Traffic estimation and modeling has been a crucial issue in many research areas of communication networks. For example, intermediate routers in networks estimate the rate of packet flows via queue size information in order to maximize throughput and provide fairness between flows. Also, many senders in the end system such as multimedia applications estimate end-to-end delay between a sender and...

For distinctively transporting voice data with silence suppression over Asynchronous Transfer Mode (Am) networks via the Variable Bit Rate (VBR) service, the problem of jitter introduced from the network often renders the speech unintelligible. It is thus indispensable to offer intramedia synchronization to remove jitter while retaining minimal playout delay. In this paper, we propose a neural–...

Real time voice applications typically produce uniformly spaced voice packets and faithful reconstruction demands that these be played out at the same intervals. Best effort packet networks, however, produce variable delays on different packets and the receiver is required to buffer the received packets before playout. Excessive buffering delays deteriorate the system performance for interactiv...

This paper examines the issues related to the transmission of voice over packet networks using the Internet Protocol (IP). We focus on studying the delays that are generated in the terminal, which is a Unix workstation equipped with IP voice application software. Delay components in the terminal are presented. We measure the processing delays in the terminal using different audio codecs and mea...

In this work, we address the transport of high quality voice over the Internet with a particular concern for delays. Transport of interactive audio over IP networks often suffers from packet loss and variations in the network delay (jitter). Forward Error Correction (FEC) mitigates the impact of packet loss at the expense of an increase of the end-to-end delay and the bit rate requirement of an...

100 Delay Reduction Techniques for Playout Bu ering Cormac J. Sreenan, Jyh-Cheng Chen, Member, IEEE, Prathima Agrawal, Fellow, IEEE, and B. Narendran Abstract| Receiver synchronization of continuous media streams is required to deal with delay di erences and variations resulting from delivery over packet networks such as the Internet. This function is commonly provided using per-stream playout ...

In Internet audio applications, delay and delay jitter affect applications' quality of service mostly. Since packet delays are different and changing over time, the receiver needs to buffer some amount of packets before playout. Therefore, the amount of buffered packets and timing of playout are very important for the performance of the applications. Here we adopt an auto-regressive (AR) model ...

Transmitting real-time voice over the Internet is a technological challenge. Variation in network characteristics introduces jitter to the propagating voice packets. Jitter hampers voice quality and makes the VoIP call uncomfortable to the user. Often buffers are used to store the received packets for a short time before playing them at equal spaced intervals to minimize jitter. Choosing optimu...

The poor quality of Voice over IP can be improved by adaptive playout buffering at the receiver. This technique dynamically adapts the playout deadline to network conditions, thus minimizing both late packet loss and buffering time. A standard playout buffer strategy uses an estimate (Exponentially Weighted Moving Average) of the mean and variance of network delay to set the playout deadline. T...