An Application-quality-based Mobility Management Scheme - CiteSeerX

An Application-quality-based Mobility Management Scheme Liam Murphy, Member, IEEE, James Noonan, Student Member, IEEE, Philip Perry, Member, IEEE, and John Murphy, Senior Member, IEEE 1 1 Abstract— Experimental results are presented for end-point controlled handover of a stream of voice-like packet data traffic between two independent wireless networks. The handover was achieved by using a version of the Stream Control Transmission Protocol (SCTP) that had been modified to monitor link quality and switch the primary connection to the “best” quality link available, where quality is evaluated in terms of application performance. The results clearly show that mobility management based solely on received signal strength is not suitable for wireless systems with contention-based access policies such as WiFi. Index Terms— Mobile Communication, Network Testing, Packet Radio, Wide area networks, Wireless LAN, Handover, VoIP, Call Quality

exemplified by the Unlicensed Mobile Access (UMA) system, which ties WiFi hotspots into a 3G core network [1]. This paper reports the first known end-point controlled handover of a data transfer session from a WiMAX system to a WiFi system, and vice versa. The data stream in question was a packet stream emulating a VoIP call being sent from a mobile node via a WiFi access link. As the WiFi link quality degraded, the call was seamlessly transferred to a WiMAX link in the sense that no packets were lost and there was no interruption to the packet stream. This represents a significant practical demonstration of mobility management in the end points, rather than relying on network-based solutions. The results also show that received signal strength is a poor indicator of expected call quality in a congested WiFi environment.

I. INTRODUCTION THE use of Voice over IP (VoIP) technology is moving from the low cost international call services market to become a serious contender in the home phone market, and is expected to continue its growth into the wireless market. There is a move towards a multitude of small inexpensive WiFi (Wireless LAN) “cells” overlaid by larger WiMAX or WiBRO cells offering competing and complementary offerings to the range of cellular systems that are available. This is leading towards a Heterogeneous Wireless Network Environment where an end user could be offered an instantaneous choice of network technologies that could provide connectivity and billing rate alternatives. These networks are expected to be owned by many different entities who may compete to provide each customer’s network connectivity. Mobility and handover have been addressed to date by using network-based entities to redirect packet streams, because any interruption to service on a voice call can have significant impact on the call quality. This approach is

1 L. Murphy, P. Perry and J. Murphy are with the Performance Engineering Laboratory, School of Computer Science and Informatics, University College Dublin, Ireland. J. Noonan is now with Avaya, although his work on this paper was performed while with the Performance Engineering Laboratory at UCD. The authors are grateful to Eircom for an equipment loan which helped make this work possible. This work was supported in part by the Informatics Research Initiative of Enterprise Ireland.

Our experiments were carried out using a private IEEE802.11b WiFi Access Point (AP) and a commercially available pre-WiMAX deployment. We developed an application that used a modified version of SCTP to trigger a handover between the two networks with no network intervention. To avoid firewall issues, the client and the server were both connected through the WiMAX network as shown in Figure 1. The rest of this paper is organized as follows: some related work is described in section II; causes of link quality degradation in wireless networks are discussed in section III. In the following two sections we present our experimental setup and results, respectively. Finally we offer some conclusions and suggestions for future work in section VI. II. RELATED WORK Handover in cellular networks is well understood: a large geographic area is divided into smaller cells, and handover is performed between cells as required [2]. This choice of network and decision to handover is normally based on the strongest available Receive Signal Level (RSL) of the available cells. The handover is normally performed with the cooperation of both the end user and the network. This approach is valid under the assumption that the network cells provide a performance that only differs based on signal quality, measured using RSL and Bit Error Rate (BER).

It is possible to use RSL as the chief metric for network decision in pure end-point driven decisions too. This has been implemented to perform handover between WiFi cells; and provides the advantage that additional network infrastructure is not required [3] [4]. However, so far this work has relied on the assumption that the network technology remains the same. There has been some work which has examined heterogeneous networks, of which the BARWAN project [5] is an example. This overlaid a number of heterogeneous networks in concentric circles. However, there was an implicit assumption that the network performance of the smaller cells would always be better than the bigger cells, and so the network decision could be made in advance, and did not require continuous re-evaluation. Our work considers situations where multiple factors contribute to the performance of a radio network, meaning that the decision is dependent on different factors and must be made dynamically. An important advantage of the end-point approach is that network decisions can be based not only on the current

moves away from the AP that it is associated with. Their radio channel will degrade with increasing distance as the RSL reduces and their Carrier to Interference Ratio (CIR) increases. Due to the high bit rates that can be used in WiFi, the multipath reflections that are usually associated with typical indoor WiFi deployment scenarios will cause significant delay spread and associated reduction in the coherence bandwidth of the channel. Despite the use of channel equalization, as the user moves away, the multi-path will eventually result in a rise in Inter-Symbol Interference (ISI). Both the reduced CIR and increased ISI will manifest themselves as increased bit errors and frame errors, resulting in frame retransmissions. These retransmissions are quickly scheduled, so that the IP packet delay will not be directly impacted significantly. However, the retransmissions mean that the WiFi station is taking the medium for a longer time that it was previously. In response to increasing frame errors, the WiFi MAC layer uses link adaptation, reducing the bit rate of their transmission which again results in an increase in the length of time taken to transmit a given IP packet. So, as a user moves away from the AP, they will seize the medium for a longer time to transmit each packet. For a fixed packet size and packet generation rate – as is typical of VoIP traffic – this means that there is an increase in the offered load to the network.

Figure 1 - Experimental Set up network conditions but also on application requirements (cellular network decisions are normally optimised for a single application, namely voice). Although only a single application is considered here, the work could easily be extended to consider different applications, perhaps with the use of a utility function. This idea has been explored in [6], [7].

III. LINK QUALITY DEGRADATION For the purposes of this discussion, the WiFi link quality is defined as being the quality of the link provided by the link layer to the IP layer: it is not the quality of the radio link. It is also assumed that the VoIP client machine is within the coverage area of the WiFi AP and a WiMAX base station. Initially the client uses that AP for sending the VoIP data stream. It is also registered on the WiMAX network and periodically probes a connection to the VoIP peer through that connection. This results in a small bandwidth overhead on the order of 5%, but removes the need for network-originated signalling and resolves the scalability issues associated with a network-controlled approach as no communication is required between the two candidate networks. There are three common circumstances that will cause degradation in the link quality. Firstly, consider that a user

The other circumstances that also lead to link quality degradation for a user are, secondly, when another user sends or receives data from the same AP. This will reduce the amount of idle time on the medium and with increasing numbers of users, will ultimately cause congestion and degradation in the quality of the voice signals that are being transported. Thirdly, another user that has an on-going data transfer session (perhaps another VoIP call) moves away from the AP. In this case, the station that moves experiences reduced radio channel quality which results in that station seizing the medium for longer to transfer its IP packets. This results in a reduced service time for all the other users in the WiFi coverage area. These three circumstances that cause a change in the available idle time on the medium will have little impact on Voice quality if the WiFi coverage area is experiencing traffic loads from only two or three other duplex data sources. In such a scenario the only reliable way to determine whether to execute a handover is to measure the RSL. This has been studied elsewhere and is not dealt with in our work. However there is no reason why information about the link quality cannot be integrated into our scheme, thereby improving both schemes. When the load begins to become significant, e.g. >5%, then the quality of a VoIP call begins to be affected by changes in the idle time. Under these circumstances, a user’s RSL measurement gives a very poor indication of whether a handover is appropriate or not. In two of the situations

outlined above, the congestion is caused by other users so that the VoIP client will experience no change in its RSL. In the other scenario the congestion level effectively reduces the size of the AP’s coverage area for VoIP terminals that require some level of QoS, so the RSL triggered handover also needs some measure of the level of congestion within the cell. The work presented here specifically targets the three circumstances that cause a degradation of voice quality to trigger a handover decision. For layers of the protocol stack above the network layer (IP in this case), the performance degradation caused by this increase in offered load manifests itself firstly as increased jitter, then increased packet delay, and finally packet loss. By monitoring packet delay and/or jitter it is possible to detect link quality degradation and perform a handover before packet loss occurs and thereby maintain an uninterrupted stream of packets [8], [9], [10]. This paper presents a set of experiments that were carried out on a real network using the approach that was first reported in [11]. The theoretical understanding of that simple handover was further extended through simulations in the work presented in [8]. The measurements shown in [9] demonstrated the possibility of using a Quality-Oriented handover scheme to maintain an unbroken stream of video packets as a user moved from one WiFi cell to another. The theoretical work was further extended in [10] when a utility function was used. The work presented in this paper takes the significant step of extending the Quality-Oriented handover concept to networks with different PHY and MAC layers, and where there are no high level agreements for handover between the networks. IV. EXPERIMENTAL SET-UP The handover was facilitated by using a modified version of the Stream Control Transmission Protocol (SCTP) that was capable of making handover decisions on the basis of end-toend link quality. This was implemented in a laptop that was connected to a peer through both a WiFi link and a WiMAX link (Figure 1). Both machines used modified versions of SCTP. The server was modified to transmit monitoring traffic

simultaneously on both paths at fixed regular intervals, a mechanism similar to the inbuilt Heartbeat in SCTP. The client was modified to monitor this traffic and make a decision about which path the server should use. This communication required to affect this handover was also included in the SCTP modifications. The load on the network was generated by a number of other computers using WiFi connections to the same AP as the modified client. These loading computers ran IPerf [12] to generate a number of voice-like background calls using the User Datagram Protocol (UDP). Typical G711 communication at 64kbps incurs an overhead of approximately 16kbps for headers that are added through the VoIP Stack. The maximum VoIP rate emulated here then was 80kbps. Lower values were used in some scenarios to test the system’s sensitivity to differing individual loads. Given the CSMA/CD scheme used in IEEE802.11 standards and the substantial overhead incurred by the various layers of the VoIP stack, the scheme was much more dependent on the number of active background stations than the offered traffic load. Since these were all using UDP with no rate adaptation, using all background stations at 80kbps did not give a sufficient granularity to control the experiments. The bit rate allowed fine-tuning of the load, but all values used were representative of VoIP traffic in a heterogeneous network with many types of codecs and applications accessing the medium. The client’s RSL (and link quality) was varied by moving it away from the access point. Figure 2 shows the room layout, and five different positions where tests were performed. By varying the amount of traffic offered by the background stations and the position of the client, it was possible to test the impact of both factors on application performance. The WiMAX link was steady throughout the experiment. It offered a lower level of performance than unloaded WiFi network with a good radio path; however when the WiFi network was used by other clients or the radio path deteriorated significantly, the WiMAX offered the application a more attractive alternative. V. RESULTS The objective of the tests was to show that both contention on the WiFi network and the distance between the client and the Access Point influenced the handover decision, thereby showing that both factors need to be considered. The results are in the form of a Time / TSN (Transmission Sequence Number) diagram. The paths are distinguished by colour and line-type: blue-dashes representing WiFi; red-dots WiMAX. A seamless handover is indicated by an unbroken, straight-line plot. Such seamless handover is a feature of our scheme: there is no need to initiate a new connection to perform a handover.

Figure 2 - Room Layout for experiments

In all three experimental scenarios, handover was demonstrated in both directions. In the sample results shown in Figures 3–5, packets received through the WiMAX link are shown as a dotted line while packet receptions through the WiFi connection are shown as a dashed line.

a) Handover Due to WiFi Congestion The initial primary path was set to be through the WiFi link. A background traffic load of 4 duplex VoIP calls at 80kbps each direction was applied incrementally until the resulting congestion caused the link quality to degrade, triggering a handover to the WiMAX network , as shown in Figure 3.

Figure 4 - Handover cause by modified node moving away

Figure 3 - Handover Due to WiFi Congestion

While the handover plots show that the system is capable of performing a handover due to either increasing network congestion and client mobility, they do not show how the decision depends on both. To do this, five locations (shown in Figure 2) were selected, and the application was tested with varying levels of background traffic. The results in Table 1

b) Handover due to mobility of the modified node The initial primary path was set to be through the WiFi link. Background traffic of 4 duplex calls at 32kbps each direction was applied to the WiFi network. The laptop running the modified version of SCTP was then carried away from the WiFi AP, and the corresponding drop in link quality caused the system to handover to WiMAX. When the laptop was moved back towards the WiFi AP, the call was handed back to the WiFi link. These results are shown in Figure 4. c) Handover due to mobility of another node The initial primary path was set to be through the WiFi link. A background load of 1 duplex call at 32kbps and 1 duplex call at 64kbps was applied from the outset. The laptop that was involved in this 64kbps simulated VoIP call had no handover capability, and was moved from an area where it had access to 11Mbps into an area where it only had 1Mbps access. The results in Figure 5 show that the system handed over the VoIP call on the handover-enabled laptop in response to the decrease in link quality caused by the reduced idle time on the medium caused by this motion of a third party.

Figure 5 - Handover due to mobility of another node show that measuring only the link quality or the congestion would not be sufficient to make the correct selection. The cells shaded grey show where WiMAX was preferred to WiFi: it is clearly dependent on both location and the level of network congestion.

Loc. 1 Loc 2 Loc 3 Loc 4 Loc 5 WiFi WiFi WiFi WiFi 0 Srcs WiFi Wifi Wifi 1 Srcs Wifi WiMax WiMax Wifi 2 Srcs Wifi WiMax WiMax WiMax Wifi WiMax WiMax WiMax 3 Srcs Wifi 4 Srcs WiMax WiMax WiMax WiMax WiMax Table 1 – Network Selection with varying Signal Quality and Congestion Level

[5]

[6]

[7]

VI. CONCLUSIONS An end-point controlled handover has been demonstrated based on a call-quality monitoring system implemented in a modified version of SCTP. This appears to be the first such experimentally demonstrated handover between a WiFi network and a WiMAX network. It is also unusual in that both networks are completely separate, i.e. not under the same administration. As the networks are completely separate, the scheme was necessarily an end-point scheme, and is the first such scheme to be deployed in a heterogeneous network environment. It has been shown here that measuring link quality only is insufficient for deciding on the best network from the point of view of application performance because the signal strength may indicate a good radio path to an access node that is heavily congested and will not be able to support a sufficiently high quality VoIP session. An application-driven end-to-end measurement scheme was therefore proposed. Although the scheme is implemented between Wifi and WiMAX, it is independent of the underlying technology, meaning it could be easily be deployed in other scenarios. Future work includes extension of the scheme to different network technologies and applications, to be coupled with more advanced handover metrics, based perhaps on utility functions. Two further objectives are to implement the scheme on a hand-held device and to reduce the scheme’s overhead, perhaps by only monitoring alternative paths selectively.

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