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The Proposal Details
Implementation of Transcoder for effective voice communication
Networking
Many a times it happens that you call a person and you find that the other person is busy. Most probable cause can be that the other person is connected to the Internet through the Dial-up connection. The person connected to the Internet has no way to know about this i.e. to get a notification of the call and ultimately the communication is not established. To solve this problem a VCAS(Voice Call Alert System) was developed. The purpose of this system is to alert a person connected to the Internet via a Dial up connection that a call is waiting for him. The system gives him the option of accepting the call and setting up a voice based communication simultaneously with the data packets over the IP channel. The Internet telephony system presently used in India uses the ADPCM encoding format for sending the digitized voice over the IP channel. But ADPCM format requires at least bandwidth of 32Kbps for efficient communication. However the current bandwidths available on the Dial-up connections rarely exceeds 28Kbps. So we require a format through which we can send digitized voice using less bandwidth. Our project is intended to improve the efficiency of this VCAS system by replacing the ADPCM packets (at least 32Kbps bandwidth requirement) with GSM 6.10 packets which require only 13.5Kbps bandwidth , thus improving the efficiency. So our aim, in this project, will be to develop a product that will receive the ADPCM packets from the VoIP server and convert it into the more efficient GSM format and dispatch it to the IP channel. Similarly fetching the GSM packets from the IP channel and converting it back into the ADPCM format and transmitting it to the VoIP server.
* Transcoder 1. This will be coded in Java, using JMF support. 2. It will take voice packets in the ADPCM format and convert them into GSM format , in real time. * Server 1. The Transcoder will sit here. On one side it will take voice packets in ADPCM format from the remote client, feed them into the Transcoder, which will return the packets after converting them into GSM 6.10 format. The server will then forward these GSM packets to the other client. * Client 1. At the client end, we will capture the voice from a microphone this captured stream of voice will be sent to the server. 2. Here the voice is captured in ADPCM format. 3. The client at the other end will receive the transcoded voice packets, sent by server and will play them. 4. NOTE : For transferring the voice , we recommend using RTP protocol.
1. Voice over IP, White Paper, Revision 1.0 Telco Systems, A BATM Company (1.4.2002) 2. http://www.voip-calculator.com/protocols.htm, Voice over IP protocol required for Voice Transmission. 3. Rappaport Theodere. S, ?Wireless Communications Principles and Practice?, Second Edition, Pearson Education (2003) 4. Digital cellular telecommunications system (Phase 2+);Full rate speech; Transcoding (GSM 06.10 version 5.2.1 Release 1996), European Telecommunications Standards Institute 5. Garg Vijay K, Wilkes Joseph E, ?Principles and Applications of GSM? ,Pearson Education(2004) 6. http://computer.howstuffworks.com/ip-telephony1.htm, How Stuff Works - VOIP. 7. Grant Mick, ?Proactively Managing Voice Quality on VoIP Networks?, White Paper 01, Voice Quality Expert, (16th November 2004) 8. Schroeder M.R. and Atal B, ?Code-Excited Linear Prediction (CELP): High Quality Speech at Very Low Bit Rates?, Proc. ICASSP-85, p. 937, Tampa, Apr. 1985 9. Oliver G, ?Understanding and Managing Delay in the Global Voice network?,PMC-seirra webinar, Dec 2003. 10. Scourias, John; Overview of the Global System for Mobile Communications http://ccnga.uwaterloo.ca/~jscouria/GSM/gsmreport.html 11. Pam, Andrew, ?A comparison of Internet audio compression formats?, Serious 12. http://www.octasic.com/data/pack.html, OCT7102 ADPCM Compression Vocoder 13. http://www.tldp.org/HOWTO/VoIP-HOWTO.html T. P. Barnwell III, K. Nayebi and C. H. Richardson, ?SPEECH CODING, A computer Laboratory Textbook?, John Wiley & Sons, Inc. 1996. 14. M.R. Schroeder and B. Atal, ?Code-Excited Linear Prediction (CELP): High Quality Speech at Very Low Bit Rates?, Proc. ICASSP-85, p. 937, Tampa, Apr. 1985 15. ftp://ftp.cs.cmu.edu/project/fgdata/speech-compression. 16. http://www.ddj.com/DrDobb?sJournal 17. http://www.data-compression.com