ملخص
Abstract
Voice over IP (VoIP) uses the Internet Protocol (IP) to transmit voice as packets over an IP network, like the Internet, Intranets and Local Area Networks (LAN). Here the voice signal is digitized, compressed and converted to IP packets and then transmitted over the IP network. However, at the receiving end, some packets may be missed in its way due to network congestion. This packets loss degrades the quality of speech at the receiving end of a voice transmission system in the IP network. Since the voice transmission is a real time process, the receiver cannot request for retransmission of the missing packets. High speed networks provide real time variable bit rate service with packet loss requirements. The burstiness of the correlated traffic makes dynamic buffer management highly desirable to satisfy the Quality of Service (QoS) requirements. This thesis presents an algorithm to improve and optimize the Adaptive Buffer Allocation Scheme to deal with input traffic based on loss of consecutive packets in data streams and buffer occupancy levels. Buffer is designed to allow the input traffic to be partitioned into different priority classes, and based on the input traffic behavior it controls the threshold dynamically. This scheme allows input packets to enter into buffer if its occupancy level is less than the threshold value for priority of that packet. The threshold is dynamically varied in runtime based on packet loss behavior. The performance evaluation is carried out using simulation and is carried out for two and multiple priority classes of the input traffic "real time and non real time classes". The simulation results show that the Modified Adaptive Partial Buffer Sharing (ADPBS) has better performance than Adaptive Partial Buffer Sharing under the same traffic conditions.