RTP/RTCP protocol suite provides the means for sending packetized voice by introducing time stamps and sequence numbers in packet headers. Playout buffering is needed to re-synchronize the received voice stream. In this paper, a new adaptive playout delay adjustment algorithm is introduced. A major problem, especially on low-bandwidth links, with Voice over IP (VoIP) packets is that they includ...

This paper addresses the problem of rate and congestion control for video transmission in large multicast groups. A classification of the receivers is performed according to a predefined similarity measure (e.g. packet loss rate observed or TCP-friendly bandwidth share). The RTCP receiver reports are then filtered accordingly. The classification of receivers done by feedback aggregation agents ...

89 Abstract—Telemedicine system means a remote support system among 4 remote telemedicine services of u-Health. A currently available service in Korea is the one checking and maintaining patient's status and supporting in emergency through video communication between an ambulance transferring patients and hospital under HSDPA and WiBro environment with coding of MPEG-2 and MPEG-4. With that, th...

In this paper, we describe and evaluate a mechanism for ltering RTCP receiver reports sent from receivers to the whole session. In large multi-cast groups, such feedbacks could induce network congestion, receivers overload and feedback delay. The proposed ltering mechanism provides a clas-siication of receivers according to a predeened similarity measure (e.g. packet loss rate observed or TCP-f...

In this paper, we outline an architecture and an implementation which enables the real-time delivery of continuous media within the WWW, monitors QoS of data transmission, and conveys minimal session control information to all participants in an audio or a video session. We describe our current implementation of the Real-time Transport Protocol (RTP) and its associated control protocol RTCP emb...

The present document exposes the development of a prototype of translation system as a Thesis Project. It consists basically on the capture of a flow of voice from the emitter, integrating advanced technologies of voice recognition, instantaneous translation and communication over the internet protocol RTP/RTCP (Real time Transport Protocol) to send information in real-time to the receiver. Thi...

To facilitate real-time voice communication through the Internet, forward error correction (FEC) and multiple description coding (MDC) can be used as low-delay packet-loss recovery techniques. We use both a Gilbert channel model and data obtained from real IP connections to compare the rate-distortion performance of different variants of FEC and MDC. Using identical overall rates with stringent...

This paper describes the design of a scaleable interactive continuous media server that is realized using a distributed memory parallel computer architectures. We present a scheme for data layout, admission control and communication scheduling that provides inherent Quality of Service support for the delivery of continuous media streams from the storage devices to the external communication int...

Status of This Memo This memo provides information for the Internet community. It does not specify an Internet standard of any kind. Distribution of this memo is unlimited. IESG Note The IESG has concerns about vendor code points allocation in this small namespace and might not approve similar documents in the future. Abstract This document describes an RTCP XR report block, which reports packe...

This document is an Internet-Draft. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet-Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropr...