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