الفهرس 
INTRODUCTIONOnly 14 pages are availabe for public view
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Abstract
Converged IP networks carry data, voice and video on the same infrastructure, the integration of all types of traffic onto a single IP network has several advantages as reducing cost and increasing mobility and functionality, but real time voice traffic, VoIP, may suffer from degraded voice quality due to changing network characteristics. The quality of VoIP communication relies significantly on VoIP equipment and on the IP network that transports the voice packets because this network does not usually guarantee the available bandwidth, delay and packet loss that are critical for real time voice traffic. In order to ensure these strict requirements are met, the underlyin network must deploy various schemes to ensure resource availability. The objective of this thesis is to optimize the quality of voice perceived by end users and maintain an efficient utilization of the available network resources by developing quality management algorithm that are constantly monitor the network in real time and dynamically selects a more appropriate VoIP codec according to network condition based on the feedback information obtained from Real Time Control Protocol (RTCP) packets that contain information about packet loss and delay. When the quality of VoIP call is out of the accepted range then the algorithm triggers a mechanism to change the used codec to a newer codec suitable for the current network condition to maintain the VoIP call with acceptable quality and to optimize the VoIP quality of service. The proposed mechanism ensures the voice call continuity while changing used codec. Simulations demonstrate that the system provides better average voice quality and more efficient network recourses utilization than traditional VoIP communication system.