An Adaptive Sender-Based VoIP Loss-Recovery

Technique

Sami S. Alwakeel*

Computer Engineering Department

College of Computer and Information Sciences, KSU

Riyadh, Saudi Arabia

swakeel@ksu.edu.sa

Adel AlBahouth

Computer Engineering Department

College of Computer and Information Sciences, KSU

Riyadh, Saudi Arabia

Albahouth@gmail.com

Abstract- Many techniques were proposed for improving quality

of service (QoS) in the face of packet loss in Voice over Internet

Protocol (VoIP) networks. The Sender-Based loss recovery

techniques is commonly used to improve QoS when packet loss

occurs by retransmitting data or by transmitting additional data.

This research introduces a newly developed adaptive sender-

based loss recovery technique. A simulation study is made to

evaluate the techniques performance and its impact on QoS for

calls delivered through the network. The performance results

showed that the proposed adaptive technique achieved a lower

average delay , delay jitter and higher throughput when

compared to traditional Redundant Technique.

Keywords—VoIP QoS Control; sender-based loss recovery

technique for VoIP; QoS in IP networks; Adaptive techniques for

VoIP networks.

I. INTRODUCTION

VoIP Voice over Internet Protocol (VoIP) is a telephony technology that commonly uses the real-time transport protocol (RTP) to transport voice packets over a IP based network. [1] RTP runs on top of the user datagram protocol (UDP), and thus it is an unreliable delivery protocol. When a packet loss occurs, the quality of the audio at the receiving endpoint degrades because the receiving endpoint does not have voice data for regenerating the lost segment of the audio [2]. Researchers proposed many techniques for improving quality of service (QoS) in the face of packet loss [2]-[6].Some of these techniques employ receiver-based packet loss concealment (PLC) approaches and others employ sender-based loss-recovery techniques (SBLR), where the sender assumes an active role to help the receiver recover lost data or improve QoS when packet loss occurs. Most SBLR mechanisms work by retransmitting voice data or by transmitting additional data. In what follow , we briefly describe various sender-based loss-recovery techniques currently used [2]-[7].

Interleaving: This technique adds part of the same voice signal segment in different packets thus spreading the impact of loss over longer time period. Forward Error Correction (FEC) works by transmitting redundant packets for error correction. Redundant Data Transmission (RDT) works by transmitting audio data more than once. This technique includes previously transmitted audio data along with new audio data in a single IP packet. Duplicate Packet technique

transmits the redundant data in separate IP packets and thereby increases bandwidth consumption by requiring additional data and header overhead.

Redundant and duplicate data transmission is shown to produce the best audio quality among other techniques in different packet loss situations (single packet, burst of 2,3… n packets) [2]. These approaches however, add overhead in term of more bandwidth requirement, CPU processing and packet transmission delay and may increase network congestion which lead to higher packet loss and may drops the voice call.

In this paper we present a newly developed modified SBLR technique. The proposed technique works toward improvement of QoS in VoIP system without adding excessive overhead in term of more network bandwidth requirement or processing/transmission delay. The proposed technique works by setting a threshold value adaptively set based on the average network loss rate seen by receiver. This threshold is used to control the volume of the transmitted redundant audio data. The rest of the paper is organized as follow: Section 2 describes various SBLR techniques operation parameters currently used in VoIP networks. In section 3 we discuss the performance measures of the technique. The design of the proposed adaptive Sender-Based VoIP Loss-recovery Technique is presented in section 4. Simulation framework that is used for evaluating the performance of techniques is described in section 5. Finally we present the main Simulation Result in section 6.

II. SBLR TECHNIQUES OPERATION FACTORS

The SBLR approach commonly is designed using several operation factors. These include:

Degree of Redundancy :-This represents the amount of previously transmitted audio data to be retransmitted along with new audio data in a single IP packet.

P-persistence parameter :- Extensions of SBLR

conventional redundant class works by randomly

transmitting redundant audio with the redundancy rate

depends on a pre-determined P-persistence parameter

to improve QoS in VoIP system [8].

Threshold Value:-The SBLR technique redundancy may depend on fixed thresholds. When new packet

978-1-4673-4410-4/12/$31.00 ©2012 IEEE

arrives the loss rate is compared to threshold if the loss is greater than the threshold, redundant data and new data will be included in the IP packet. Otherwise only new data will be sent.

Network QoS Factors:-The path taken by a VoIP

packet traveling across the network depends on a large

number of factors, including routing protocols and per

network buffering policies. These factors may strongly

impact the quality of VoIP. SBLR technique is

activated based on report about the actual network QoS

measures. These include: packet Loss Rate, Packet

Discard Rate -because of jitter and/or large delay- , the

distribution of lost and discarded packets , packets

Round-Trip Delay corresponding to the packet path

delay and End System Delay, which represents the

delay that the VoIP endpoint adds (because of

encoding, decoding and the jitter smoothing buffer).

III. SBLR TECHNIQUES PERFORMANCE MEASURES

In our research, we used the following measures to evaluate the performance of our proposed sender-based loss recovery scheme

Mean Opinion Score (MOS).

Throughput.

Delay.

Jitter.

Cost of Technique

Power Ratio.

MOS is a common benchmark introduced by ITU recommendation G.107 for measurement of the subjective quality of human speech, represented as a rating index with a maximum value equals to 5. MOS is derived by taking the average of numerical scores given by juries to rate quality and using it as a quantitative indicator of system performance [9]. SBLR techniques generate more data bits than plain technique. We define The measure “Cost of Technique “ to compare the number of bits generated by specific technique to the number of bits generated by plain technique.

Power Ratio P: This measure helps in a collective analysis of different measures we used in this study. It is defined as a sum of functions of scaled performance measures including normalized throughput, normalized MOS, normalized delay, normalized jitter, cost of technique.

IV. ADAPTIVE REDUNDANT SBLR TECHNIQUE

DESIGN

In this section we describe the proposed adaptive Sender-Based VoIP Loss-recovery Technique [10]. This technique works by incorporating an adaptive threshold for redundant transmission of voice audio data. The redundancy depends on two operation factors which are the actual current VoIP network packet loss and a threshold variable parameter set equal to the average network loss rate seen by the receiver. If network current loss rate is greater than the threshold, previous

Fig 1. Adaptive Threshold Redundant Delivery Flow

data and new data will be included in single IP packet. Otherwise only new data will be sent. The threshold average is updated whenever a new data segment arrives to the new loss average based on the current measured report of the loss rate seen by the receiver. Figure 1 shows the flow of Adaptive Threshold Redundant Delivery technique.

In mathematical notation;

Let threshold T = average loss. Let L= current loss rate. For

every packet

if L >T send packet j in frames i and i+1, otherwise send

packet j in frame i

Update T to a new Average based on current and previous

loss rates Based on the above, the technique requires the knowledge of network loss rate to control sending redundant data. However, this requirement will not introduce extra overhead traffic to network because RTP Control Protocol, commonly used for VoIP calls packets transport, defines and reports a set of performance metrics such as Packet loss and discard rate, delay, and Call Quality (MOS)[11]. Thus, the proposed technique can be implemented with no additional changes to existing VoIP System.

V. ADAPTIVE REDUNDANT SBLR MODELING AND

SIMULATION

In our study we used Network simulator 2 (NS2) to build a VoIP system simulation software of the proposed adaptive SBLR technique [12]. VoIP extension which is an enhancement to NS2 is implemented to allow a reliable VoIP user-level performance analysis to be carried out for the proposed techniques. Figure 2. shows a diagram of simulated VoIP network model as shown in the diagram, the network

model framework consist of following components:- VoIP Source ,VoIP Sender gateway , Network Node , VoIP Receiver gateway and receiver decoder. The sender gateway incorporates: SBLR processor, VoIP Header and system Agent. The source model is implemented with call duration having exponential distribution with frequent transitions between active and inactive states (ON/OFF patterns). The length of (ON, Off) states is exponentially distributed. The voice source generates talk-spurts and passes them to VoIP sender. VoIP sender generates voice data periodically, and passed them to VoIP encoder which in turn interacts with SBLR processor or VoIP Header to generate the outgoing frame in sender gateway. According to network operational state VoIP SBLR Processor – technique main component – may replicate voice data and pass them to VoIP header. VoIP header adds RTP/UDP/IP headers and pass it network agent then to network nodes. Network node injects voice frame into network. When VoIP reaches the Destination node gateway, node will pass frame to VoIP receiver where VoIP frame is decoded and the voice data inserted into play-out buffer.

Fig 2. VoIP Network Simulation Model.

Sender Gateway

Recievr

VOIP Source

VOIP EncoderVOIP SBLR

processor

VOIP Packet

Header

Network

Transmission

System

Decoder

Talkspurt()

Sendmsg()

Process_data()

Recv()

Recv()

Gateway

Recv()

RTCP Feedback

Agent 2

Agent N

Agent M

Agent 2

Fig 3. VoIP Simulation Component.

VI. ADAPTIVE REDUNDANT SBLR TECHNIQUE

PERFORMANCE

Here we will present all result obtained from simulation

A. Mean Opinion Score

Figure 4 shows the results of MOS measure for Redundant and adaptive Redundant algorithms versus the loss rate. The MOS value decreases as loss rate increases. For Redundant , the affect of loss rate is reduced because of the increase in the number of recovered voice frame. As the redundancy decrease the MOS value decrease.

B. Throughput

Figure 5 shows the results of throughput measure for Redundant and Adaptive Redundant algorithms versus the loss rate. The Throughput value decreases as loss rate increases. As the redundancy decrease the throughput value increase. The adaptive redundant technique achieved better result compared to Redundant technique.

C. Delay

Figure 6 shows the results of Delay measure for for Redundant and Adaptive Redundant algorithms versus the loss rate. The Delay values increase as loss rate increases. The Redundant technique scored higher delay (worst) because the delay of recovered voice frame will be double of normal frame delay. Adaptive technique achieved better delay (lower value) than Redundant Technique.

Fig 4. MOS Values for Adaptive Threshold Redundant SBLR

Fig 5 Throughput Values for adaptive Redundant techniques.

Fig. 6 Delay Values for adaptive Redundant SBLR

Fig 7 Jitter Values for Adaptive Redundant SBLR.

Fig 8 Power Values for Adaptive Redundant SBLR

D. Jitter

Figure 7 shows the results of Jitter measure for different Redundant algorithms versus loss rate. The Jitter values increase as loss rate increases. The Redundant technique scored

higher Jitter (worst) as the delay of recovered voice frame is doubled relative to normal frame jitter.

E. Power Ratio

Figure 8 shows the results of Power measure for redundant algorithms as a function of loss rate. The Power values decrease as loss rate increases. The Adaptive techniques scored higher values for loss rate 0 – 7%. For higher loss rate value, redundant technique scored higher values.

VII. CONCLUSION AND SUMMARY

In this paper we presented an extension of SBLR passive

redundant class that works by transmitting redundant audio

data with the redundancy rate depending on an adaptive

threshold parameter. The threshold value is set equal to the

average loss rate of the network. Whenever a voice segment is

to be sent , the threshold value is compared to current measured

network loss rate - seen by receiver- to control the data

redundancy rate. The performance of this technique was

evaluated for various QoS measures using a simulation system

developed by NS2 software. The proposed technique achieved

better voice quality compared to Plain techniques as reflected

by the MOS measure. Our techniques also achieved better

throughput , delay , and jitter performance than conventional

redundant techniques. Overall , using the power ratio measure

for the adaptive redundant based technique, we can conclude

that the proposed redundant techniques achieves a better

performance than conventional passive redundant technique

whenever the loss rate is less than or equal to 6%.

VIII. REFERENCES

[1] Uyless Black, “Voice over IP”, Prentice Hall, Second Edition,2002 .

[2] Teck-Kuen Chua and David C. Pheanis “QoS Evaluation of Sender-Based Loss-Recovery Techniques for VoIP”, IEEE Network Vol. 20, Number 6, November 2006.

[3] H. Sanneck, N. Le, A. Wolisz and G. Carle , "Intra-Flow Loss Recovery and Control for VoIP" , in Proceedings ACM Multimedia 2001, Ottawa, ON, September 2001

[4] Chinmay Padhye, Kenneth J, Christensen, Wilfrido Moreno, “A New Adaptive FEC Loss Control Algorithm for Voice Over IP Application” IEEE Computing, and Communications Conference, IPCCC '00. pp, 307 –313 , 2000

[5] Philippe Gournay, Francois Rousseau and Roch Lefebvre, “Improved packet loss recovery using late frames for prediction-based speech coders” IEEE International Conference on Acoustics, Speech, and Signal Processing, Proceedings. (ICASSP '03). Vol. 1 , pp 108,111, 2003.

[6] Colin Perkins, Orion Hodson, and Vicky Hardman “A Survey of Packet Loss Recovery Techniques for Streaming Audio” , IEEE network Vol. 12 . issue 5. pp , 40-48. 1998.

[7] Angelos D. Keromytis “A Survey of Voice over IP” Proceeding of Information Systems Security, 5th International Conference, ICISS 2009, pp. 1-17 , 2009

[8] Sami Al-Wakeel, Adel AlBahouth , "Performance Study of A Random P-Persistence Sender-Based VoIP Loss-recovery Technique " IEEE International Systems Conference 2012 SYSCON'12, March 19-23, 2012 , Vancouver, British Columbia, Canada.

[9] ITU-T Recommendation G.107, “The E-Model, a computational model for use in transmission planning”, December 1998.

[10] Adel AlBahouth, Development and Performance Study of Sender-Based Loss Recovery Techniques for VoIP systems, M.Sc Thesis, Computer

Engineering department, College of Computer and information science King Saud University, Riyadh , 2010

[11] IETF RFC3611 “RTP Control Protocol Extended Reports (RTCP XR)”

, 2003.

[12] A. Bacioccola, C. Cicconetti, and G. Stea, “User-level performance evaluation of VoIP using ns-2,” in ValueTools ’07: Proceedings of the 2nd international conference on Performance evaluation methodologies and tools. ICST, Brussels, Belgium, Belgium: ICST (Institute for Computer Sciences, Social-Informatics and Telecommunications Engineering), pp. 1–10. , 2007.