In this paper we investigate the performance of various buffer algorithms that might be implemented in H.323 VoIP applications. The main objective of those algorithms is to minimize effect of the delay jitter. We have tested those algorithms in the Internet using H.323 VoIP terminals. Our results show that the algorithm proposed by us can achieve the lowest rate of lost packets while adding acceptably small delays
Real time voice applications typically produce uniformly spaced voice packets and faithful reconstru...
"Thesis (M.S.)--Wichita State University, College of Engineering, Dept. of Electrical and Computer E...
The end to end delay is a critical factor in the perceived quality of service for Voice over IP appl...
This work presents mathematical properties to estimate the optimal buffer delay used in applications...
Factors like network delay, latency and bandwidth significantly affect the quality of communication ...
Factors like network delay, latency and bandwidth significantly affect the quality of communication ...
A novel playout algorithm for VoIP applications is presented. The playout times of voice packets are...
Audio communication over IP-based networks represents one of the most interesting research areas in...
Abstract — In this paper, a new adaptive receiver buffer adjust algorithm is proposed for Voice and ...
As the Internet is a best-effort delivery network, audio packets may be delayed or lost en route to...
In Internet-protocol (IP) telephony, problems of transmission delay variations are frequently addres...
To combat jitter problems in voice streaming over packet networks, playout buffering algorithms are ...
Abstract—The quality of service limitation of today’s Internet is a major challenge for real-time vo...
Recent interest in supporting packet-audio applications over wide area networks has been fueled by t...
The MPEG-4 Enhanced Low Delay AAC (AAC-ELD) codec extends the application area of the Advanced Audio...
Real time voice applications typically produce uniformly spaced voice packets and faithful reconstru...
"Thesis (M.S.)--Wichita State University, College of Engineering, Dept. of Electrical and Computer E...
The end to end delay is a critical factor in the perceived quality of service for Voice over IP appl...
This work presents mathematical properties to estimate the optimal buffer delay used in applications...
Factors like network delay, latency and bandwidth significantly affect the quality of communication ...
Factors like network delay, latency and bandwidth significantly affect the quality of communication ...
A novel playout algorithm for VoIP applications is presented. The playout times of voice packets are...
Audio communication over IP-based networks represents one of the most interesting research areas in...
Abstract — In this paper, a new adaptive receiver buffer adjust algorithm is proposed for Voice and ...
As the Internet is a best-effort delivery network, audio packets may be delayed or lost en route to...
In Internet-protocol (IP) telephony, problems of transmission delay variations are frequently addres...
To combat jitter problems in voice streaming over packet networks, playout buffering algorithms are ...
Abstract—The quality of service limitation of today’s Internet is a major challenge for real-time vo...
Recent interest in supporting packet-audio applications over wide area networks has been fueled by t...
The MPEG-4 Enhanced Low Delay AAC (AAC-ELD) codec extends the application area of the Advanced Audio...
Real time voice applications typically produce uniformly spaced voice packets and faithful reconstru...
"Thesis (M.S.)--Wichita State University, College of Engineering, Dept. of Electrical and Computer E...
The end to end delay is a critical factor in the perceived quality of service for Voice over IP appl...