公开(公告)号：US06408327B1

公开(公告)日：2002-06-18

申请号：US09218135

申请日：1998-12-22

申请人： R. Scott McClennon ,  Andre J. Robert

发明人： R. Scott McClennon ,  Andre J. Robert

IPC分类号： G06F1516

CPC分类号： H04S1/00 ,  H04L12/1813 ,  H04M3/56 ,  H04M3/568 ,  H04M7/006

摘要： A method of and system for facilitating synthetic stereo audio conferencing of a plurality of users over a local or wide area network (LAN/WAN). The disclosure describes a system architecture using a LAN/WAN to support synthetic audio conferencing thus allowing, at a receiving user location, for the spatially distinct virtual “placement” of the other parties involved in a conference call. This enables the user to differentiate between the various other parties and to discern which one or more of the parties are talking at any given time. In a first embodiment of the invention, both stereo synthesis and conferencing are client-based functions. In a second embodiment of the invention, both stereo synthesis and conferencing are server-based functions. In a third embodiment of the invention, stereo synthesis is a server-based function and conferencing is a hybrid client/server function using multicast.

摘要翻译： 一种用于促进通过本地或广域网（LAN / WAN）的多个用户的合成立体声音频会议的方法和系统。 本公开描述了使用LAN / WAN来支持合成音频会议的系统架构，从而在接收用户位置处允许在电话会议中涉及的其他方的空间上不同的虚拟“放置”。 这使得用户能够区分各方，并在任何给定的时间辨别哪一方或多方当事人正在交谈。 在本发明的第一实施例中，立体声合成和会议都是基于客户端的功能。 在本发明的第二实施例中，立体声合成和会议都是基于服务器的功能。 在本发明的第三实施例中，立体声合成是基于服务器的功能，会议是使用多播的混合客户端/服务器功能。

公开(公告)号：US06327276B1

公开(公告)日：2001-12-04

申请号：US09218142

申请日：1998-12-22

申请人： Andre J. Robert ,  R. Scott McClennon

发明人： Andre J. Robert ,  R. Scott McClennon

IPC分类号： H04L1216

CPC分类号： H04L65/4038 ,  H04J3/0632 ,  H04J3/1694 ,  H04L12/1813 ,  H04L29/06027 ,  H04M3/56 ,  H04M7/006 ,  H04M7/1205

摘要： A method of and system for conferencing a plurality of clients over a local or wide area network using multicasting where the conferencing function is distributed between a server and the clients. The invention takes advantage of certain capabilities of existing client equipment and multicasting capabilities of the server to distribute the conference function between the server and the clients in a conference call over a LAN/WAN in a way that reduces network congestion and protocol complexity. The system of the present invention for providing conferencing over a local or wide area network, includes: a plurality of clients connected to the network for transmitting signals to and receiving signals from the network; and a server connected to the network for receiving a plurality of signals transmitted from the plurality of clients, mixing the received signals to create a single multicast signal, and transmitting the multicast signal to each of the clients. Each of the plurality of received signals is made up of data packets of a defined length. The server includes: a jitter buffer for synchronizing the data packets received from the plurality of clients; an adjustable gain/loss controller for applying adjustable gain/loss to the synchronized data packets—according to speech activity and average level at each client signal; a mixer for mixing the data packets of the plurality of signals to create the single multicast signal for transmission to each of the clients. An individual client receiving the multicast signal transmitted from the server includes an echo controller for estimating and removing, from the multicast signal, a signal component corresponding to a signal transmitted from that client.

摘要翻译： 一种用于使用组播在本地或广域网上会议多个客户端的方法和系统，其中会议功能分布在服务器和客户端之间。 本发明利用现有客户端设备的某些能力和服务器的组播能力，以减少网络拥塞和协议复杂度的方式通过LAN / WAN在电话会议中在服务器和客户端之间分配会议功能。 本发明的用于在局域网或广域网上提供会议的系统包括：连接到网络的多个客户端，用于向网络发送信号并从网络接收信号; 以及连接到网络的服务器，用于接收从多个客户端发送的多个信号，混合接收到的信号以创建单个多播信号，并将多播信号发送到每个客户端。 多个接收信号中的每一个由具有限定长度的数据分组组成。 服务器包括：抖动缓冲器，用于同步从多个客户端接收的数据分组; 根据每个客户端信号的语音活动和平均电平，可调节的增益/丢失控制器用于对同步的数据分组施加可调节的增益/损失; 混合器，用于混合多个信号的数据分组，以创建单个多播信号，以便传输给每个客户端。 接收从服务器发送的组播信号的个体客户机包括：回声控制器，用于从多播信号中估计和去除与从该客户端发送的信号对应的信号分量。

公开(公告)号：US06324170B1

公开(公告)日：2001-11-27

申请号：US09150698

申请日：1998-09-10

申请人： R. Scott McClennon ,  Andre J. Robert

发明人： R. Scott McClennon ,  Andre J. Robert

IPC分类号： H04B320

CPC分类号： H04M3/002 ,  H04B3/23 ,  H04J3/0682 ,  H04M7/1275

摘要： The present invention provides a method of and system for controlling and effectively reducing the echo introduced in telephone connections over variable delay networks such as LAN/WAN networks. The invention includes estimation of the round-trip user-to-user delay in the telephone connection. This estimation can be made in a number of different ways, including using timestamps in existing data packets according to a Real-Time Protocol (RTP) or a Real-Time Control Protocol (RTCP) and by sending timestamps in data packets according to an Internet Control Message Protocol (ICMP). The invention further includes determination of an echo control target for the telephone connection based upon the estimated user-to-user delay. This target represents the total amount of echo control required to limit the probability of objectionable echo to less than a predetermined threshold. The amount of echo control required increases as the length of the estimated user-to-user delay increases. The invention further includes determination of the minimum amount of signal-dependent loss required to supplement the acoustic echo cancellation in order to reach the determined echo control target. Then, in the invention, the determined amount of signal-dependent loss is applied to the telephone connection to effectively reduce echo based upon the estimated user-to-user delay.

摘要翻译： 本发明提供一种用于控制和有效地减少通过诸如LAN / WAN网络的可变延迟网络引入的电话连接中的回波的方法和系统。 本发明包括对电话连接中往返用户到用户延迟的估计。 可以以多种不同的方式进行该估计，包括根据实时协议（RTP）或实时控制协议（RTCP）在现有数据分组中使用时间戳，并且根据因特网在数据分组中发送时间戳 控制消息协议（ICMP）。 本发明还包括基于估计的用户到用户延迟来确定用于电话连接的回波控制目标。 该目标表示将令人不快的回波的概率限制在小于预定阈值所需的回波控制的总量。 随着估计的用户到用户延迟的长度增加，所需的回声控制量增加。 本发明还包括为了达到所确定的回波控制目标而确定补充声学回声消除所需的信号相关损耗的最小量。 然后，在本发明中，将所确定的信号相关损失量应用于电话连接，以根据所估计的用户到用户延迟来有效地减少回波。