A Real-time Congestion Control Mechanism for multimedia transmission over 3G Wireless Networks

Xinxing Zhao1 Yuning Dong1 Hai-tao Zhao1 Zhang Hui 1, 2 Jiazhou Li3 Can sheng 3
1. College of Communication and Information Engineering, Nanjing University of Posts and Telecommunications Nanjing 210003, China 2. National Mobile Communications Research Laboratory, Southeast University, Nanjing 210096, China E-mail: zhaobofu@gmail.com, dongyn@njupt.edu.cn, zhaohtmail@gmail.com, zhhjoice@126.com 3. ZTE Communications Co Ltd, Shenzhen (China)
Abstract-In this paper, we first investigate the performance of the 3G WCDMA wireless channel, and we find a special character, that sometimes a sudden link blockage/ congestion happens and causes heavy throughput drops during a certain time in the given 3G channel. After getting the basic characteristics and information about the performance, a novel transmission control scheme is proposed based on the Gilbert wireless error model by analysis of the characteristics which we notice of the wireless channel. The proposed scheme takes into consideration of the link blockage/sudden congestion which last for a short time in the 3G networks, the packet loss rate, and congestion level to increase packet successful delivery rate, reduce congestion and to improve the overall performance in the 3G networks by a adjustable rate control mechanism. Extensive simulation results demonstrate that the proposed scheme is effective in rate control and in decreasing the congestion level.

channel error, or caused by the elements like the storm or the lightening, can be detected clearly, we may improve deliver efficiency effectively and the overall performance of the networks. In this paper, we propose a RTT (round trip time) based rate control (RBRC) scheme, where the sender can detect the sudden losses by compare the current RTT with an average one constantly and adjust its sending rate properly according to the congestion level and the current RTT value. II THE GILBERT-ELLIOTT CHANNEL MODEL As we observe the data we got from UMTS networks in Nanjing, we find that, no matter what the sending rate is, there are always sometimes the sudden blockage/congestion happens, and after we record and represent the statistics we got in coordinates, we find that the throughput, the packets loss rate, the RTT values are correlative, by that we mean they are synchronizing. Fig.1 to Fig.3 show the performance of 3G network with the sending rate 128Kbps. In all these figures the overall recording time is about 4 minutes. We can observe that during the certain span of a time, the throughput, the loss rate, the RTT are intertwined, i.e. the throughput drops heavily, the loss rate increases dramatically when the RTT increases a lot than the average ones, and the throughput, the loss rate keeps fluctuate a bit around a normal level when the RTT keeps around a average value. In fact we have recorded a lot of different groups of data with different sending rate and had each a different time span, they all have the similar characteristics, by that we mean the throughput, the packets loss rate, the RTT values are correlative, so we use the 128kbps with about 4 minuets span here to serve as a typical example.

INTRODUCTION

With the rapid development of the wireless networks, more and more applications based on wireless multimedia networks are now emerged. To improve the overall performance of these wireless networks, lots of researches have been made, and among which the rate control mechanism becomes the popular ones. An Explicit Congestion Notification (ECN) [1] based congestion control algorithm controls sending rate by marking the IP packets but can not control rate properly when a successive packets loss happens. The end-system based source algorithm detects the nature and discriminate packet losses types to improve network performance by relative one-way trip time (ROTT) [2] or packet inter-arrival times [3].The performances of all these algorithms depend on network topologies and the number of competition flows, and most of all they can not deal with the sudden link blockages/ congestions when the throughput drops heavily due to various reasons. Our focus is to design a rate control mechanism which can adjust the sender behavior smoothly and when sudden congestion happens we can detect it and control the sending rate instantly, thus we may get a higher utilization of the networks, improve the throughput, increase the packet successful delivery rate of the wireless networks. To better support multimedia transmission over wireless networks, we believe that if the sudden blockage/congestion which last for a short time span as we mentioned early due to different causes, such as the actual congestion losses,
This work was supported in part by National Natural Science Foundation of China (No.60972038), the Jiangsu Province Universities Natural Science Research Key Grant Project (07KJA51006), the Natural Science Foundation of the Higher Education Institutions of Jiangsu Province (No.10KJB510013), the Open Research Fund of National Mobile Communications Research Laboratory (N200911), Jiangsu Province Graduate Innovative Research Plan (CX09B_149Z), and ZTE Communications Co Ltd (China).

Fig. 1 The throughput of the 3G network.

G =

PBG PBG + PGB PGB B = PBG + PGB

(2)

(3)

III RATE CONTROL DESIGN BASED ON CONGESTION LEVEL As we mentioned above, there is certain time when sudden blockage/congestion happens, in fact the congestion loss actually happens a lot during all the process, and caused by various reasons, the major difference here is that when sudden blockage/congestion happens, the RTT values is much longer than the average ones, and the time span of it is about1 to 4 seconds each time form the groups of data we got while other kinds of congestion loss last longer. So the proposed algorithm considers the long congestions loss and short lasted congestions mentioned above as two different events and deal with them differently, first we consider the longer ones. We know that the long congestion may be caused by different reasons, such as channel error or a sudden link break. When this happens we deal with it by following mechanism, TFRC [6] (TCP friendly Rate Control) algorithm is a widely used equation-based [7] end to end mechanism, it adopts an equation as shown by (4). Where s is the size of the packet, RTT is the round trip time, RTO is retransmission time, p is the packet loss rate, and a modification here is that we calculate loss event rate k instead to approximate the packet loss rate p when we calculate the p in (4). And in turn we get the loss event rate which denotes the k indirectly by an average loss interval s number of received packets between two immediate loss . events, and k= 1/ s

Fig. 2 Packet loss rate of the 3G network.

Fig. 3 The RTT of the of the 3G network.

In order to simulate the 3G wireless channel, we use the Gilbert-Elliott wireless error model here, the GE model [4]. It is a two state one-order Markov model. We have a good state and a bad state as shown in Fig.4, the packets loss probability by the wireless channel is determined by the transition of the states. The losses occur with lower probability PG when in good state while in the bad state they occur with higher probability PB. The PBG is the probability of transforming from a bad state to a good state while PGB is the transition from good state to bad state. The steady state probabilities of being in the state of G and B are PGG and PBB respectively. The average packet loss rate produced by the model is shown by equation (1), we use G denotes the probability when a channel is in a good state, while B means the probability of a channel in a bad state, and with the relation (2) and (3).

RL =

s 2p 3p RTT + RTO(3 ) p(1 + 32 p 2 ) 3 8

(4)

Fig. 4. The Gilbert-Elliott channel model

For the sake of simplicity, here we use the similar settings as in [4, 5], that is, we set PGB0.04 and PBG0.06. And specially we here set PG 0, that means when a channel in the good all sent packets through it will be accept by the receiver end successfully without a wireless error, thus we can simulate the wireless loss under different strength of error interference only by change the value of PB. Pavg = PG G + PB B (1)

In order to get a more smoother sending rate we adopt with certain modification a MIMD (Multiplicative-Increase Multiplicative-Decrease) [8] rate control method here, and describe it as following: a and b are increase and decrease parameters of the MIMD algorithm which play a dominant role in the rate control process. First, we start at a congestion point as a new beginning of a congestion cycle, and during this congestion control cycle (one or more RTT time), the transmission end holds the sending rate R. If there is no congestion packet loss occurs during the control cycle, then the sending ends rate will be (a +1) * R, otherwise, the sender's transmission rate will be (1-b) * R. The congestion control cycle length can be different when increase or decrease the sending rate, in general, the congestion control cycle length when reduce the sending rate is shorter than that of the length when increase the sending rate, of course this is a proper action, since if we find that the congestion happens, we had better decrease the sending rate quickly, while when the channel sate seems quite good we had better get more information about it,

rather to increase the sending rate immediately, after we get a full information about the channel, we then increase the sending rate. Generally we reduce the sending rate when congestion happens during a RTT while we increase it when there is no congestion happens during 3 RTT time. So the rate control mechanism based on congestion level can be described as following: In the absence of packet loss the sender's rate control mechanism completely rely on to the modified MIMD (a, b) congestion control mechanism, we set a = 1/5, b = 1/10 in our simulation. If there is a packet loss, then first of all we use the equation (4) of the TFRC congestion control algorithm to calculate the throughput, and use this throughput according certain rule to get yet another sending rate, if the another sending rate is quite different from the current sending rate then do not use it, but let MIMD (a, b) congestion control mechanism to control it. IV TRANSMISSION CONTROL MECHANISM BASED ON RTT LEVEL Now we consider the second sort of the congestion, which happens unpredictably and lasts for a comparatively short time, and the reason for it may due to some interference which is not known for sure. As observed from figure 3, when the sudden congestion happens the current RTT is much larger than the normal level, and can last as shown in same figure at most about 4 RTT time. So we can develop the rate control mechanism as following: We define a queue which can record the latest 8 RTT (since the time span when the sudden congestion happens is less than 8 RTT generally) values, which can update as the process going on, and we can get the average RTT with these recordings, compare the current RTT with the average one, if the current RTT is 40 percent or more than the average one, we drop the sending rate to 2/3 of the calculated rate in part III, if not, we let the rate control mechanism in part III take the charge, thus we may achieve a better networks utilization and can adjust the sending behavior more smoothly.

Fig. 5 The flow chart of the RBRC

mechanism.

So the overall congestion rate control mechanism can be described as following: When the sending process started, our rate control mechanism is similar to that of the slow start phase of the TCP congestion control. Our sender at first at a low sending rate, and doubles its sending rate for every RTT after having received the acknowledgment packets until the first congestion loss packet was detected, we start the congestion mechanism, and we record the RTT and update the queue, if the congestion happens and belongs to the first kind we give it to the mechanism proposed in part III, and give it to the mechanism in part IV if it belong to the other kind, and thus we may achieve a better networks utilization and can adjust the sending behavior promptly and more smoothly according to the state of the networks. The over all mechanism is represented as figure5.

V SIMULATION RESULTS The performance of the proposed rate control mechanism is evaluated by using NS-2 simulator over a wireless channel. A. Simulation environment and performance metrics We consider a wireless last hop network topology (see Fig. 6), which with 3 wired nodes,1 base station node and 2 wireless nodes that spread over an area of 500500m2, the simulation runs for 500 seconds, and the default routing protocol is DSDV. The Gilbert error model is used to simulate the 3G wireless channel. In Fig. 6 the dotted links are wireless and other parts are wired links with the same settings as in [9] in our simulation. We send one data stream from node 1 to node 5, and an ON/OFF exponential UDP flow from node 0 to node 4 as the interference flow with the following settings: packet size 800KB, mean ON time 1.2 sec (since the span of the sudden link blockage/congestion time usually lasts for 1 to 4 sec). We set 3 groups of sending rate: 100, 200, 400kbps; 4 groups of mean OFF time: 20, 40, 60, and 80 seconds for the ON/OFF UDP flow. So when the UDP flow starts, it will compete with data flow in the wired shared link, which serves in our simulation as the sudden congestion source. The design goal of the RBRC mechanism is to realize the dynamic adjustment of the sending rate, to achieve a better network throughput and reduce packets loss probability caused by various reasons. Therefore, we evaluate the performance of the proposed mechanism by two metrics, the network throughput and packets loss rate, with the wireless channel bit error rate BER = 5 10-5.

RL = RTT

s 2p 3p + RTO(3 ) p( 1+ 32 p2 ) 3 8

0
sender sender

150Kbps,10ms 10Mbps,1ms Wired shared link

4
receiver

2 1
10Mbps,1ms

260Kbps,10ms

3
receiver
150Kbps,10ms

5
Fig. 6 The wireless last hop network topology.

B.

Simulation Results In the network topology, there are 6 nodes which are distributed over the area of 500500m2. The wireless nodes are within the communication distance from the base station. Since the SPLD [10] and ZIGZAG [11] algorithms are popular protocols used in differentiating different loss types and controlling the rate accordingly, so we compare our mechanism with them in the same simulation environment. One can see from Fig. 7, the performance of each protocol in the given network topology. As shown in Fig.7, the normalized throughput (the ratio of actual throughput to 150kbps) of all flows increases as the mean OFF time is getting longer and as the interference sending rate getting lower. The proposed RBRC mechanism has the best performance, maintains a comparatively high level throughput. From Fig. 8 we can see that the average packet loss rate (congestion + wireless losses) of the proposed mechanism is lower than that of the ZIGZAG or the SPLD under different interference sources. The superiority of the RBRC is mainly because it can detect the network state better than others when the sudden link blockage/congestion happens and can adjust the sending rate more reasonably and promptly.

VI CONCLUSION In this paper, after finding the sudden link blockage/congestion that happens from time to time in the UMTS 3G channels, we develop a congestion control mechanism RBRC to deal with the congestion which lasts for a few seconds, as well as the normal congestions which last much longer. The RBRC algorithm can adjust the sending rate based on the congestion level effectively and can detect the sudden blockage/congestion by comparing the value of the current RTT with the average RTT which can be got by an updating queue. After the comparison, we then adjust the sending rate according to certain rules. The NS2 simulation results show that the proposed mechanism can detect the nature of the congestion effectively, and take the proper action to adjust the sending rate, improve the throughput and the network utilization with a lower packet loss probability. REFERENCES
[1] H.J. Lee, H.J. Byun, J.T. Lim, TCP-friendly congestion control for streaming real-time applications over wireless networks, IET Communications, 2008, vol. 2, no. 1, pp. 159-163. [2] N. Nguyen, E.H. Yang, End-to-end loss discrimination for improved throughput performance in heterogeneous networks, In: 2006 3rd IEEE Consumer Communications and Networking Conference (CCNC 2006), Las Vegas, NV, January 2006, pp. 538-542. [3] Y. Tobe, Y. Tamura, A. Molano, S. Ghosh, H. Tokuda, Achieving moderate fairness for UDP flows by path-status classification, In: 25th Annual IEEE Conference on Computer Network (LCN 2000), Tampa, FL, November 2000, pp. 252-261. [4] Cheng-Han Lin, Chih-Heng Ke, Ce-Kuen Shieh, Naveen K Chilamkurti, The Packet Loss Effect on MPEG Video Transmission in Wireless Networks, 20th International Conference on Advanced Information Networking and Applications 2006 (AINA 2006), April 2006, vol. 1, no. 18-20, pp.565-572. [5] Zhao Haitao, Dong Yuning, Zhang Hui and Li Yang, A Wireless Multimedia Transmission Control Algorithm over Heterogeneous IP Networks, Journal of Electronics(China), Jan. 2010, vol. 27 , no.1, pp. 29-36. [6] Sally Floyd, et al. Equation-Based Congestion Control for Unicast Applications. In Proceedings of ACM SIGCOMM Conference, 2000, pp.45-58. [7] Injong Rhee, et al. TEAR: TCP emulation at receiversflow control for Multimedia.http://www.csc.ncsu.edu/faculty/rhee/export/tearpage/teachreport/tearf.pdf. [8] Z.W. Zhou, Y.N. Dong, Research on a transmission control algorithm of the real-time video stream AVTC over IP network, Computer Research and Development, 2004, vol. 41, no. 5, pp. 812820. [9] Yu-ning Dong and Meng-yue Chen, Real time video transmission control in wireless-wired IP networks, in Proc. IEEE Wireless Communications and Networking Conference (WCNC2007), Hong Kong, Mar.2007,pp.3687-3691. [10] Min Kyu Park, Kue-Hwan Sihn, Jun Ho Jeong, A Statistical Method of Packet Loss Type Discrimination in Wired-Wireless Networks, 2006 3rd IEEE Consumer Communications and Networking Conference, CCNC 2006, Las Vegas, NV, United States, 1, 2006, pp.458-462. [11] C. Song, C. Pamela, Cosman, End-to-End Differentiation of Congestion and Wireless Losses, IEEE/ACM Transactions on Networking, 11 (5), 2003: 703-717.

Fig. 7 The normalized throughput of data streams under different situations.

Fig. 8 The average packet loss rate of data streams under different situations.