Real-time Transport Protocol Control Protocol Research Articles

Real Time Communication over Modified UDP Protocol

The objective of this paper is to provide a uniform means of transmitting data (audio, video) to real time constraints over multicast or unicast network services and providing much enhanced experience of the Real-time communication over the existing systems. In this paper we will explore usage of the dominating technologies in the real time systems and shows how to design and develop communications within such systems. In particular, it proposes reliable protocol over UDP transport layer that gains less than one millisecond average round trip time (RTT) and processes dozens of thousands of messages every second at each and every port in network. This paper presents the current internet protocols: Real-time Transport Protocol (RTP), RTP Control Protocol (RTCP) and Real-time Streaming Protocol (RTSP). Together these protocols may provide controlled delivery of multimedia traffic over the Internet. The feedback mechanisms for the quality of service (QoS) monitoring, such as delay, jitter and packet loss calculations, are described in detail.

Intermedia synchronization for videoconference over IP

This paper deals with the problem of audio and video synchronization issues for real-time audio-visual communication over IP-based networks. Starting from the real-time transport protocol (RTP) specifications (Schulzrinne, 1995), it provides an accurate description on how to recover a reliable absolute time reference for audio and video signals from header information in RTP and RTP control protocol packets. Such temporal informations allow to synchronize both media within acceptable perceptual bounds for reconstruction at any receiver end, in a possibly multi-point videoconference. This may occur independently of the fact that all (audio/video) packets reach destination, or that multiple replications of such packets arrive at destination. The proposed solution does not take into account the possible different delays that may be introduced between the different communication points, thus resulting eventually in potential jitters in the reconstruction of the audio-visual streams between different receivers. Each receiver handles its reconstruction independently of any transmission/processing delay. In order to ensure a better quality of the reconstructed material, priority is given to audio information. If the audio stream anticipate the video stream, the receiver simply discards video packets. Conversely, when video is ahead of audio information, the video decoding stage is interrupted till audio information arrives. Experimental simulations over a LAN have demonstrated the validity of the proposed approach.