An Efficient Scheme to Increase the Throughput of VoIP Traffic over Satellite while Reducing Bandwidth Utilization

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1 An Efficient Scheme to Increase the Throughput of VoIP Traffic over Satellite while Reducing Bandwidth Utilization Sayid Mohamed Abdule 1, Wan Tat Chee 1, Ahmed Mustafa 2 & Aisha Hassan 2 1 Network Research Group School of Computer Science University of Science Malaysia Penang, Malaysia 2 ECE Department International Islamic University Malaysia Jalan Gombak, KL 53100, Malaysia Tel: sayidabdule@hotmail.com ABSTRACT Voice over Internet Protocol (VoIP) offers a wide range of benefits to both enterprises and communications & network service providers. It provides an alternate way to make a telephone call as compared to the Public Switched Telephone Network (PSTN). The industry is slowly recognizing the potential of Voice over IP and many VOIP applications have been developed in the last ten years. This paper is based on an ongoing study (Msc) and aims at examining key phenomena that causes major obstacles to VoIP. This paper proposes and presents a new model of packet building namely multiple frames into a single packet (MFSP). We first, have studied the way of the VoIP packets are organized, then illustrated advantages and limitations of existing voice packet s structures. Lastly, we have proposed a new model of packet building namely multiple frames into a single packet (MFSP). The new model mitigates the existing VoIP obstacles such as bandwidth utilization and traffic congestions. The proposed model uses to reduce the traffic through the network and packet overhead. Simulation was done using NS-2 simulator. Simulation results show the ability of MFSP to exploit the bandwidth consumption and traffic minimizing efficiently without effect on packetzing delay. Keywords VOIP, Multiple Frames, congestion mitigation, Bandwidth-utilization. 1.0 INTRODUCTION VoIP technology has been known for long time, but it was in early 1990s when the concept was taken more seriously. Today, there are many private companies or organizations which have their own private networks and mostly are using with Very Small Aperture Terminal (VSAT) technologies (Schulzrinne, 1999). VOIP is rapidly moving from a tactical cost saving effort to a more long term strategy of productivity improvements and reduced cost of network ownership. These include unified management of voice and data infrastructure and perhaps most important the ability to deploy a new generation of converged voice/data applications together. Although use of VoIP is growing rapidly, several aspects have inhibited more rapid adoption across all market segments. These aspects include concerns about maintaining consistent voice quality over IP networks, especially during periods where other types of traffic on the IP network, bandwidth utilities and momentarily threatening voice quality. Generally, the quality of VOIP is less compared with the PSTN (Public switched Telephony Network), but still users are willing to accept the lower quality of Internet telephony because of its reduced cost. Improvements in service, due to increasing bandwidths and more reliable equipment, are improving the quality of internet telephony. Though it has been tried an uncountable solutions to improve the VOIP quality but still it

2 seams there are many obstacles which can compromise the voice quality. Moreover, the technology is a relatively new concept and there are many factors which may influence the VOIP quality. Most of these factors are delay (network delay, propagation delay), jitter(delay variation), packet loss, congestion and the amount of bandwidth required to transmit a voice call over an IP network lead to unpredictable of voice quality (Carlo Demichelis, 1999). This study presents an efficient approach to provide VOIP over satellite namely; multiple frames into a single Packet (MFSP) architecture. Multiple individual VoIP sessions into combined IP packets, significantly reducing the total amount of bandwidth needed to support voice calls over IP networks because of it minimizes the traffic throughput. 2.0 RELATED WORKS As described in (Richharia, 1999) satellite communication systems are fundamental part of most major telecommunication networks throughout the world (Jeremy, Allnutt, 2003). For the last three decades, satellite communications were the essential part of telecommunication system leading the market. The combination of voice over IP & Satellite is recently technology industry which is no much developed (Daniel, 2003). Moreover, the VOIP technology it self is a relatively new concept which is one and half decade since the concept was taken more focuses. This combination proves to be an even more efficient connectivity option as it virtually eliminates traditional telephony long distance costs (Black, 2002). As far as we are considered the combination of these technologies is needed more investigation and this is the one of the major motivation back of this study. According (Net.com, 2004) frame packet is proposed idea about minimizing of bandwidth utilization through multiple frames. This suggestion has been introduced by private company with their white paper in This idea is aimed at one very effective technique to reducing, and in some cases nearly eliminating the IP overhead bandwidth in VoIP networks. But the proposal neither implemented nor simulated. Our work is to simulate the idea through the Geostationary-satellite using NS2 simulator and will be examined behavior of the multiple frames through a satellite with regard to usage of bandwidth and overall impact of the network health including delay. Theatrically, since the satellite is very powerful regarding the tolerance of large packets. Thus, the proposed Idea namely, large packet size is excellent for it (Daniel, 2003) 3.0 NS-2 In order to implement the study, network simulator (NS2) application version 2.28 will be utilized. NS2 is a discrete event simulator meant for networking research (Peterson, 2000) NS2 is the preferred software for the study since it gives Simulation both satellite and non-satellite networks. Moreover, it gives useful information about a system which is being investigated. NS2 is written in OTcl (Object-Oriented Tool Command language) and C++. OTcl is its primary Command and Configuration Language. It implements network protocols such as TCP and UDP over wired and wireless (both local and satellite) networks, and also traffic source behavior such as FTP, Telnet, Web, CBR and VBR (Kevin, 2003), 4.0 TOPOLOGY The topologies used in the ns-2 implementation consist of three main components. The telephone which provides the VoIP stream packets The routers that perform multiple frame packaging and transmit to the satellite The satellite node that provides the satellite links between the two sites. The topology assumes a group of telephones situated in two separated sites. Each group is connected to one router which has a satellite transmitter and receiver that can send and receive data with a geostationary satellite. The telephones send data with a constant bit rate (CBR). This bit rate depends on the voice codec used at the telephones. The packet size and the interval between each packet take its value form the specification of the voice codec.

3 5.0 PROPOSED PROTOCOL The proposed model is simulated by Ns2 (Network simulator second version) with 7 nodes. The simulation nodes and satellite terminals (stations) are organized as follows: Since all our traffic is traveling through over Satellite, it has to establish two satellite terminals, assuming that one is located in New York while another is located in Sun Francisco. Each of them has been equipped network infrastructure such as satellite antenna, satellite alignments (satellite-location, altitude and longitude), VoIP gateways and telephone terminals (traffic generators). As described elsewhere in the study, the process goes through 7 nodes and they are as following: smaller voice frames are forwarded to the sink node. 3. MFSP-NY-sink-Node is the telephone receiver device representative which receives the smaller voice frames from MFSP model and then play pack. 4. Sat-Node is the satellite node which the traffic from both sites goes through. The task of this node is only to transmit the data. 5. MFSP-SF-Node, MFSP-SF-Sink and SFsource-Node function similarly to the NYsource-Node, MFSP-NY-Node, and MFSP-NYsink-Node explained above. Figure 1 illustrated below shows the above mention process containing the seven nodes. 1. NY-source-Node 2. MFSP-NY-Node 3. MFSP-NY-sink-Node 4. Sat-Node 5. MFSP-SF-Node 6. MFSP-SF-Sink 7. SF-source-Node 1. The NY-source-Node is the node which creates the traffic, namely the analogue telephone devices. The number of these data generators could increase and decrease as it always depends on the analysis criteria. For instance, in this study, we use 20 telephone devices which are randomly connected to the VoIP gateway, namely MFSP-NY-Node. 2. MFSP-NY-Node is the actual VoIP gateway node which is being implemented in our proposed MFSP model. This node receives smaller voice frames from NY-source-Node (from normal phones). In addition, this node has ability to collect several frames and then put them into a large packet and subsequently forward it to the destination node through satellite. The MFSP-NY-Node has also another function, namely regrouping the received packet. In other words, the MFSP-NY-Node has a mechanism which enables the large (MFSP) packet sent from the other site to be regrouped in manner that the telephone devices can play back as a normal 20 bytes payload after codec s behavior (e.g. G729 codec). Finally, these Figure 1: Proposed model s process: Source: Author

4 6.0 EVALUATION OF PROPOSED PROTOCOL Since we have now simulated the proposed protocol in NS2 and conducted the experiments, and evaluated the usefulness of the proposed protocol. The goal here is to analyze the behavior of our protocol under a range of various scenarios, among of them are (bandwidth utilization, end to end delay, delay jitter or variance delay and overall network throughput reducing). The following Simulations were analyzed for MFSP protocol with different frame sizes. A number of experiments with regard to the number of frames (MFSP) were conducted while calculating bandwidth consumption, end to end delay and finally single packet bandwidth consumption was compared to MFSP with 4 frames. Figure 2 presents the bandwidth utilization for MFSP with 4 frames which indicates that the maximum bandwidth usage is kbps and this value was compared to the following single packet bandwidth usage. B a n d w d it h u s a g e ( K b p s ) MFSP with 4frames bandwidth utilization 0 Time MFSP with 4frames bandwidth utilization consumption is very high according the bandwidth consumption for MFSP model. While we conducted the experiments we discovered that the single packet scenario is 3 times more traffic generated than MFSP model. Of course there are many reasons that single packet scenario uses much bandwidth than MFSP model but more traffic generating could be of these reasons that it uses more bandwidth. B a n d w id t h u s a g e ( K b p s ) bandwidth utilization for single packet time bandwidth utilization for single Figure 3: Bandwidth utilization for single packet Figure 4 shows MFSP vs. single packet which presents the bandwidth consumption. The figure 3 illustrated that the scenario with MFSP model has minimum bandwidth consumption while the scenario with a single packet has the maximum bandwidth utilization. This strongly shows that the multiple frames protocol is better solution than the way of the existing VoIP packet is organized (single packet). Figure.2: MFSP with 4 frames bandwidth usage Figure 3 shows the bandwidth usage for single packet. This shows that the bandwidth

5 B a n d w id th u s a g e (K b p s ) MFSP Vs. Single packet Time bandwidth utilization for single MFSP with 4frames bandwidth utilization Figure 4: MFSP vs. Single packet Figure 5 Shows the round trip delay round trip delay for MFSP you can see the delay effect of proposed protocol is relatively reasonable which is 5 ms comparing with the single packet. R o u n d t rip d e la y ( S c ) Round Trip delay for MFSP with 4frames Figure 6 shows the single round trip delay. It proves that maximum delay is ms, only 5 ms less than MFSP with 4 frames. R o u n d T rip D e l a y ( S c ) Round Trip delay for single packet Round Trip delay for single packet Figure 6: Round trip delays for single packet 7.0 CONCLUSIONS In this paper which based on an ongoing research (Msc), it has clearly demonstrated enhancement of bandwidth utilization through MFSP protocol. The experiment shown also the new protocol not affected on delay issues neither propagation delay nor processing delay. In addition, it has addressed how the major network challenges of VoIP can be overcome, including reducing number of the traffic through the network. Joining these results, an overall solution much improved than what is experienced in today s VoIP designs can be obtained. Round Trip delay for MFSP with 4frames Figure 5: Round trip delay for MFSP with 4 frames

6 REFERENCES Schulzrinne H. (1999) Converting on Internet Telephone: Service for Telecom. Version 2, IEEE Internet Computing Carlo Demichelis, (1999), CSELT; Enrico Petrov, CSELT - "Instantaneous Packet Delay Variation (ipdv)", IEEE Workshop on QoS support for Real-Time Internet Applications -Vancouver, Richharia, M. (1999). Satellite Communication System; McGraw-Hill, New York Jeremy E. Allnutt; (2003). Satellite Communication, John Wiley &Sons, Daniel Enns,, (2003), Satellite Network Products at Comtech EF Data. Black, U. (2002),Voice over IP, Prentice Hall PTR.,. Net.com,(2004), Network equipments Technologies, white paper, Daniel, E, (2003).Satellite Network Products at Comtech EF Data Peterson, L. (2000), Computer Networks, Academic Press Kevin, F. (2003), the VINT Project, the NS Manual.