Abstract
Problem statement: In Voice Over IP (VOIP) network, when more calls are admitted to the network, more voice packet traffic is created. Since bandwidth is always limited, this may result network congestion and/or may affect voice quality. Thus, we needed a mechanism for improving the Quality of Service (QoS) by controlling VOIP calls admission. Approach: Given a specified bandwidth and a constant background data rate, we attempted to explore the effect of Open Window and Leaky Bucket admission schemes on VoIP calls quality. These policy-based admission controls were simulated using NS-2 Simulator. The inter-arrival time distribution for the network background data traffic was assumed to be deterministic with a Constant Bit Rate (CBR). Voice packets traffic inter-arrival time is assumed to have an exponential distribution. Each voice call has a rate of 64 kbps for duration of 120 min. Results: Various performance measures of VoIP calls and packet traffic were evaluated including: packet loss, packet drop rate, delay, jitter and call rejection rate. Performance results of the experiment are summarized in a power ratio index which presented the impact of a collection of performance parameters on VoIP service quality. Conclusion: Implementing a policy based admission scheme on VoIP network will improve its QoS and the degree of improvement depends on the network setting parameters. If threshold rate for call admission is set above network ceiling bandwidth, leaky bucket will result a higher and unacceptable jitter. Overall, leaky bucket scheme was considered inferior when compared to open window for improving QoS of VoIP.
Highlights
Voice and video transmission over telecommunication networks requires specific performance quality
In the no-policy experiment, the average power ratio is found equal to 0.029. This result indicates that, For packet loss, we found that leaky bucket reduces the loss to virtually zero, while open window still have a packet loss of 0.45% at 3 Mbps ceiling bandwidth. (Fig. 6)
No significant difference exist at any with no admission control for packets arriving at the threshold rate value
Summary
Voice and video transmission over telecommunication networks requires specific performance quality. If such quality is not maintained, the receiving end will suffer-e.g., the received video freezes or there will be unacceptable delay in voice. Transmitting voice over IP networks will have the same challenge With this in mind, a call admission controller in VoIP networks is needed to maintain voice quality over a limited bandwidth link. Several admission mechanisms are available for Call Admission Control (CAC) over the Internet Example of these are IntServ architecture which uses RSVP signaling protocol for reserving resources in a router[1] and EMBAC protocol which use probes transmission to estimates networks state from sender to receiver[2-4]. CAC mechanisms are considered for wireless networks and in IEEE 802.11e standard environment to enhance its performance[4,7]
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