An Approach to Transport Layer Handover of VoIP over WLAN John Fitzpatrick, Seán Murphy and John Murphy Performance Engineering Lab UCD School of Computer Science and Informatics University College Dublin Dublin 4 Ireland E-mail:
[email protected] Abstract-In this paper a transport layer handover mechanism for Voice over Internet Protocol (VoIP) using the Stream Control Transmission Protocol (SCTP) is proposed. This handover mechanism utilizes the multi-homing feature of SCTP to allow connection to multiple wireless networks. Each available wireless network is probed with simulated VoIP data to obtain network performance metrics. Handover decisions are made based on the Mean Opinion Score (MOS) calculated from the ITU-T E-Model for voice quality assessment using the obtained measurements. Implementation and simulations were carried out using the NS2 network simulator. The results show the MOS obtained using the proposed handover mechanism on IEEE 802.11b WLAN access points (APs) under varying load conditions. High correlation is shown between the MOS from the proposed handover scheme and the MOS experienced by a VoIP node present in the network. A simulation showing an SCTP endpoint handover between AP’s is presented. Index Terms— SCTP, E-Model, VoIP, Handover, Heterogeneous Networks
I. INTRODUCTION The ever increasing demand by users for ubiquitous access to wireless services has led to the deployment of many wireless access technologies. Wireless access technologies such as WLAN, GPRS, EDGE, 3G, and Wi-Max all offer differing levels of quality, range and bandwidth. In the future there will be more multimode devices which can access multiple radio access networks. Moreover in the future we will see greater overlap between the coverage provided by the differing access technologies. The common view among researchers of Fourth Generation (4G) mobile communication is that it will be a heterogeneous network environment, offering seamless services across multiple wireless access technologies. All these wireless access technologies are anticipated to be Internet protocol (IP) based. For services such as VoIP to operate seamlessly across multiple wireless networks robust handover mechanisms are needed. Mobile IP has been proposed as a network layer host mobility
solution. However, two key problems with Mobile IP are high handover latency and network infrastructure modifications. Mobile IP requires network layer support with the deployment of special nodes within the network. These problems make Mobile IP unsuitable as a mobility solution for real time interactive applications, such as VoIP. There is still a lot of debate within the research community about the optimum layer to perform handovers. In [1] each of the layers was analyzed as a candidate for implementing a mobility solution. In this work it was concluded that implementing mobility solutions at the transport layer was most suitable. The most common IP transport protocol in use today is the Transport Control Protocol (TCP). TCP is a reliable connection oriented protocol for providing data services. TCP was designed to operate in wired networks where packet losses occur mainly due to congestion. Using TCP becomes problematic when it is applied to wireless heterogeneous networks. Wireless network environments have higher loss and error rates. When a loss is experienced TCP reduces the transmission rate as it assumes losses are due to congestion, this leads to degraded performance. If a transport layer handover mechanism is to be used in heterogeneous network environments, then a more suitable transport protocol is needed. Stream Control Transmission Protocol (SCTP) is the third transport layer protocol to be ratified by the Internet Engineering Task Force (IETF). Originally developed to transport Signaling System 7 (SS7) across IP packet networks. SCTP provides reliable, connection oriented communication to endpoints that may have multiple IP addresses. Allowing a connection to span across multiple IP addresses is known as multi-homing, and it is just one of the features of SCTP which has researchers interested in using it as more than a signaling transport protocol. Since the multi-homing feature of SCTP provides an effective method for attaching to multiple IP networks, our work uses this feature to allow seamless handovers in heterogeneous network environments. Similar work was proposed by researchers at the
University of Oklahoma, with a mechanism called SIGMA, a seamless handover scheme for data networks [6]. Their work used the multi-homing feature of SCTP to enable seamless handovers across base stations in different IP domains. Also, in [11] a delay centric handover mechanism utilizing the multihoming feature of SCTP was proposed. In [12] a transport layer handoff solution entitled cellular SCTP (cSCTP) was proposed. This solution utilized the dynamic address reconfiguration extension [13] for SCTP and used the mobility management functionality of Session Initiation Protocol (SIP) to enable seamless handover. Our work differs from those discussed by focusing on making handover decisions based on the quality of service (QoS) obtained from the available networks, specifically for VoIP. Probing is used to obtain network parameter measurements for each of the available access networks. The handover decision is based on the MOS from the ITU-T E-Model calculated using the measured network parameters. It is envisaged that the proposed mechanism be used in heterogeneous network environments. However, initial design and validation of the work as presented here is done in a homogeneous context using 802.11b WLAN APs. This paper is structured as follows. Section II gives an overview of the features of SCTP and the E-Model, pertinent to the proposed handover mechanism. In Section III, the design and implementation of the handover scheme is discussed. Section IV describes the simulation setup and is followed by a discussion of the results obtained in section V. This paper is then concluded in section VI. II. SCTP & THE E-MODEL In this section, the technical details of SCTP and the E-Model relevant to the proposed handover mechanism are described. A. SCTP Stream Control Transmission Protocol is a reliable transport layer protocol, which inherited many of the core features of TCP such as, congestion control and retransmission. SCTP also includes enhancements over TCP. A central concept in SCTP is the definition of an association. An association in SCTP is analogous to a connection in TCP. A key difference between TCP and SCTP is that in SCTP an association can span multiple IP addresses, as can be seen in Figure 1. An important feature of SCTP is Multi-Streaming. This feature allows independent streams of data to be transmitted across a single association with no reliance on the delivery order of packets in other streams. Multi-streaming is used in SCTP to solve the TCP problem of head of line (HOL) blocking that arises from TCP’s strict byte ordered delivery. Another important feature of SCTP is multi-homing. SCTP
was designed as a reliable transport protocol for critical signaling data and hence high reliability was needed. To this end the capability for spanning an association across multiple IP interfaces increases the link tolerance to physical failures. 1) Multi-Homing An example of SCTP multi-homing is shown in Figure 1. In this example each endpoint has two physical links and two network layer interfaces at each endpoint. These are bound to a single association. Although there are only two IP addresses used in this example, SCTP has the ability to support many IP addresses defined in an association. SCTP Association SCTP
SCTP IP 1
IP
IP Network
IP 2
IP 1 IP IP 2
Host To Network
Host To Network
Link 1 Link 2 Multihomed
Multihomed
host A
host B
Figure 1 - SCTP Association When an association is setup between two endpoints, one of the IPs at each endpoint is defined as the primary address. All communication to that endpoint is routed to its primary address until a failure is detected or an upper layer specifically requests the use of an alternate IP. 2) Heartbeat Each endpoint maintains a list of peer endpoint IP address states. This is done by every endpoint sending probing messages to each of the remote addresses defined in the association, these probes are known as heartbeats. This mechanism allows the endpoint to know which addresses defined in the association are reachable. A heartbeat is sent periodically every retransmission Time-out (RTO), RTO being a user/application definable time. When a HEARTBEAT packet is received by an endpoint, the packet is processed and a HEARTBEAT ACK packet is sent back. Each HEARTBEAT packet contains a timestamp of when it was sent. When the HEARTBEAT ACK packet is received, the time delay difference can be used to estimate the propagation delay between the endpoints. This is used to estimate the Round Trip Time (RTT) which is important in the context of this work and will be discussed in detail below. B. E-Model The E-model is a computational model for estimating the
subjective quality of a VoIP call. It is standardized by the International Telecommunications Union Technical standards (ITU-T) as G.107 [5]. The primary use of the E-model is in the design of codecs and transmission networks. The output of the Emodel algorithm is a scalar rating of call quality called the R value. Our approach is to use the E-model in real time to obtain voice quality values for each network and make handover decisions based upon this metric. The R value is calculated from: R = Ro − Is − Id − Ie + A Ro : Basic signal-to-noise ratio Is : Impairments simultaneous to voice encoding Id : Impairments due to network transmission Ie : Effects of equipment (e.g. low bit rate) A : Advantage factor The E-model output R can be converted into the more commonly known metric the mean opinion score (MOS). The MOS system is more commonly used to measure how a user rates call quality. The formula for the conversion is given by: 1 MOS = 1 + 0.035R + R ( R − 60)(100 − R) × 7 × 10− 6 4 .5
R 100
From the G.107 recommendation the following ratings are given to a VoIP call: Table 1 - G.107 VoIP call ratings R value (lower limit)
MOS (lower limit)
90
4.34
Very Satisfied
80
4.03
Satisfied
70
3.6
Some users dissatisfied
60
3.1
50
2.58
Many users dissatisfied Nearly all users dissatisfied
User Perception
III. HANDOVER DESIGN This section details the design and implementation of the proposed handover mechanism.
A. Heartbeat modifications The original heartbeat mechanism in SCTP was modified to send multiple heartbeat packets with particular time spacing between consecutive packets. The number of packets and the time spacing between them can both be modified, to allow the probing mechanism to model various VoIP encoding schemes. To correctly estimate the MOS of a call on a WLAN, the heartbeat mechanism needs to mimic the behavior of the VoIP
codec being used. A VoIP call using the G.7111 codec can be approximated as sending out 80 byte packets every 10ms. Therefore, each heartbeat packet is 80 bytes long with a time spacing of 10ms set between them. This means that the SCTP heartbeat probe very closely simulates G.711 encoded VoIP data.
B. Implementing the E-model While the E-model was primarily designed as a network planning tool, some work has been done on applying the EModel to real time environments. In [2] specific values were chosen for parameters within the E-Model. This simplified variant can then be used in a real time context. Using their work the E-model calculation can be reduced to: R = 93.34 − Id − Ie Id takes into account network delay and jitter parameters. Ie is the equipment impairment factor and is mainly codec and loss dependent. In the ITU-T recommendation G.113 [14], the values for Ie under varying loss conditions are given for each codec. Within a VoIP system only Id and Ie are variable. One of the delay parameters on which Id is dependent is jitter. To obtain values for jitter the E-Model recommended RTP jitter algorithm from RFC1889 [10] is used. C. Handover decision Network measurements are used to calculate the MOS of each network. The measurements are calculated using the RTT values obtained from heartbeat packets. A train of heartbeat packets is sent to each endpoint in the association every T seconds. The number of heartbeat packets in each probe train is important for the accuracy of the handover mechanism. In the results section, finding the optimum number of heartbeats to use will be discussed. Every T seconds the E-model algorithm calculates a MOS for each network link. These values can then be used to make a handover decision. The MOS values are passed to another algorithm to decide which network link to use. Essentially the handover mechanism will select the network link that offers the best quality of service (QoS). The QoS metric for call quality is the MOS and so this is used to make the handover decision. A small amount of hysteresis (0.1 MOS) is used to prevent rapid handovers due to small network fluctuations. If loss is present in the network not all heartbeat packets will be successfully received. For each probe train a percentage loss calculation is made. The Ie parameter can be obtained based on the loss calculation and the codec from the values given for each codec in [14]. The handover is done using a standard feature of SCTP that 1 There are many different codecs used for VoIP. G.711 was chosen as it is the simplest and therefore is commonly used. Also G7.11 is supported by the majority of voice gateways, unlike other codecs.
allows a new primary address to be selected from the list of alternate addresses. This feature allows the primary interface, through which all the data is flowing to be changed. The change does not cause any delay or interruption to the flow of data to the endpoint. This makes the handover seamless and transparent to the user. IV. SIMULATION SETUP Implementation and simulations were carried out using the Network Simulator 2 (NS2) [7]. NS2 is a discrete event simulator for networking research. It provides implementations of commonly used protocols and access technologies. An SCTP implementation for NS2 from the University of Delaware [8] was used as the basis for the handover scheme. Figure 2 shows one of the simulated scenarios. Other simulations used the same topology but differed in the number of nodes. The wireless SCTP node is in the overlapping region of both WLANs and has a connection to both networks. The SCTP endpoint is where the VoIP call from the wireless SCTP node would be terminated, and so it is the quality of the link to this endpoint that is being measured by the wireless node. To simulate wireless background traffic, constant bit rate (CBR) sources are used. The G.711 codec is being modeled and so each CBR source transmits 80byte packets every 10ms, this gives a data rate of 64kbs, these settings simulate G.711 simplex voice traffic. In NS there is no support for full duplex links. Duplex traffic was modeled using two simplex traffic sources.
number of heartbeat packets. Seven VoIP calls were placed on a WLAN. Seven was chosen as this number of calls loads the network and creates a high amount of delay variation and loss. Hence, this is a difficult region in which to estimate the call quality. The length of the probe was varied from 3 to 100 heartbeat packets. In Figure 3 the results of the simulation runs are shown. The MOS experienced by a CBR traffic source is compared to the MOS estimated by the proposed SCTP probing mechanism. There is a tradeoff in selecting the optimum probe train length, between the accuracy of the measured parameters and the amount of load the probing places on the network. 25 heartbeat packets were chosen as the probe train length, as it gives an accurate estimation of the MOS without needlessly loading the network. Using the optimum heartbeat probe train length of 25, simulations were setup to verify the accuracy of the mechanism over a varying number of VoIP calls. As can be seen in Figure 4 there is a high correlation between the actual MOS experienced by a CBR source and the MOS estimated by the proposed mechanism. Each data point on the graph is the result of the mean of 5 simulation runs. The max and min of which are shown by the error bars. The values corresponding to seven VoIP calls have the largest standard deviation with a value for the estimated MOS of 0.0764 and 0.0524 for the actual MOS experienced by a CBR source.
Figure 2 - Simulation Topology
WLAN1 contains multiple simulated VoIP calls, WLAN2 is left free of any other nodes so that the only traffic present is the SCTP data from the wireless node. All nodes were situated in close proximity to one another and to the AP. This was done so all nodes would have the same data rate and also eliminates any hidden terminal problems. V. RESULTS A set of simulations were setup to calculate the optimum
Figure 3 - Optimum Number of Heartbeats
With the accuracy of the MOS estimation mechanism established, a simulation utilizing the mechanism to perform a seamless handover is presented. A simulation using the topology shown in Figure 2 was setup. The results of this simulation are shown in Figure 5. Three calls were placed on WLAN 2 giving a mean MOS of 4.39. The time interval T between handover
decisions was set at 2 seconds for this simulation. Initially at time 0 there was one call present on WLAN 1, then every 100 seconds a further one call was added. At time 500 seconds the sixth call was added to WLAN 1. As can be seen the MOS of WLAN 1 then falls below that of WLAN 2. A hysteresis value of 0.1 MOS was used to prevent rapid handovers due to small network variations. This means that a handover from WLAN 1 to WLAN 2 does not take place until 504.74 seconds.
given showing how the proposed scheme estimates call quality. The accuracy of the results obtained by the proposed mechanism were demonstrated by showing the high correlation between the estimated MOS and the MOS experienced by a traffic source present in the same network. A simulation showing a handover based on the MOS of each available network was presented. ACKNOWLEDGMENTS The support of the Irish Research Council for Science, Engineering and Technology (IRCSET) and the Informatics Research initiative of Enterprise Ireland is gratefully acknowledged. REFERENCES [1] [2] [3]
Figure 4 - Estimated MOS for varying number of VoIP calls
[4]
[5] [6]
[7] [8] [9] [10] [11]
[12] [13]
Figure 5 - Simulation showing Handover from WLAN 1 to WLAN 2
VI. CONCLUSION In this paper a new VoIP handover scheme for heterogeneous networks has been described. The relevant features of the Stream Control Transmission Protocol were also detailed. An ITU-T recommendation for estimating end user perceived quality for VoIP, called the E-Model was also discussed. Modifications to the E-Model to enable real time quality assessment measurements were used in the proposed handover scheme, to estimate call quality on each available network. Results were
[14]
Wesly M. Eddy. “At What Layer Does Mobility Belong”.IEEE Communications Magazine: 155-159, Oct 2004. Psytechnics. “Estimating E-Model Id within a VoIP Network”.Technical Report 2002. Haowei Bai.”Transport Layer Design In Mobile Wireless Networks”. Invited book chapter in Design and Analysis of Wireless Networks, Yi Pan and Yang Xiao (Eds.), Nova Science Publishers, 2004. R. Stewart, Q. Xie, K. Morneault , C. Sharp, H. Schwarzbauer, T. Taylor, I. Rytina, M. Kalla, L. Zhang, V. Paxson.”Stream Control Transmission Protocl”.RFC2960, October 2000. ITU-T.”The E-Model, A Computational Model In Use In Transmission Planning”.G.107, March 2003. S. Fu, L. Ma, M. Atiquzzaman and Y. Lee, “Architecture and performance of SIGMA: A seamless handover scheme for data networks,” Accepted for presentation at ICC, Seoul, South Korea, May 16-20, 2005. “The Network Simulator- ns-2”.Available at http://www.isi.edu/nsnam/ns/ A. Caro Jr, J Iyengar”NS-2 SCTP Module”.http://pel.cis.udel.edu/. L. Ong, J. Yoakum.”An Introduction To The Stream Control Transmission Protocol (SCTP)”.RFC3286, May 2002. H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson.” RTP: A Transport Protocol for Real-Time Applications”.RFC1889, Jan 1996. A. Kelly, GM. Muntean, P. Perry and J. Murphy. ”Delay Centric Handover in SCTP over WLAN” Transactions on Automatic Control and Computer Science, Vol.49, 2004. I. Aydin, W. Seok and C. Shen. “Cellular SCTP: A Transport-Layer Approach to Internet Mobility” ICCN 2003. R. Stewart, M. Ramalho, Q. Xie, M. Tuexen, I. Rytina, M. Belinchon, and P. Conrad, “Stream Control Transmission Protocol (SCTP) Dynamic Address Reconfiguration,” February 2003, draft-ietf-tsvwg-addip-sctp 07.txt ITU-T ”Series G: Transmission Systems and Media Digital Systems and Networks” G.113, February 2001.
