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G. Barberis

Bio: G. Barberis is an academic researcher from CSELT. The author has contributed to research in topics: Transmission delay & Packet switching. The author has an hindex of 1, co-authored 1 publications receiving 15 citations.

Papers
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Journal ArticleDOI
TL;DR: In this article, the authors considered the possibility of introducing packetized voice traffic into a packet-switched network and proposed simplified protocols and priority rules for voice handling, which are compared by means of analytical tools and simulation experiments considering the presence of voice, interactive, and batch data packets.
Abstract: This paper considers the possibility of introducing packetized voice traffic into a packet-switched network. It is well known that the network must assure voice packets sufficient delay characteristics for conversational speech, i.e., low delay between speaker and listener and low delay jitter or variance. To reach these goals, simplified protocols and priority rules for voice handling are proposed and evaluated. A model of a packet switching node structure capable of handling both data and voice is derived for both analytical and simulation approaches. The use of low bit rate voice encoders is considered. The necessity of avoiding the transmission of silent intervals is discussed in relation to the behavior of packet voice receivers. Proposed strategies are compared by means of analytical tools and simulation experiments considering the presence of voice, interactive, and batch data packets.

15 citations


Cited by
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Patent
08 Jun 1994
TL;DR: In this article, the authors present a digital communication system where multiple media data sources are time multiplexed into a packetized data stream, each packet having an assigned priority and the packetised data stream transmitted in substantially the order of assigned priority.
Abstract: The present invention is embodied in a digital communication system where multiple media data sources are time multiplexed into a packetized data stream, each packet having an assigned priority and the packetized data stream transmitted in substantially the order of assigned priority. At both the transmit side, and the receive side, audio packets are given priority processing over video packets, which in turn have priority over text/graphics data packets. Continuous real time audio playback is maintained at the receiver by delaying the playback of received audio in a first in/first out (FIFO) buffer providing a delay at least equal to the predicted average packet delay for the communication system. Optionally, the average system delay is continuously monitored, and the audio playback delay time is adjusted accordingly. Audio playback is slowed or accelerated in order to shrink or grow the difference in time between the sender and receiver. In another aspect of the invention, a conference of three or more callers is created by broadcasting a common packetized data stream to all conference callers.

210 citations

Patent
25 Sep 1996
TL;DR: In this paper, a utility for enabling real-time, point-to-point communications over computer networks between users having dynamically assigned Internet Protocol addresses includes the ability to identify incoming communications, and, based on the identity of the incoming communication, selectively responding.
Abstract: A utility for enabling real-time, point-to-point communications over computer networks between users having dynamically assigned Internet Protocol addresses includes the ability to identify incoming communications, and, based on the identity of the incoming communication, selectively responding. In one embodiment, an information profile contained within an incoming signal is compared to a plurality of information profiles contained within the personal directory of a user, and, if a match occurs, the notification signal associated with the matched profile is generated. In an alternative embodiment, the information profile contained within the incoming communication includes an identifier of a notification signal which is used to select from a number of predefined notification signals resident within the user's private directory.

81 citations

Patent
01 Mar 2002
TL;DR: In a telephone conferencing system, a digital communication network such as the Internet is used to establish and control the telephone connections between multiple conferees with the telephone network being the means of exchanging verbal information as mentioned in this paper.
Abstract: In a telephone conferencing system, a digital communication network such as the Internet is used to establish and control the telephone connections between multiple conferees with the telephone network being the means of exchanging verbal information. Each conferee may have a computer connected to the digital network, and each has an independent telephone instrument connected to the public switched telephone network. An in-charge conferee utilizes his computer containing appropriate software to initiate the conference and to control the participation of the conferees. The in-charge conferee sends digital control signals to a switch interface controlling a telephone switch as a gateway to the telephone network using SS7 control signals. These SS7 control signals include the commands by which the conferee telephones are rung up, brought on line, or dropped from the conference. The switch provides telephone status information back over the digital network, and the in-charge conferee, as well as other conferees provided with appropriate software, display this status information on their PC monitors.

72 citations

Patent
Guy G. Riddle1
28 May 2004
TL;DR: In this paper, the first endpoint transmits a request message to each of the plurality of second endpoints in order to query each of them whether they can receive transmissions broadcast to the first multicast address.
Abstract: A method and apparatus for optimizing transmission of data to a plurality of second endpoints in a system wherein a first endpoint is providing data to the plurality of second endpoints each connected by a point-to-point communication channels. This may be useful in teleconferencing applications with a plurality of participants (endpoints) or broadcast server applications. The first endpoint activates a multicast communication channel having a first multicast address and commences broadcast of the data over the multicast communication channel. The first endpoint transmits a request message to each of the plurality of second endpoints in order to query each of the second endpoints whether they can receive transmissions broadcast to the first multicast address. Certain of the plurality of second endpoints transmit an acknowledgment message if they can receive transmissions broadcast to the first multicast address, and the first endpoint receives the acknowledgment message. Then, for each acknowledgment message received from certain of the plurality of second endpoints, the first endpoint deactivates the point-to-point communication channel and the certain of the plurality of second endpoints.

63 citations

Journal ArticleDOI
TL;DR: The focus is on the integration of voice and data at the switching level, and different switching approaches are compared and some integrated-switching systems are described.
Abstract: A basic understanding of the technical problems in integrating voice and data is provided. The different types of traffic found in communication systems are examined. Integration is investigated at different levels. In particular, the focus is on the integration of voice and data at the switching level. Different switching approaches are compared and some integrated-switching systems are described. >

49 citations