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

I

NTRODUCTION

In the last few years, the wireless mesh network (WMN) has drawn significant attention in the research community as a fast, easy, and inexpen-sive 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 networks.

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 proto-col for third generation (3G) IP-based mobile networks [6]. In this article, we address how to deploy SIP in WMNs to support quality of ser-vice (QoS), guaranteed multimedia communica-tion. Particularly, we assume that a WMN is connected through a gateway to the IP core net-work, which employs multi-protocol label switch-ing (MPLS) technology [7].

To deploy SIP in such a network architecture, we must deal with many new technical chal-lenges that have never been faced in wired net-works. These new challenges are raised by the

inherent combination of wireless infrastructure, user mobility, and heterogeneous network com-puting [1–3]. In this article, we mainly address the following three challenging issues. First, the interaction between a WMN and the IP core network 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 con-ditions. Therefore, it is necessary to design a dynamic access bandwidth prediction and reser-vation 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 predict-ed/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 network for all SIP ter-minals 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 orga-nized as follows. We first introduce the back-ground of WMNs and SIP-based VoIP. We then discuss the challenges of deploying SIP in a WMN and develop an enhanced SIP proxy serv-er to deal with these challenges. Finally, we eval-uate the performance of our proposed approach and conclude the article.

SIP-

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WMN

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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 network of fixed routers, except that they are connected by wireless links through wireless technologies such as IEEE 802.11. We can see

A

BSTRACT

The wireless mesh network (WMN) has emerged recently as a promising technology for next-generation wireless networking. In WMNs, it is important to provide high quality multime-dia 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.

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DVANCES IN

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Bo Rong, International Institute of Telecommunications

Yi Qian, National Institute of Standards and Technology

Hsiao-Hwa Chen, National Sun Yat-Sen University

An Enhanced SIP Proxy Server for

Wireless VoIP in

from Fig. 1 that the WMN can access the Inter-net through a gateway mesh router that is con-nected to the IP core network with physical wires. In this study, we assume that the IP core network employs MPLS technology. MPLS oper-ates at an open systems interconnection (OSI) model layer that lies between traditional defini-tions of layer 2 (data link layer) and layer 3 (net-work 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 802.11 access point) that interacts with individu-al mesh clients. The mesh router relays the aggregated data traffic of mesh clients to and from the IP core network. Typically, a mesh router has multiple wireless interfaces to com-municate 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 inter-faces are usually running on different frequen-cies and built on either the same or different wireless access technologies, such as IEEE 802.11a/b/g/n. It is also possible that directional antennas are employed on some interfaces to establish wireless channels over long distance.

SIP-B

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IP

SIP is defined in RFC 2543 as an Internet Engi-neering Task Force (IETF) standard for multi-media conferencing over IP. It is an ASCII-based, application-layer control protocol that can be used to establish, maintain, and ter-minate 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 network. Signaling enables call information to be carried across network 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 effi-cient than H.323, and it takes advantage of exist-ing 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 tele-phone network (PSTN) system.

Figure 2 demonstrates the deployment of SIP-based 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 network. Proxy servers can provide functions such as routing, reliable request retransmission, authentication, authoriza-tion, and security. Moreover, each WMN is con-nected to the MPLS-based IP core network through a label edge router (LER) that operates at the edge of an MPLS network and uses routing information to assign labels to datagrams and then forwards them into the MPLS domain.

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In this article, we address the issues of how to use SIP to support the wireless VoIP in WMN-accessed IP networks. 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 environ-ment 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 admis-sion control.

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In the real world, a WMN usually serves as an access network to the Internet. To provide guar-anteed QoS, IP core networks 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 Inter-net. Therefore, when SIP is used to set up a VoIP session, it must face a heterogeneous net-work environment. This heterogeneous netnet-work environment increases the complexity of the sig-naling process and causes a long call set up delay.

■Figure 1. An example of wireless mesh network.

G_i: Wireless mesh router with gateway r_i: Wireless mesh router Wireless mesh clients Coverage area of mesh router r_i Wireless mesh backbone

MPLS-based IP core network

r₀ r₅ r₁₀ r₁₁ r_i r₁₃ r₁₂ r₁₄ r₁₅ r₁₆ r₁₇ r₉ r₆ r₄ r₁ r₂ r₇ r₃ r₈ Gi

■Figure 2. The deployment of SIP based VoIP in WMNs.

Wireless mesh network

MPLS-based IP core network SIP proxy server

SIP terminal

LER

Wireless mesh network SIP proxy server

SIP terminal LER

Without losing the capability of generalizing, this article studies the scenario where a WMN is connected to the MPLS-based IP core network. We assume that the MPLS network runs with traffic engineering capability, which is an essen-tial step to achieve high efficiency. In traffic engineering-enabled MPLS networks, Con-straint-based Routing Label Distribution Proto-col (CR-LDP) or Resource Reservation ProtoProto-col 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 MPLS-based IP core network.

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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 net-work, 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 vari-able 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 band-width that a WMN ever requires. Nevertheless, both of these methods have their shortcomings. The optimal user satisfaction scheme is not eco-nomic, 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 distinc-tion between the exact access bandwidth require-ment and the predicted access bandwidth requirement, the call admission control mecha-nism must be implemented.

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A CAC mechanism must be employed when the predicted and reserved access bandwidth is dif-ferent from the real one. CAC is used to accept or reject connection requests based on the QoS requirements of these connections and the sys-tem state information. CAC prevents oversub-scription of VoIP networks and is a concept that applies only to real-time media traffic but not to data traffic.

A CAC mechanism complements the capabil-ities 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 network. CAC can also help wire-less mesh networks to provide different types of traffic load with different priorities by manipu-lating their blocking probabilities.

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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 deploy-ment, 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 LER about the overall access bandwidth requirement on behalf of all SIP terminals in a WMN (not only one SIP terminal). Then, the LER exchanges traffic engineering signaling with other routers inside the MPLS core network to set up the corresponding LSPs. In this way, the LSPs required by SIP telephony are set up in the MPLS core network before SIP calls are made. As a result, the SIP call set up delay in the MPLS network is decreased significantly.

We use a set of time marks {t0, t1, t2, …, tn–1,

tn, tn+1, …} to distinguish the time instances of

the system. If the enhanced SIP proxy server knows that the overall access bandwidth require-ment of its WMN during [tn–1, tn] is exactly

Bn–1,n, then at time tn–1, the enhanced SIP proxy

server negotiates with the MPLS network to obtain Bn–1,noutgoing 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 [tn, tn+1]

changes to Bn,n+1, then at time tn, the enhanced

SIP proxy server should renegotiate with the MPLS network to increase/decrease the band-width requirement to Bn,n+1.

■Figure 3. The framework of the enhanced SIP proxy server.

SIP and COPS protocol stacks Call admission control Access bandwidth prediction SIP messages COPS messages Access bandwidth negotiation

However, it is impossible for the enhanced SIP proxy server to know the exact value of

Bn–1,nbefore the time instance of tn–1. Usually,

the enhanced SIP proxy server can employ only a certain bandwidth prediction algorithm to give an approximate value of Bn–1,n, which can be

defined as B^

n–1,n. If B^n–1,n< Bn–1,nduring [tn–1,

tn], 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> Bn–1,n}

during [tn–1, tn], 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 con-ducted 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 pre-diction, a multiresolution finite-impulse-response (FIR) neural-network-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 opti-mization 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-■Figure 4. The signaling flow with the enhanced SIP proxy server.

MPLS-based IP core network Cops REQ Cops DEC INVITE If declined Decline Cops REQ Cops DEC Cops REQ Cops DEC Cops REQ Cops DEC If accepted INVITE If accepted INVITE If declined decline INVITE INVITE ACK ACK Decline Decline 180 ringing 200 ACK 180 ringing 200 ACK ACK Traffic stream

If bandwidth negotiation is needed 180 ringing 200 OK LER SIP terminal Local client network LER SIP terminal Enhanced SIP proxy server Enhanced SIP proxy server Local client network 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 network.

constrained, greedy approximation algorithm, which can be easily implemented in the enhanced SIP proxy server.

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Figure 4 shows the signaling flow in the pro-posed SIP architecture with an enhanced SIP proxy server. The call set up starts with a stan-dard SIP INVITE 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 require-ments, 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 INVITE 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 pre-diction 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 net-work to set up new LSPs with the required band-width. 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 network 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 network before SIP calls start.

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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 11.0 to simulate the network environ-ment 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 0.6 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 pre-diction 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 admis-sion control algorithm. Therefore, the

perfor-■_{Figure 6. The decomposition of call setup delay.}

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)

Time (h) 0

0

Average SIP and MPLS signaling

delay (s) 1 2 4 3 5 6 0.5 1 1.5 2 2.5 3 3.5 4

■Figure 5. The average call setup delay of SIP-based wireless VoIP over four

hours.

Time (h) 0

0

Average SIP call setup delay (s) 1

2 3 4 5

0.5 1 1.5 2 2.5 3 3.5 4 With traditional SIP proxy server

mance of access bandwidth prediction and call admission control depends on which algorithm the enhanced SIP proxy server employs.

C

ONCLUSION

In this article, we investigated the deployment of SIP-based VoIP in WMNs. We first discussed the technical challenges in a wireless VoIP sys-tem, such as call set up delay, access bandwidth prediction and reservation, call admission con-trol, and so on. We then proposed a novel approach of an enhanced SIP proxy server to deal with these challenges. The analysis of sig-naling process and the study of simulation results have shown the advantages of our proposed approach.

R

[1] I. F. Akyildiz and X. Wang, “A Survey on Wireless Mesh Networks,” IEEE Commun. Mag., vol. 43, no. 9, Sept. 2005, pp. S23–S30.

[2] A. Raniwala and T. Chiueh, “Architecture and Algo-rithms for an IEEE 802.11-based Multi-Channel Wireless Mesh Network,” IEEE INFOCOM 2005, vol. 3, Mar. 2005, pp. 2223–34.

[3] H. Jiang et al., “Differentiated Services for Wireless Mesh Backbone,” IEEE Commun. Mag., vol. 44, no. 7, July 2006, pp. 113–19.

[4] J. Rosenberg et al., “SIP: Session Initiation Protocol,” RFC 3261 IETF, June 2002.

[5] U. Black, Voice Over IP, Prentice Hall, 2000.

[6] 3GPP, “Technical Specification Group Services and Sys-tem Aspects; Network Architecture (Release 5),” Techni-cal Report TS23.002, 3GPP, Mar. 2002.

[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. 2006, pp. 208–20.

[9] B. Rong, Y. Qian, and K. Lu, “Integrated Downlink Resource Management for Multiservice WiMAX Net-works,” IEEE Trans. Mobile Computing, vol. 6, no. 6, June 2007, pp. 621–32.

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BORONG[M’07] ([email protected]) 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 2001. Currently, he is a researcher at the International Institute of Telecommunications, Mon-treal, Canada. His current research interests focus on mod-eling, simulation, and performance analysis for next-generation wireless networks. 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 tech-nologie 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.

YIQIAN[M’95, SM’07] ([email protected]) received a Ph.D. degree in electrical engineering with a concentration in telecommunication networks from Clemson University. He is with the National Institute of Standards and Technology, in Gaithersburg, MD. His current research interests include network security, network design, network modeling, simu-lations and performance analysis for next generation wire-less networks, wirewire-less sensor networks, broadband satellite networks, optical networks, high-speed networks and the Internet. He has publications and patents in all these areas. He was an assistant professor in the Depart-ment of Electrical and Computer Engineering, University of Puerto Rico at Mayaguez (UPRM) between July 2003 and July 2007. At UPRM, he taught courses on wireless works, network design, network management, and net-work performance analysis. Prior to joining UPRM in July 2003, he worked for several start-up companies and con-sulting firms, in the areas of voice over IP, fiber optical switching, Internet packet video, network optimizations, and network 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 pro-ject leader for various wireless and satellite network prod-uct 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, 2007). He is a member of ACM.

HSIAO-HWACHEN[S’89, M’91, SM’01] ([email protected]) received B.Sc. and M.Sc. degrees from Zhejiang University, China, and a Ph.D. degree from the University of Oulu, Fin-land in 1982, 1985, and 1990, respectively, all in electrical engineering. He is currently a full professor and was the founding director of the Institute of Communications Engi-neering at the National Sun Yat-Sen University, Taiwan. He has authored or co-authored over 200 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 Tech-nologies]], both published by John Wiley and Sons in 2005 and 2007, respectively. He has been an active volunteer for various IEEE technical activities for over 20 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 confer-ences, including VTC, ICC, GLOBECOM, WCNC, and so on. He served or is serving as associate editor and/or guest edi-tor of numerous important technical journals in communi-cations. He is serving as the chief editor (Asia and Pacific) for Wiley’s Wireless Communications and Mobile Comput-ing (WCMC) Journal and Wiley’s International Journal of Communication Systems. He is the Editor-in-Chief of Wiley Security and Communication Networks journal (www.inter-science.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