Optimization of source and channel coding for voice over IP

Yicheng Huang; Jari Korhonen; Ye Wang

doi:10.1109/ICME.2005.1521388

Optimization of source and channel coding for voice over IP

Yicheng Huang^*, Jari Korhonen, Ye Wang

^*Corresponding author for this work

National University of Singapore

Research output: Chapter in Book/Report/Conference proceeding › Published conference contribution

10 Citations (Scopus)

Abstract

Voice over Internet Protocol (VoIP) applications must typically choose a tradeoff between the bits allocated for Forward Error Correcting (FEC) and that for the source coding to achieve the best speech quality at a given packet loss rate. In this paper, we present a new scheme to optimize the speech quality subject to the bandwidth constraints and the packet loss rate. The scheme adopts Adaptive Multi-Rate (AMR) speech codec along with a FEC scheme based on Exclusive OR (XOR) operations. Retransmission is also taken into account if the Round Trip Time (RTT) is within a certain limit. We use a simplified E-Model as objective metric. Subjective listening tests show that our scheme improves the perceptual speech quality significantly compared to the non-adaptive baseline speech transmission system.

Original language	English
Title of host publication	IEEE International Conference on Multimedia and Expo, ICME 2005
Publisher	IEEE Press
Pages	173-176
Number of pages	4
ISBN (Print)	0780393325, 9780780393325
DOIs	https://doi.org/10.1109/ICME.2005.1521388
Publication status	Published - 2005
Event	IEEE International Conference on Multimedia and Expo, ICME 2005 - Amsterdam, Netherlands Duration: 6 Jul 2005 → 8 Jul 2005

Conference

Conference	IEEE International Conference on Multimedia and Expo, ICME 2005
Country/Territory	Netherlands
City	Amsterdam
Period	6/07/05 → 8/07/05

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10.1109/ICME.2005.1521388

Cite this

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title = "Optimization of source and channel coding for voice over IP",

abstract = "Voice over Internet Protocol (VoIP) applications must typically choose a tradeoff between the bits allocated for Forward Error Correcting (FEC) and that for the source coding to achieve the best speech quality at a given packet loss rate. In this paper, we present a new scheme to optimize the speech quality subject to the bandwidth constraints and the packet loss rate. The scheme adopts Adaptive Multi-Rate (AMR) speech codec along with a FEC scheme based on Exclusive OR (XOR) operations. Retransmission is also taken into account if the Round Trip Time (RTT) is within a certain limit. We use a simplified E-Model as objective metric. Subjective listening tests show that our scheme improves the perceptual speech quality significantly compared to the non-adaptive baseline speech transmission system.",

author = "Yicheng Huang and Jari Korhonen and Ye Wang",

year = "2005",

doi = "10.1109/ICME.2005.1521388",

language = "English",

isbn = "0780393325",

pages = "173--176",

booktitle = "IEEE International Conference on Multimedia and Expo, ICME 2005",

publisher = "IEEE Press",

note = "IEEE International Conference on Multimedia and Expo, ICME 2005 ; Conference date: 06-07-2005 Through 08-07-2005",

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T1 - Optimization of source and channel coding for voice over IP

AU - Huang, Yicheng

AU - Korhonen, Jari

AU - Wang, Ye

PY - 2005

Y1 - 2005

N2 - Voice over Internet Protocol (VoIP) applications must typically choose a tradeoff between the bits allocated for Forward Error Correcting (FEC) and that for the source coding to achieve the best speech quality at a given packet loss rate. In this paper, we present a new scheme to optimize the speech quality subject to the bandwidth constraints and the packet loss rate. The scheme adopts Adaptive Multi-Rate (AMR) speech codec along with a FEC scheme based on Exclusive OR (XOR) operations. Retransmission is also taken into account if the Round Trip Time (RTT) is within a certain limit. We use a simplified E-Model as objective metric. Subjective listening tests show that our scheme improves the perceptual speech quality significantly compared to the non-adaptive baseline speech transmission system.

AB - Voice over Internet Protocol (VoIP) applications must typically choose a tradeoff between the bits allocated for Forward Error Correcting (FEC) and that for the source coding to achieve the best speech quality at a given packet loss rate. In this paper, we present a new scheme to optimize the speech quality subject to the bandwidth constraints and the packet loss rate. The scheme adopts Adaptive Multi-Rate (AMR) speech codec along with a FEC scheme based on Exclusive OR (XOR) operations. Retransmission is also taken into account if the Round Trip Time (RTT) is within a certain limit. We use a simplified E-Model as objective metric. Subjective listening tests show that our scheme improves the perceptual speech quality significantly compared to the non-adaptive baseline speech transmission system.

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U2 - 10.1109/ICME.2005.1521388

DO - 10.1109/ICME.2005.1521388

M3 - Published conference contribution

AN - SCOPUS:33750547559

SN - 0780393325

SN - 9780780393325

SP - 173

EP - 176

BT - IEEE International Conference on Multimedia and Expo, ICME 2005

PB - IEEE Press

T2 - IEEE International Conference on Multimedia and Expo, ICME 2005

Y2 - 6 July 2005 through 8 July 2005

ER -

Optimization of source and channel coding for voice over IP

Abstract

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