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Improving Communication Quality with Reed Solomon Code in Internet Voice Broadcasting System Shingo Kashima Kyushu University, Japan Asia-Pacific Advanced.

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Presentation on theme: "Improving Communication Quality with Reed Solomon Code in Internet Voice Broadcasting System Shingo Kashima Kyushu University, Japan Asia-Pacific Advanced."— Presentation transcript:

1 Improving Communication Quality with Reed Solomon Code in Internet Voice Broadcasting System Shingo Kashima Kyushu University, Japan Asia-Pacific Advanced Network 2003 January 21-24, 2003 Fukuoka, Japan

2 2 Plan of Presentation Background (Existing Issue) Objective Icecast Solution of Issue FEC and Reed Solomon Code Proposal of System Evaluation Future Work Summary

3 3 Background [1 of 2] Music Delivery in Real Time –1 to 1 communication using client-server model

4 4 Background [2 of 2] Issues in the Current Model I.A heavy load is applied at the network between client and server. II.The number of clients is limited by bandwidth of network between client and server. III.Sound is interrupted or noise occurs. Solution Modify existing applications Expensive

5 5 Objective Improve the communication quality of a large scale real time voice broadcasting on the Internet without modifying the existing applications.

6 6 Icecast Audio streaming server Developed under the GNU General Public License Support MP3, HTTP / TCP Support many client applications –ex. WindowsMediaPlayer, Winamp, XMMS, etc Relay function (described later)

7 7 Issues in the Current Model I.A heavy load is applied at the network between client and server. II.The number of clients is limited by bandwidth of network between client and server. III.Sound is interrupted or noise occurs.

8 8 Issues in the Current Model I.A heavy load is applied at the network between client and server. II.The number of clients is limited by bandwidth of network between client and server. III.Sound is interrupted or noise occurs.

9 9 Solution of Issues I, II Distributed Delivery by Relay Server –The load of network is reduced. –The number of clients increases. n n n m m n×m

10 10 Issues in the Current Model I.A heavy load is applied at the network between client and server. II.The number of clients is limited by bandwidth of network between client and server. III.Sound is interrupted or noise occurs.

11 11 Solution of Issue III [1 of 5] Factor of Issue III Delay by Retransmission of TCP TCP ・・・ Transmission Control Protocol

12 12 Solution of Issue III [2 of 5] The Communication with TCP Ack nowledgement Client has received the packet.

13 13 Solution of Issue III [3 of 5] The Communication with TCP LOSS! When a packet is lost. Ack nowledgement ??

14 14 Solution of Issue III [3 of 5] The Communication with TCP When a packet is lost. RetransmissionNo ack nowledgement!

15 15 Solution of Issue III [4 of 5] Delay by Retransmission Control of TCP Buffer data at clients Replace TCP with UDP in transport layer – UDP has no Retransmission Control – UDP ・・・ User Datagram Protocol

16 16 Solution of Issue III [5 of 5] UDP is not reliable for arrival of packet Need to guarantee for packet loss in application layer FEC (Forward Error Correction) – FEC resotores lost packets

17 17 FEC CodeErrorUsage Hamming code Ramdom Error semiconductor memory storage of computers BCH codesatellite broadcasting Fire code Burst Error magnetic memory storage of computers Reed Solomon codeaudio compact disk Burst Error Reed Solomon code

18 18 Reed Solomon Code 4bit (8, 4) RS code divide into 4 blocks every 4 bits encode errors correct 2 error blocks available in the network with knowing packet loss rate

19 19 Existing System Proposal of System [1 of 4] Provide Gateways Proposed System little packet loss much packet loss Do not modify applications

20 20 Proposal of System [2 of 4] The Stream of Sound Data HTTP/TCP RS code/UDP HTTP/TCP UDP→TCP RS decoding UDP→TCP RS decoding TCP→UDP RS encoding

21 21 Proposal of System [3 of 4] RS encoding and division into packets

22 22 Proposal of System [4 of 4] Packet Format –block number position of the packet –number in a block position of the packet in a block –real data size date size of a block brefore encoding (generally 4096 bytes)

23 23 Evaluation [1 of 5] Compare the communication quality in the proposed model with the existing model –Interruption and Noise –Connect-able Time

24 24 Evaluation [2 of 5] Evaluating environment –server ・・・ Internet Radio Station FOR (in IPU) –router ・・・ 20 –MP3 bitrate ・・・ 32kbps –Reed Solomon code ・・・ 8 bits (32, 16) RS –packet loss rate ・・・ unknown listen to noise or interruption mesure connect-able time

25 25 Evaluation [3 of 5] Interruption and Noise experimental time : 300 seconds less interruption and noise

26 26 Evaluation [4 of 5] Connect-able Time (connection between client and relay server) –Existing model 5 minutes at the worst –proposed model never disconnect (300 minutes) more difficult to disconnect

27 27 Evaluation [5 of 5] Interruption and Noise –decreased Connect-able Time –increased Communication quality improved in the real network without knowing packet loss rate.

28 28 Future Work Value-added services provided between a server and relay servers –bitrate conversion for bandwidth constraint environment (ex. PHS, mobile user) –different Commercial Message for each relay server Value-added Servive

29 29 Summary The issue of Interruption and Noise UDP and Reed Solomon code Provided gateways using Reed Solomon code into the existing system. –Not modify the existing applications. The communication quality improved in the proposed system than the existing system –In the real network without knowing packet loss rate

30 30 Thank You!


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