DocumentCode :
1749849
Title :
Study on the application of an AMR speech codec to VoIP
Author :
Seo, Jeong Wook ; Woo, Se Jeong ; Bae, Keun Sung
Author_Institution :
Sch. of Electron. & Electr. Eng., Kyungpook Nat. Univ., Taegu, South Korea
Volume :
3
fYear :
2001
fDate :
2001
Firstpage :
1373
Abstract :
Degradation of speech quality caused by packet loss of voice traffic is still one of critical technical barriers of the VoIP system. We propose a new VoIP system that can adapt transmission bit rate flexibly to network conditions to reduce packet loss. In order to determine the transmission bit rate depending upon the network conditions on a frame basis, we use the time-stamp parameter in the RTP of the H.323 protocol. Experimental results demonstrate that the proposed system is very promising to reduce packet loss that leads to improvement of speech quality
Keywords :
Internet telephony; adaptive codes; packet switching; protocols; speech codecs; AMR speech codec; H.323 protocol; RTP; VoIP; adaptive multi-rate speech codec; network conditions; packet loss; real-time protocol; speech quality; transmission bit rate; voice traffic; Bit rate; Communication system control; Control systems; Degradation; Internet telephony; Propagation losses; Protocols; Quality of service; Speech codecs; Telecommunication traffic;
fLanguage :
English
Publisher :
ieee
Conference_Titel :
Acoustics, Speech, and Signal Processing, 2001. Proceedings. (ICASSP '01). 2001 IEEE International Conference on
Conference_Location :
Salt Lake City, UT
ISSN :
1520-6149
Print_ISBN :
0-7803-7041-4
Type :
conf
DOI :
10.1109/ICASSP.2001.941184
Filename :
941184
Link To Document :
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