Proactive QoS Enhancement Technique for Efficient VoIP Performance over Wireless LAN and Cognitive Radio Network

Abstract

VoIP over Wireless LANs is greatly affected bypath-loss, RF interference and other sources of signalattenuation in addition to network congestion. The primaryfactors involved in effective real-time communication,namely delay and loss, must be within certain controlledlimits in such a scenario. Wireless LAN Access Points must,therefore, act in tune with the codecs to ensure high qualityof the ongoing VoIP sessions. Further deployment of VoIPin opportunistic communication medium like cognitiveradio network requires thorough analysis of the factorsinvolved in design and implementation of such networksalong with modifications in VoIP parameters. In this paper,the various codec parameters are analyzed with respect towireless LAN access points. Thereafter, an optimizationalgorithm is developed based on proactive strategy to keeploss and latency within tolerable limits. The proposedalgorithm is hence implemented in the SIP enabled test-bedto provide optimum Voice over IP (VoIP) performance. Theproactive strategy is further applied to study VoIPperformance over cognitive radio networks. A simulationmodel is developed for VoIP transmission in cognitive radionetworks with two scenarios where the primary and thesecondary user involve in VoIP transmission respectively.For each scenario, extensive analysis of both VoIP and CRNparameters is performed and an algorithm is proposedbased on codecs and active queues to ensure highthroughput of VoIP traffic with enhanced QoS. Resultsverify significant performance improvement in VoIP callquality under both wireless LAN and cognitive radionetwork environment. Simulation and measured resultshave also been included

Secondary VoIP Capacity in Opportunistic Spectrum Access Networks with Friendly Scheduling

“…An appropriate solution would be a combination of the two approaches, where the codec configuration of the virtual VoIP call used in the VN algorithm is set to G711 with 20 ms, while on the secondary side, the adaptive VoIP codec scheme would track the spectrum holes. In the recent literature, online scheduling algorithms were proposed to adapt the VoIP codec used in the secondary network based on the primary user activity [24] [25]. Reference [26] considers a system which controls VoIP call distortion by varying the VoIP source coding rate and other parameters in a downlink wireless system.…”