Kim Dong Kyoo, Kang Min Gyu, Hwang Seung Koo

The Transactions of the Korea Information Processing Society (1994 ~ 2000), Vol. 5, No. 12, pp. 3047-3062, Dec. 1998
10.3745/KIPSTE.1998.5.12.3047,   PDF Download:

Abstract

This paper describes the design and the implementation of the Realtime Transport Protocol(RTP)/Realtime Control Protocol(RTCP) (RFC 1889, 1890) that is used to trnsmit the audio/video data to any destination and to feedback the Quality of Service(QoS) information of the received media data to the sender, in the teleconferencing systems proposed by ITU-T. These protocols are implemented with multi-thread technique and run on top of UDP/IP-Multicast through the socket interfeace as the underlying protocol. The upper layer is implemented such that it can be accessed by the H.245 conference control protocol. The RTP packetizes the digitized audio/video data from the encoder into a fixed format, and multicast to the participants. The RTCP monitors RTP packets and extracts the QoS values from it such as round-trip delay, jitter and packet loss to form RTCP packets and non-periodically sends them to the sender site. In this paper, we also describe the study of measurement and analysis for QoS factors that observed on performing teleconferencing system over Internet. The results from this experiment is indicate that RTT and Jitter value are acceptable even network load is high. However, it appears that packet loss rate is high in daytime and most losses periods have length one or two.

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