. Compared to H.323, SIP is a much more streamlined protocol, developed specifically for IP telephony [5]. SIP is simpler and more efficient than H.323, and it takes advantage of existing protocols to handle certain parts of the process. For example, media gateway control protocol (MGCP) is used by SIP to establish a gateway connecting to the public-switched telephone (PSTN) system. Figure 2 demonstrates the deployment of SIPbased VoIP in WMNs. Here, the SIP proxy server is an intermediate device that receives SIP requests from a client and then forwards the requests on behalf of the client. Basically, proxy servers receive SIP messages and forward them to the next SIP server in the. Proxy servers can provide functions such as routing, reliable request retransmission, authentication, authorization, and security. Moreover, each WMN is connected to the MPLS-based IP core through a label edge router () that operates at the edge of an MPLS and uses routing information to assign labels to datagrams and then forwards them into the MPLS domain. r r 5 r 1 r 4 r 2 r 7 r 3 Figure 1. An example of wireless mesh. SIP backbone r 6 r8 r 1 r 11 MPLS-based IP core SIP proxy server r 12 r 14 r 16 TECHNICAL CHALLENGES OF DEVELOPING SIP-BASED VOIP IN WMNS r 15 r 17 In this article, we address the issues of how to use SIP to support the wireless VoIP in WMNaccessed IP s. According to the original design, SIP sets up and tears down only sessions with a minimal focus on the management of active sessions. To deploy SIP in WMNs, we must face many new challenging issues that are caused by the instability of the wireless environment and by user mobility. In this article, we mainly study three technical challenges in WMN SIP deployment, that is, call set up delay, access bandwidth prediction/reservation, and call admission control. CALL SET UP DELAY In the real world, a WMN usually serves as an access to the Internet. To provide guaranteed QoS, IP core s are built with wired technologies, such as MPLS. Moreover, most VoIP applications intend to go out of their own local WMNs for counterparts in the Internet. Therefore, when SIP is used to set up a VoIP session, it must face a heterogeneous environment. This heterogeneous environment increases the complexity of the signaling process and causes a long call set up delay. G i r 9 MPLS-based IP core Figure 2. The deployment of SIP based VoIP in WMNs. r 13 r i Coverage area of mesh router r i G i : router with gateway r i : router clients SIP proxy server SIP IEEE Communications Magazine January 28 19

SIP messages COPS messages SIP and COPS protocol stacks Figure 3. The framework of the enhanced SIP proxy server. Call admission control Access bandwidth prediction Access bandwidth negotiation Without losing the capability of generalizing, this article studies the scenario where a WMN is connected to the MPLS-based IP core. We assume that the MPLS runs with traffic engineering capability, which is an essential step to achieve high efficiency. In traffic engineering-enabled MPLS s, Constraint-based Routing Label Distribution Protocol (CR-LDP) or Resource Reservation Protocol with Traffic Engineering Extensions (RSVP-TE) is employed to set up a label-switched path (LSP) dynamically for a connection with QoS requirements. As a result, the total session set up delay of a VoIP call should be the sum of SIP signaling and MPLS signaling times if one SIP client in a WMN wants to communicate with its counterpart in another WMN through MPLSbased IP core. ACCESS BANDWIDTH PREDICTION AND RESERVATION When designing a SIP architecture for WMNs, we must note two facts: The users in WMNs are free to move to anywhere at anytime. The wireless channel conditions may vary from time to time. Clearly, these two facts can result in varying access bandwidth requirements in WMNs. To accommodate this variation, the best way is to let WMN gateway mesh routers dynamically reserve access bandwidth from the IP core, because the fixed bandwidth reservation approach is not efficient in this scenario. For example, there can be two straightforward ways to reserve the fixed access bandwidth for variable requirements. The first way is called the optimal user satisfaction scheme, which reserves the maximum bandwidth that a WMN ever requires. The second way is called the optimal cost scheme, which reserves the minimum bandwidth that a WMN ever requires. Nevertheless, both of these methods have their shortcomings. The optimal user satisfaction scheme is not economic, although it can always provide enough access bandwidth for WMN users. The optimal cost scheme may not ensure that all users are satisfied, although it is able to reduce the expense of WMN operators. Dynamic access bandwidth reservation requires the prediction of outgoing traffic load in WMNs. Because there always exists a distinction between the exact access bandwidth requirement and the predicted access bandwidth requirement, the call admission control mechanism must be implemented. CALL ADMISSION CONTROL A CAC mechanism must be employed when the predicted and reserved access bandwidth is different from the real one. CAC is used to accept or reject connection requests based on the QoS requirements of these connections and the system state information. CAC prevents oversubscription of VoIP s and is a concept that applies only to real-time media traffic but not to data traffic. A CAC mechanism complements the capabilities of QoS tools to protect audio/video traffic from the negative effects of other audio/video traffic and to keep excessive audio/video traffic away from the. CAC can also help wireless mesh s to provide different types of traffic load with different priorities by manipulating their blocking probabilities. AN ENHANCED SIP PROXY SERVER FOR WIRELESS VOIP Conventionally, a proxy server is an optional SIP component that handles routing of SIP signaling but does not initiate SIP messages. Proxy servers also can provide some auxiliary functions such as authentication, authorization, reliable request retransmission, and security. To overcome the technical challenges in wireless VoIP deployment, we develop an enhanced SIP proxy server with the framework shown in Fig. 3. Particularly, the enhanced SIP proxy server utilizes COPS messages to negotiate with the MPLS about the overall access bandwidth requirement on behalf of all SIP s in a WMN (not only one SIP ). Then, the exchanges traffic engineering signaling with other routers inside the MPLS core to set up the corresponding LSPs. In this way, the LSPs required by SIP telephony are set up in the MPLS core before SIP calls are made. As a result, the SIP call set up delay in the MPLS is decreased significantly. We use a set of time marks {t, t 1, t 2,, t n 1, t n, t n+1, } to distinguish the time instances of the system. If the enhanced SIP proxy server knows that the overall access bandwidth requirement of its WMN during [t n 1, t n ] is exactly B n 1,n, then at time t n 1, the enhanced SIP proxy server negotiates with the MPLS to obtain B n 1,n outgoing bandwidth by using COPS messages. As time goes by, if the enhanced SIP proxy server knows that the overall bandwidth requirement of the WMN during [t n, t n+1 ] changes to B n,n+1, then at time t n, the enhanced SIP proxy server should renegotiate with the MPLS to increase/decrease the bandwidth requirement to B n,n+1. 11 IEEE Communications Magazine January 28

SIP If declined Decline Decline Enhanced SIP proxy server Local client If accepted Decline MPLS-based IP core If accepted If declined decline Enhanced SIP proxy server Local client SIP A CAC mechanism complements the capabilities of QoS tools to protect audio/video traffic from the negative effects of other audio/video traffic and to keep excessive audio/video traffic away from the. 18 ringing 18 ringing 18 ringing 2 OK 2 ACK 2 ACK ACK ACK ACK Traffic stream If bandwidth negotiation is needed Figure 4. The signaling flow with the enhanced SIP proxy server. However, it is impossible for the enhanced SIP proxy server to know the exact value of B n 1,n before the time instance of t n 1. Usually, the enhanced SIP proxy server can employ only a certain bandwidth prediction algorithm to give an approximate value of B n 1,n, which can be defined as B^n 1,n. If B^n 1,n < B n 1,n during [t n 1, t n ], the WMN does not have enough outgoing bandwidth to accommodate all SIP calls, and the enhanced SIP proxy server must utilize a call admission control mechanism to decline some of the call requests. In contrast, if B^n 1,n > B n 1,n } during [t n 1, t n ], some of the outgoing bandwidth resource of the WMN would be wasted. From the above discussion, we can conclude that the algorithms of access bandwidth prediction and call admission control running on an enhanced SIP proxy server are critical to our approach. In the literature, extensive studies have been conducted on access bandwidth prediction and call admission control. As a result, our proposed enhanced SIP proxy server can directly inherit these research results. For access bandwidth prediction, a multiresolution finite-impulse-response (FIR) neural--based learning algorithm was developed in [8], using the maximal overlap discrete wavelet transform (MODWT). This algorithm is a good choice for the enhanced SIP proxy server, because it has satisfactory trade-off between prediction accuracy and computational complexity. On the other hand, the design of call admission control can be formulated as an optimization problem, where the demands of both the WMN service provider and the user are taken into account. To solve this optimization problem, the authors of [9] proposed a utility- IEEE Communications Magazine January 28 111

Average SIP call setup delay (s) 5 4 3 2 1 Average SIP and MPLS signaling delay (s) 6 5 4 3 2 1 With traditional SIP proxy server With enhanced SIP proxy server.5 1 1.5 2 2.5 3 3.5 4 Time (h) Figure 5. The average call setup delay of SIP-based wireless VoIP over four hours. SIP signaling delay (traditional SIP proxy server) MPLS signaling delay (traditional SIP proxy server) SIP signaling delay (enhanced SIP proxy server) MPLS signaling delay (enhanced SIP proxy server).5 1 1.5 2 2.5 3 3.5 4 Time (h) Figure 6. The decomposition of call setup delay. constrained, greedy approximation algorithm, which can be easily implemented in the enhanced SIP proxy server. PERFORMANCE ANALYSIS SIGNALING PROCESS WITH AN ENHANCED SIP PROXY SERVER Figure 4 shows the signaling flow in the proposed SIP architecture with an enhanced SIP proxy server. The call set up starts with a standard SIP message sent from the caller to the local enhanced SIP proxy server in a WMN. This message carries the callee URL in a SIP header and the QoS requirements of a SIP call-in body Session Description Protocol (SDP). Regarding the caller ID, the QoS requirements, and the remaining outgoing bandwidth in a local WMN, the enhanced SIP proxy server decides whether this SIP call request is admitted. If the call request is admitted, an enhanced SIP proxy server will forward the original message to the callee; otherwise, it simply sends the caller a DECLINE message to drop the call. Furthermore, whether the call is admitted or not, it is registered in the enhanced SIP proxy server for the purposes of access bandwidth prediction and call admission control in the future. If the access bandwidth must change, the enhanced SIP proxy server uses COPS messages to negotiate with the MPLS-based IP core to set up new LSPs with the required bandwidth. The COPS Protocol is part of the Internet protocol suite as defined by IETF RFC 2748. COPS specifies a simple client/server model for supporting policy control over QoS signaling protocols (e.g., RSVP). As shown in Fig. 4, the enhanced SIP proxy server uses COPS request (REQ) and COPS decline (DEC) messages to make bandwidth negotiations with the MPLS core in an on-demand manner. The previous discussions clearly show that the approach of an enhanced SIP proxy server can reduce considerably the call set up delay, because the LSPs required by SIP telephony are set up in the MPLS-based IP core before SIP calls start. SIMULATION RESULTS To further demonstrate the advantages of our approach, we conducted a simulation study to compare the performance of a traditional SIP proxy server and an enhanced SIP proxy server in terms of call set up delay. We used OPNET Modeler 11. to simulate the environment as shown in Fig. 1. In the simulation, we studied the case of medium traffic load, which is incurred by real-time multimedia applications. We programmed the traditional SIP proxy server according to [4] and the enhanced SIP proxy server according to the architecture proposed in this article. For simplicity, in our simulation, the enhanced SIP proxy server employs a complete sharing (CS) CAC policy, which means that an incoming connection is accepted if sufficient bandwidth resources are available. Figure 5 demonstrates the average call set up delay of SIP-based wireless VoIP during four hours. It is seen that the SIP call set up delay varies between three and four seconds when a traditional SIP proxy server is employed. On the other hand, the delay is as low as.6 second or even less when an enhanced SIP proxy server is employed. Figure 6 illustrates the decomposition of call set up delay, which includes the SIP signaling delay and the MPLS signaling delay. As we can see, an enhanced SIP proxy server generates a much shorter call set up delay than a traditional SIP proxy server, because it has a significant decrement of MPLS signaling delay. It is noted that we also should be concerned about the performance of an enhanced SIP proxy server in terms of access bandwidth prediction and call admission control. From existing studies as previously mentioned, an enhanced SIP proxy server can directly borrow an access bandwidth prediction algorithm and call admission control algorithm. Therefore, the perfor- 112 IEEE Communications Magazine January 28

mance of access bandwidth prediction and call admission control depends on which algorithm the enhanced SIP proxy server employs. CONCLUSION In this article, we investigated the deployment of SIP-based VoIP in WMNs. We first discussed the technical challenges in a wireless VoIP system, such as call set up delay, access bandwidth prediction and reservation, call admission control, and so on. We then proposed a novel approach of an enhanced SIP proxy server to deal with these challenges. The analysis of signaling process and the study of simulation results have shown the advantages of our proposed approach. REFERENCES [1] I. F. Akyildiz and X. Wang, A Survey on Wireless Mesh Networks, IEEE Commun. Mag., vol. 43, no. 9, Sept. 25, pp. S23 S3. [2] A. Raniwala and T. Chiueh, Architecture and Algorithms for an IEEE 82.11-based Multi-Channel Wireless Mesh Network, IEEE INFOCOM 25, vol. 3, Mar. 25, pp. 2223 34. [3] H. Jiang et al., Differentiated Services for Wireless Mesh Backbone, IEEE Commun. Mag., vol. 44, no. 7, July 26, pp. 113 19. [4] J. Rosenberg et al., SIP: Session Initiation Protocol, RFC 3261 IETF, June 22. [5] U. Black, Voice Over IP, Prentice Hall, 2. [6] 3GPP, Technical Specification Group Services and System Aspects; Network Architecture (Release 5), Technical Report TS23.2, 3GPP, Mar. 22. [7] T. Li, MPLS and the Evolving Internet Architecture, IEEE Commun. Mag., vol. 37, no. 12, Dec. 1999, pp. 38 41. [8] V. Alarcon-Aquino and J. A. Barria, Multiresolution FIR Neural-Network-Based Learning Algorithm Applied to Network Traffic Prediction, IEEE Trans. Systems, Man and Cybernetics, Part C: Applications and Reviews, vol. 36, no. 2, Mar. 26, pp. 28 2. [9] B. Rong, Y. Qian, and K. Lu, Integrated Downlink Resource Management for Multiservice WiMAX Networks, IEEE Trans. Mobile Computing, vol. 6, no. 6, June 27, pp. 621 32. BIOGRAPHIES BO RONG [M 7] (bo.rong@ieee.org) received a B.S. degree from Shandong University in 1993, an M.S. degree from Beijing University of Aeronautics and Astronautics in 1997, and a Ph.D. degree from Beijing University of Posts and Telecommunications in 21. Currently, he is a researcher at the International Institute of Telecommunications, Montreal, Canada. His current research interests focus on modeling, simulation, and performance analysis for next-generation wireless s. After receiving his Ph.D., he worked as a software engineer for a start-up company in Beijing for one year, as a postdoctoral fellow in the Department of Electrical Engineering, Ecole de technologie superieure, Universite du Quebec for three years, and then as a postdoctoral fellow in the Department of Electrical and Computer Engineering, University of Puerto Rico at Mayaguez for one and a half years. YI QIAN [M 95, SM 7] (yqian@nist.gov) received a Ph.D. degree in electrical engineering with a concentration in telecommunication s from Clemson University. He is with the National Institute of Standards and Technology, in Gaithersburg, MD. His current research interests include security, design, modeling, simulations and performance analysis for next generation wireless s, wireless sensor s, broadband satellite s, optical s, high-speed s and the Internet. He has publications and patents in all these areas. He was an assistant professor in the Department of Electrical and Computer Engineering, University of Puerto Rico at Mayaguez (UPRM) between July 23 and July 27. At UPRM, he taught courses on wireless s, design, management, and performance analysis. Prior to joining UPRM in July 23, he worked for several start-up companies and consulting firms, in the areas of voice over IP, fiber optical switching, Internet packet video, optimizations, and planning as a technical advisor and a senior consultant. He also worked several years for the Wireless Systems Engineering Department, Nortel Networks in Richardson, Texas as a senior member of the scientific staff and as a technical advisor. While at Nortel, he was a project leader for various wireless and satellite product design projects, customer consulting projects, and advanced technology research projects. He was also in charge of wireless standard development and evaluations. He is the author of the book Information Assurance Dependability and Security in Networked Systems (Morgan Kaufmann, 27). He is a member of ACM. HSIAO-HWA CHEN [S 89, M 91, SM 1] (hshwchen@ieee.org) received B.Sc. and M.Sc. degrees from Zhejiang University, China, and a Ph.D. degree from the University of Oulu, Finland in 1982, 1985, and 199, respectively, all in electrical engineering. He is currently a full professor and was the founding director of the Institute of Communications Engineering at the National Sun Yat-Sen University, Taiwan. He has authored or co-authored over 2 technical papers in major international journals and conferences, five books, and several book chapters in the areas of communications, including the books, [[Next Generation Wireless Systems and Networks]] and [[The Next Generation CDMA Technologies]], both published by John Wiley and Sons in 25 and 27, respectively. He has been an active volunteer for various IEEE technical activities for over 2 years. Currently, he is serving as the chair of IEEE Communications Society Radio Communications Committee. He served or is serving as symposium chair/co-chair of many major IEEE conferences, including VTC, ICC, GLOBECOM, WCNC, and so on. He served or is serving as associate editor and/or guest editor of numerous important technical journals in communications. He is serving as the chief editor (Asia and Pacific) for Wiley s Wireless Communications and Mobile Computing (WCMC) Journal and Wiley s International Journal of Communication Systems. He is the Editor-in-Chief of Wiley Security and Communication Networks journal (www.interscience.wiley.com/journal/security). He is also an adjunct professor at Zhejiang University, China and Shanghai Jiao Tung University, China. To further demonstrate the advantages of our approach, we conducted a simulation study to compare the performance of a traditional SIP proxy server and an enhanced SIP proxy server in terms of call set up delay. IEEE Communications Magazine January 28 113