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L.R.He, B.M.G. Cheetham Mobile Systems Architecture Group, Department of Computer Science, University of Manchester, Oxford Rd, M13 9PL, U.K. •Application of real time multimedia communications to distance learning over IP networks. . • Current RTP systems are based on comparing frame loss rate, as reported by RTCP, with thresholds. • We presents a flow/congestion control mechanism for adapting to congestion using measurements of time delay, jitter & speech packet loss rate. •We propose a dynamic assignment of priority to the speech, image & data as appropriate to distance learning activity. • To improve the flow/congestion control mechanism based on RTP and RTCP •To allow dynamic priority assignment determining how the available capacity is divided between speech / image / data. Distance learning is applied in these fields: * Providing open learning environments * Offering more information for traditional teaching * Providing continuing education after graduation * Developing academic co-operation Three types of distance education -Paper based learning -TV and radio based learning -Internet based learning Database (Course Information, Content, Assignments, Problems and Solutions, Audio-video Information, Examinations, Announcements, Student Records) Web server Internet Browser (student 1) Browser (student N) Browser (teacher 1) Browser (teacher M) Real Time Multimedia has the following advantages for Distance Learning : * Liveliness * Efficiency * Interactivity Most of the real-time applications are based on the end-to-end real time transport protocol (RTP) and real time control protocol (RTCP). RTCP sender report Camera & Receiver 1 Data microphone Receiver 2 Check result Internet RTCP Receiver Report Real time transport protocol is used for end-to end data transfer. RTP transports real-time media data along with synchronization information over a datagram protocol. G.711,G.722,G.723.1, G.728,G.729 H.261, H.263 RTP User Datagram Protocol Network Layer Link Layer Physical Layer RTCP Real-Time control Protocol has a feedback function. Feedback from the receivers is necessary for diagnosing distribution faults. RTCP produces sender and receiver reports. • Stream statistics • Packet counts • Sender identification • Quality of service • Lost packets Receiver classifies loss error Loss rate(%) Network Bit-rate state 100% Packet loss --- Decrease Loaded --- Hold Unloaded --- Increase c probability n Congested u 0 Problems lie in dynamics of bit-rate adaptation Priority weighting “congested” high priority weighting or low priority weighting “congested” Packet loss probability Speech packet unusable probability “loaded” Average jitter Management “unloaded” “unloaded” 1 Determination of the network states As before, the network will be defined to have three different states, “congested”, “loaded”, and “unloaded”. If 1 λn λc or (λn λu and λsn > λsc), the network state is “congested”. λsc is a threshold. If λn < λu, then the network is “unloaded”. If (λn λu and λsn λsc), the network is “loaded”. The probability of a received speech packet being unusable λsn as follows: nth control period T (400Pd) ms Pd is the processing delay. 2 Management • The priority of a given packet • If the network state is “congested”, H = H – H. • If the network state is “unloaded”, H = H + H. • If the network state is “loaded”, the sender hold the image and other packets transmission rate, checks the average jitter, Jn, to decide how to adjust the speech packet transmission rate. Average Jitter Speech transmission rate Decreasing Hs + Constant Hs Increasing Hs * Hsmin Hs Hsmax Table: Determination of the requested sending rate of speech packet Transmission rate (kb/s) Fig.5 Speech packet transm ission rate 100 50 40 30 20 40 50 80 60 70 Tim e (sec) 60 40 Previ ous method New method 20 0 0 100 200 300 Tim e (second) Fig.6 Speech transm ission rate Packet loss probability Transmission rate (kb/s) Loss probability (%) Fig.4 Loss probability 60 60 55 50 45 245 250 255 260 265 270 275 Tim e (sec) Previous method New method • A flow/congestion control mechanism based on RTP and RTCP has been investigated. • A more appropriate variation in transmission rate is achieved by the new method. • It can be appropriately applied to distance learning over internet.