A Secure Architecture for Open Source VoIP Solutions

Fahad Sattar1 and M. Hussain1 Shaheed Zulfikar Ali Bhutto Institute of Science and

Technology, Islamabad, Pakistan. mhussain@szabist-isb.edu.pk,

fsattar@hotmail.com

Kashif Nisar2

Department of Computer & Information Sciences,Universiti Teknologi PETRONAS,

Bandar Seri Iskandar, 31750 Tronoh, Perak, Malaysia. kashif_nisar@utp.edu.my

Abstract—Voice over IP “VoIP” is a form of voice speech that uses data networks, for transmission of voice traffic over IP-based networks to transmit voice signals. The signal is correctly encoded at one end of the communication channel, sent as packets through the data network, then decoded at the receiving end and transformed back into a voice signal. It has a potential and offering a choice to the traditional telephony systems due to its high resource utilization and cost efficiency. Meanwhile, Wireless Local Area Networks (WLANs) have become an ever present networking technology that has been deployed around the world. Driven by these two popular technologies, Voice over WLAN (VoWLAN) has been promising as an infrastructure to provide low-cost wireless voice services. However, wireless technology was not designed to support voice transmission and recently enhancing its service quality is one of the main issues to be solved. Several metrics such as delay, jitter, and packet loss have impact on the QoS of VoIP. Minimizing total transmission delay, jitter and packet loss would maximize transmission throughput thus improving quality of service. In this research a VoIP transmission algorithm would be designed to reduce transmission delay in order to increase quality of service of VoIP. This research work is aimed to utilize IP protocols and provide a method to improve VoIP performance. Keywords: Packet transmission, VoIP, WLAN. 1.1 VoIP Network Systems Voice over IP is a form of voice speech that uses data networks to transmit voice signals. The signal is properly encoded at one end of the communication channel, sent as packets through the data network, then decoded at the receiving end and changed back into a voice signal [1] and [2]. At present, around 1 billion fixed telephony lines and 2 billion mobile-phone lines exist in the world. Now, we are moving ahead to IP network based protocols known as Voice over Internet Protocol [3]. Wireless local area network (WLAN) allows devices within the local radio coverage to be connected together wirelessly. WLAN devices or terminals communicate with each other through an access point (AP) forming a one-hop network. Thus when any terminal wants to send packets to other terminal, packets

would be sent to AP first which will forward this voice traffic to the destination [4]. 1.2 VoIP Network Systems

Figure.1 displays the setup of a VoIP system and how audio signals are processed by it. At the sender the digitized voice of the speaker is encoded by a speech encoder, then packed and sent through the protocol stack. If the speaker is silent, the voice activity detection (VAD) recognizes this and a packet without payload or no packet is generated. Afterwards the packets are sent over the network, for instance an IP-based LAN. Here is a diagram of the process:

Figure 1.1 Processing of audio signals in a VoIP System. At the receiver side the protocol stack processes the packets. Then lost packets are detected and substituted. Since the packets might not arrive in a unvarying flow, they are collected in a jitter buffer that adjusts time differences and the arrival order. Finally, the decoder decodes the packets and outputs them via the sound system. There are a number of standards used for signaling, speech coding and the transport of the voice packets over IP. This modularization makes VoIP flexible and the standards interchangeable when new requirements and applications emerge. The following sections will give an introduction in the most important standards and issues related to VoIP [5], [6], [7] and [8]. The IEEE 802.11 WLANs networks, we called as a wireless Ethernet and play a significant part in future-generation

networks. Medium Access Control (MAC) sub-layer categorizes two functions, Distributed Coordination Function (DCF) and Point Coordination Function (PCF). The IEEE 802.11 WLANs network support both contention-based DCF and contention-free PCF functions. We focus contention-based DCF function. DCF uses Carrier Sensing Multiple Access/Collision Avoidance CSMA/CA as the access method [9], [10], [11] and [12]. 1.3 Problems Statement Network layered architecture does not support the objectives of transmitting real time applications such as VoIP over WLAN. Each layer is functioning independently. Wireless LAN faces different issues with the performance of real time traffic transmission. One of the issues, transmission delay which has a great impact on the throughput of VoIP. Delay may occur for different reasons such as network congestion. Delay may also cause a pocket to be discarded at the receiving end due to time outs, thus raising another issue in VoIP transmission. It is believed that minimizing transmission delay would improve QoS of VoIP. The availability of redundant functions at different layers of the TCP/IP protocol affects the overall performance of wireless applications, increasing the overall transmission delay. Reducing or combing these functions together would help in minimizing transmission delay thus improving VoIP QoS. These above problems degrade the QoS of VoIP to IP-based networks. We need to introduce as algorithm to solve above VoIP traffic issue. New method should be an efficient; using less bandwidth thus will enhance performance of VoIP over WLAN networks. 1.5 Project Aim and Objectives The aim of this research paper is to develop an algorithm that supports the VoIP application over WLAN Networks. We will assume a fundamental related work to examine the available algorithms outcome and drawbacks. In this we will introduce an algorithm to enhance the performance of VoIP over WLAN networks using IEEE 802.11 standards. We will evaluate, examine, and simulate the techniques with related algorithms for real-time applications. By improving the real-time traffic algorithm, it is possible to resolve many of these problems. In this research the specific objectives are as following:

• To develop an algorithm for VoIP traffic that can be proficient to fulfill the scheduling requirements over WLAN network.

• To compare algorithm with other algorithms, to evaluate and validate.

• To enhance the capacity of node over WLAN network, using our test-bed in VoIP Lab.

The paper is prepared as follows. In section II we will discuss the related work with different scheduling algorithms

and initiate their limitation for multimedia application. In section III, we have proposed an algorithm and methodology. In section IV we describe simulation scenario to compare the efficiency between novel VoIP algorithm and other related scheduling algorithm. Section V, we describe the results and in the last section VI we conclude this paper with future research work remarks. 2. Related Work Garg and M. Kappes [13] initiated the channel Utilization deduction for completely evaluating the network capacity and network procedure of a wireless local area network. Novelist in this research work would like to state that the advance sketched in this research work can be comprehensive in a normal way to the QoS improved DCF MAC system as planned in the present sketched of the IEEE 802.11e standard for QoS in wireless networks. As congregated networking in the wired world adds grip, it is likely that wireless networks, specially, Wireless LANs (802.11) will also be gradually more implemented for voice traffic flow. As researches in also exposed, insertion an additional call or an additional data connection that go beyond the capacity of the wireless network will result in undesirable call excellence for all continuing VoIP calls. D. Goa et al. [14] and [15] introduced the employ of the examination inequity provided by IEEE 802.11e EDCA to solve the bottleneck difficulty of VoIP over WLANs and develop the voice capacity. In exacting, the novelist suggests the distribution of advanced priority admission group (AC) to the AP while assigning lower priority AC to mobile stations. The issue is how to choose the best probable EDCA parameters so that the maximal VoIP capacity can be achieved. In this paper, he has shown that, for VoIP over infrastructure WLANs, the Access point is the bottleneck that limits the VoIP capacity. He has proposed the use of the 802.11e EDCA mechanism to provide service differentiation to the AP and the mobile stations to improve the WLAN VoIP capacity. D. Hole and F. Tobagi [16] evaluated an upper bound on the capacity of an IEEE 802.11b network carrying voice calls, and found it to be tight in scenarios where channel quality is good and delay constraints are weak or absent. Then he shown that capacity is highly sensitive to the delay budget allocated to packetization and wireless network delays. In this paper author consider both G.711 and G.729 voice encoding schemes and a range of voice packet sizes. The paper first present an analytical upper bound and show it to be tight in scenarios where channel quality is good and delay constraints are weak or absent. Then author use simulation to show that capacity is highly sensitive to the delay budget allocated to packetization and wireless network delays.

In this paper the author investigate whether IEEE 802.11 devices could be used to create a low-cost wireless voice network that could be integrated with wired Voice over IP networks, or connected directly to cellular networks. Author focuses on the capacity of the wireless network as the principal metric of interest; this is important not only for deployment of these networks, but also as a means of comparing protocols and techniques in future works. The wireless network operates using the Distributed Coordination Function (DCF) MAC protocol (without the RTS/CTS mechanism enabled); although the Point Coordination Function (PCF) protocol was designed to better handle stream-type traffic, this has not been widely implemented. 3. Proposed Solution Methodology Our objective is to implement best codec technique to improve mobile nodes over WLAN networks. Particularly, this technique will implement on IEEE 802.11b standard. In figure 3.1 shows the proposed methodology, where we use one server, with router and one AP (Access Point) system and how audio signals are processed by it through different Mobile Nodes.

Figure 3.1Secure Architecture for VoIP over WLANs At the sender the digitized voice of the speaker is encoded by a speech encoder, then packed and sent through the protocol stack. There are many types of voice codec used in IP telephony, namely, G.711, G.723.1, G.726, G.728, and G.729 [19]. The codec used by us in this scenario is G.711, (G.711 initially accommodated in an 802.11b six nodes on 10 Packet Interval (ms)), G.729 accommodated 6 to 7 and G.723 cannot accommodate any node, due to the reason packet interval increased the nodes will be increased. On the 60 (ms) G.711 managed 25 to 29 node over 64 bit rate (kbps). The standard used in it is IEEE 802.11 b, which support up to 50 users in it. The thing which we are trying to

resolve through this paper is VoWLAN quality through increase of nodes, VOIP quality is closely related to three factors, packet end-to-end delay, delay jitter (delay variation), and packet loss.

Algorithm for Increased Number of Node on G.711 Initialize Traffic Flows:* for ( n = 0; i < n; n = n + 1)** n = 0; Increased the Traffic Flows) If (node (n) = VoIP traffic (VT)) then*** send VoIP traffic to the G.711 Codec Else Send to CBR & FTP; End if; (On arrival of VoIP Traffic) If ( node (n) <= VoIP Traffic then send to VoIP Traffic End if;Parameter *Initialize = Number of Nodes. **n = number of VoIP nodes **VT = VoIP Traffic

3.2 Proposed Algorithm for Increased Number of Node on G.711 Codec In this paper, we assume that RTP [22] over UDP is used for the VoIP transfer. When IP datagram is transferred over the 802.11 WLAN networks, the datagram is typically encapsulated by an IEEE 802.11. Accordingly, the VoIP packet size at the 802.11 MAC service Access Point become as following: 160-byte DATA + 12-byte RTP header + 8-byte UDP header + 20-byte IP header + 8-byte SNAP header = 208 bytes per VoIP packet 4. Simulation and Experimental Setup In this section, we evaluate the performance of the proposed model with related work. Our model will authenticate on NS-2 simulator [20] to show the utility of the method for the VoIP service over an infrastructure WLAN environment. We use the 802.11b for our simulations, and all the stations transmit packets at 11Mbps, which is the highest transmission rate of the 802.11b standard. 4.1 IEEE 802.11 Transmission Ranges of Access Point (AP) We will implement two different types of the traffic for our simulation in NS-2 namely, Voice and Data traffic flow. The voice traffic is modeled by a two-way constant bit rate (CBR) session according to G.711 codec technique. The data traffic application is modeled by a unidirectional File Transfer Protocol (FTP) flow with 1000-byte. The detail network topology for our simulation is shown in figure 4.1. Each mobile station involving with a VoIP session generates and receives only voice traffic over WLAN networks.

Figure 4.1Simulation Topology for VoIP over WLANs The other mobile stations either generate or receive only FTP packets, and each of them treats only one FTP flow. This topology can be often found in the real WLAN networks with mixed mode VoIP IP-based network. In the following section, the results and dissection will be defined in detail with graphs. Therefore the next section will be a comparison and evaluation of VoIP WLAN nodes. 5. Results and Discussion Relate research [20] indicates that the Mean Opinion Score (MOS) is a way which provides us, a measure for capable of being heard voice quality. Table-2 [21] shows the description of quality ranges from 1 to 5.

Table 2: Mean Opinion Score Ratings [21]

Score Quality Description of Quality 5 Excellent Imperceptible 4 Good Just perceptible, but not

annoying 3 Fair Perceptible and slightly

annoying 2 Poor Annoying but not objectionable 1 Bad Very annoying and

objectionable

In our results, we consider packet interval 50 (ms) over network. Figure 5.1 shows the Mean Opinion Score (MOS) of two well know codec techniques name are as G.711 and G.729, and where MOS ranging is based on 1 to 5. The MOS of the G.711 is 4.1 and number of nodes support over network 25 nodes. The MOS of the G.729 is 3.7 and number of nodes supports 35 nodes.

Table 2: MOS score compression methods [20]

Compression

Method

Bit Rate (kbps)

MOS Score

G.711 64 4.1

G.729 8 3.92

G.729 x 2 Encodings 8 3.27

Figure 5.1 Comparison of the Maximum Number of VoIP Nodes on Packet

Interval 50 (ms) If we have a look into table-2 then we have a better idea that our proposed algorithm have a batter MOS as compared with G.711 and G.729 codec. Figure 5.2 shows the MOS of G.711, G.729 and proposed algorithm. The proposed algorithm has 4.3 MOS until 15 nodes. After that, it decreased step by step until 4.1 over 35 nodes. We have noticed that proposed algorithm has better MOS as compared with G.711 and G.729 codec which shown in Table-3.

Table 3: Proposed Algorithm MOS score

Proposed Algorithm

No. Of Nodes Increased

MOS Score

15 4.3

…. ….

35 4.1

Figure 5.2 Comparison of the Maximum Number of VoIP Nodes on Packet

Interval 50 (ms) with Propose Algorithm Figure 5.3 shows the throughput of proposed algorithm over WLAN networks IEEE 802.11 Standards. The throughputs (Mbps) shows all types of traffic like data traffic, FTP traffic and VoIP traffic nodes, the time in (sec) during the NS-2 simulation. The graph shows the results of proposed algorithm over IP-based networks.

Figure 5.3 Comparison of the Maximum Throughput with other Traffic Flows

The proposed algorithm based on high data rate started from 3 (Mbps) until 100 (sec) and from 200 (sec) to until 600 (sec) it little decreased to 2.1 (Mbps). We have noticed that throughout proposed algorithms gave best results as compare with data traffic and FTP traffic over WLAN networks Fairness could be measured in fairness of time allocation, which is known as temporal fairness measurement an important parameter in the field of computer networks. Temporal fairness could be measured as throughput fairness, delay fairness. 2 2 1

Figure 5.4 shows the fairness index according to the mobility of mobile station. The fairness measured from 0 to 1 and above than 0 considered best. We also compared proposed algorithm with data traffic and FTP traffic over WLAN network. We noticed in the proposed algorithm start its fairness from 1 and step by step decreased until 0.96 fairness index. As we can see from graph the FTP and data traffic these have less fairness index as compared with proposed algorithms over WLAN network

Figure 5.4 Comparison of the Maximum Fairness Index with other Traffic Flows

6. Conclusion and Future Work In our research paper we have presented a proposed algorithm over WLAN network. At the same time we have studied Codec techniques, WLAN network for secure VoIP Solution. Based on our simulation results, we analyzed the G.711, G.729 codec, date traffic and FTP traffic with proposed algorithm. From all of the above fact and figure we verified/evaluated with simulation that our technique is better for enhanced number with get better Mean Opinion Score (MOS) over WLAN networks. If we talk about future work, then these techniques can be implemented on VoIP radio, IPTV and Video conferencing over WLAN networks.

ACKNOWLEDGMENT We would like to thanks Shaheed Zulfikar Ali Bhutto Institute of Science and Technology (SZABIST), Islamabad for supporting our MS research work.

REFERENCES [1]V. Soares, P. Neves, and J. Rodrigues, “Past, Present and

Future of IP Telephony,” International Conference on Communication Theory, Reliability, and Quality of Service, Bucharest, pp. 19–24. 2008.

[2]R. Beuran “VoIP over Wireless LAN Survey,” Internet

Research Center Japan Advanced Institute of Science and Technology (JAIST,) Research report. Asahidai, Nomi, Ishikawa, Japan, pp. 1-40. 2006.

[3]V. Mockapetris, “Telephony's next act”, in IEEE

Spectrum, vol.43, Issue 4, pp. 1-5, 29 April 2006 [4]M. ALAkhras, “Quality of Media Traffic over Lossy

Internet Protocol Networks: Measurement and Improvement”, Software Technology Research Laboratory, De Montfort University, United Kingdom, PhD thesis, 2007

[5]P. Dely “Adaptive Aggregation of Voice over IP in Wireless Mesh Network”, Department of Computer Science, Karlstad University, Master’s Project, 28 Jun 2007

[6]Q. Ni, and T. Turletti “QoS Support for IEEE 802.11

Wireless LAN”, the French Ministry of Industry in the Context of the National Project RNRT-VTHD, , 28 Jun 2007

[7] L. Sun “Speech Quality Prediction for Voice over

Internet Protocol Networks”, School of Computing, Communications and Electronics, Faculty of Technology University of Plymouth, United Kingdom, PhD thesis, January 2004

[8] A. Markopoulou, F. Tobagi, and M. Karam,

“Assessing the Quality of Voice Communications Over Internet Backbones,” IEEE/ACM Transactions on Networking, Stanford, CA 94305 USA, Vol. 11, No. 5, pp. 747-760, October. 2003.

[9] Q. Cao, T. Li, Tianji and D. Leith “Achieving

fairness in Lossy 802.11e wireless multi-hop Mesh networks,” Third IEEE International Workshop on Enabling Technologies and Standards for Wireless Mesh Networking MESH, Macau SAR, P.R. China, pp. 1-7, 10, November. 2009.

[10]P. Dini, O. Font-Bach, and J. Mangues-Bafalluy,

“Experimental analysis of VoIP call quality support in IEEE 802.11 DCF,” Communication Systems, Networks and Digital Signal Processing, 2008. CNSDSP2008. 6th International Symposium, pp. 443 – 44729, August. 2008.

[11]T. Li, Q. Ni, T. Turletti and Y. Xiao “Performance analysis of the IEEE 802.11e block ACK scheme in a noisy channel”, Broadband Networks, 2005. BroadNets 2005. 2nd International Conference. Hamilton Inst. Ireland, Vol 1, pp. 511-517, 13, February. 2006.

[12]Q. Ni, T. Li, T. Turletti, and Y. Xiao. "Saturation

Throughput Analysis of Error-Prone 802.11 Wireless Networks," Wiley Journal of Wireless Communications and Mobile Computing (JWCMC), INRIA, France, Vol. 5, Issue 8, pp. 945-956. December. 2005.

[13]S. Garg and M. Kappes, “Admission control for VoIP

traffic in IEEE 802.11 networks,” in IEEE GLOBECOM, pp. 3514 – 3518. 2003.

[14]D. G. et al, “Improving wlan voip capacity through

service differentiation,” IEEE Trans. on Vehicular Technology, vol. 57, no. 1, pp. 465 – 474, 2008.

[15]S. Garg and M. Kappes, “Can I add a VoIP call,” in

IEEE ICC, 2003, pp. 779 – 783. 2003. [16]D. Hole and F. Tobagi, “Capacity of an IEEE 802.11b

Wireless LAN Supporting VoIP,” in IEEE ICC, 2004. [17]Y. Zhang and B.-H. Soong, “Performance of mobile

networks with wireless channel unreliability and resource insufficiency,” IEEE Trans. on Wireless Communications, vol. 5, pp. 990 – 995, 2006. unbalanced traffic,” IEEE Trans. on Vehicular Technology, vol. 55, no. 3, pp. 752–761, 2006.

[18]Daniel Collins, Carrier Grade Voice over IP, 2nd

Ed.,McGraw-Hill, September 2002. [19]“The Network Simulator – ns-2,” [20]Holly Xiao and peter Zarreela “ Qualit Efects of

Wireless VOIP Using Security Solutions”, in IEEE, 2004. [21]]http://www.cisco.com/en/US/docs/ios/12_4/ip_sla/conf

iguration/guide/hsvoipj.pdf [22]Florian Evers and Yevgeniy Yeryomin “Handover-

aware SIP-based VoIP provided by a Roaming-Enabled Architecture (REACH)” in IEEE, 2010.