Analysis of Traffic Characteristics of Skype Video Calls Over the Internet

Abstract

Skype is an important application of real-time systems (RTS). It uses Transmission Control Protocol (TCP) for connection establishment and User Datagram Protocol (UDP) port for transfer of audio and video data. In spite of the popularity of Skype, relatively little is known about its traffic characteristics. In this paper, the sender is sending the audio visual data using UDP port at a constant bit rate (CBR). The destination receives the data and checks with its buffer threshold values. The minimum and maximum threshold values are fixed based on the receiver buffer. When the sender data is higher than the threshold value, the destination asks source to reduce the flow rate by sending an explicit control packet. When the sender rate is lower than the threshold value, the destination can ask the source to increase the sending rate. The introduction of feedback system using a control message overcomes the congestion at the receiver, minimizes the data loss, increases optimal utilization of resources, and enhances the quality of expectation. In Skype application, an ordinary node acts one time as a sender and other time as a receiver. The proposed technique is to be built at both the end nodes. Important quality parameters such as packet loss due to congestion, one-way packet delay, effect of queuing delay on sender performance of feedback are analyzed using graphs and statistical data. Mathematical models are used to analyze the Skype performance. MATLAB software is used to simulate the system and for model authentication.