公开(公告)号：US20020118650A1

公开(公告)日：2002-08-29

申请号：US09796400

申请日：2001-02-28

Inventor： Ramanathan Jagadeesan ,  Luke K. Surazski

IPC: H04Q011/00 ,  H04L012/16 ,  H04L012/66

CPC classification number: H04L65/4038 ,  G10L19/012 ,  H04L29/06027 ,  H04M3/567 ,  H04Q11/0407 ,  H04Q2213/13034 ,  H04Q2213/1305 ,  H04Q2213/13201 ,  H04Q2213/1324 ,  H04Q2213/13389

Abstract: Devices, software and methods are provided for generating aggregate comfort noise for teleconferencing over IP networks. A transcoding component includes a decoder for decoding streams of packets. A summing component has a summer with summing inputs to receive the decoded streams of packets. The summing component has at least one silence flag input, and an additional signaling path is used by the transcoding component to signal to the silence flag input if any of the decoded streams of packets includes a silence identification packet. In another embodiment, the summing component may or may not include the silence flag input, but the device includes an aggregate comfort noise generation component. The aggregate comfort noise may be programmed to be a balanced representation of all background noises.

Abstract translation: 提供设备，软件和方法，用于产生通过IP网络进行电话会议的聚合舒适噪声。 代码转换组件包括用于解码分组流的解码器。 求和分量具有一个具有加法输入的加法器，用于接收经解码的数据包流。 求和组件具有至少一个静音标志输入，并且如果任何解码的分组流包括静音识别分组，则代码转换组件使用附加信令路径来发信号到静音标志输入。 在另一个实施例中，求和组件可以包括或不包括静音标志输入，但是该设备包括聚合舒适噪声生成组件。 总体舒适噪声可以被编程为所有背景噪声的平衡表示。

公开(公告)号：US06377671B2

公开(公告)日：2002-04-23

申请号：US09870486

申请日：2001-06-01

Applicant: Toshiaki Tanaka ,  Hiroshi Mano ,  Setsuo Kimura ,  Yoshihito Nagano ,  Masayuki Tsurusaki ,  Takashi Watanabe ,  Terunori Suwa ,  Isaku Komuro ,  Yoshihiro Kawauchi ,  Nobuhiro Masaki

Inventor： Toshiaki Tanaka ,  Hiroshi Mano ,  Setsuo Kimura ,  Yoshihito Nagano ,  Masayuki Tsurusaki ,  Takashi Watanabe ,  Terunori Suwa ,  Isaku Komuro ,  Yoshihiro Kawauchi ,  Nobuhiro Masaki

IPC: H04M100

CPC classification number: H04Q3/625 ,  H04M1/57 ,  H04M3/22 ,  H04M3/26 ,  H04M3/42 ,  H04M3/42059 ,  H04M3/42127 ,  H04M3/42195 ,  H04M3/42314 ,  H04M3/42323 ,  H04M3/428 ,  H04M3/4283 ,  H04M3/533 ,  H04M3/58 ,  H04M9/00 ,  H04M9/002 ,  H04M2201/38 ,  H04M2207/08 ,  H04M2242/22 ,  H04Q1/028 ,  H04Q1/10 ,  H04Q3/545 ,  H04Q3/58 ,  H04Q11/0407 ,  H04Q2201/10 ,  H04Q2201/806 ,  H04Q2213/13003 ,  H04Q2213/13034 ,  H04Q2213/13036 ,  H04Q2213/13076 ,  H04Q2213/1308 ,  H04Q2213/13091 ,  H04Q2213/13093 ,  H04Q2213/13096 ,  H04Q2213/13097 ,  H04Q2213/13102 ,  H04Q2213/13103 ,  H04Q2213/13106 ,  H04Q2213/1315 ,  H04Q2213/13152 ,  H04Q2213/1316 ,  H04Q2213/13162 ,  H04Q2213/13173 ,  H04Q2213/13174 ,  H04Q2213/13175 ,  H04Q2213/1318 ,  H04Q2213/13196 ,  H04Q2213/13199 ,  H04Q2213/13201 ,  H04Q2213/13203 ,  H04Q2213/13204 ,  H04Q2213/13209 ,  H04Q2213/13213 ,  H04Q2213/13216 ,  H04Q2213/1322 ,  H04Q2213/13242 ,  H04Q2213/13251 ,  H04Q2213/1326 ,  H04Q2213/13282 ,  H04Q2213/13292 ,  H04Q2213/13299 ,  H04Q2213/13302 ,  H04Q2213/13305 ,  H04Q2213/1332 ,  H04Q2213/13322 ,  H04Q2213/13334 ,  H04Q2213/1334 ,  H04Q2213/13362 ,  H04Q2213/13377 ,  H04Q2213/13392 ,  H04Q2213/13405

Abstract: According to this invention, a digital key telephone system connected to an analog public network NW having a function of transmitting a ringing signal including identification information of a calling line through a subscriber line (CO line), accommodating a plurality of extension lines each connected to a digital key telephone (DKT) 2 or a standard telephone (STT) 4 as an extension terminal, and having a function of switching and connecting the subscriber line to the plurality of extension lines or the extension lines to each other includes a called party storage means storing, in advance, information representing the correlation between the calling line and the extension terminals 2 and 4 as a called terminal. When a ringing signal arrives from the analog communication network NW, calling line identification information (caller ID) contained in the ringing signal is detected by a calling line identification information interface unit (RCIU) 12. A control unit (RCTU) 16 determines the called extension terminal on the basis of the detected caller ID and the information stored in the storage means, so the extension terminal receives the call from the digital key telephone interface unit (RDKU) 13 or a standard telephone interface unit (RSTU) 15.

公开(公告)号：US20010028708A1

公开(公告)日：2001-10-11

申请号：US09870682

申请日：2001-06-01

Applicant: Kabushiki Kaisha Toshiba

Inventor： Toshiaki Tanaka ,  Hiroshi Mano ,  Setsuo Kimura ,  Yoshihito Nagano ,  Masayuki Tsurusaki ,  Takashi Watanabe ,  Terunori Suwa ,  Isaku Komuro ,  Yoshihiro Kawauchi ,  Nobuhiro Masaki

IPC: H04M001/00

CPC classification number: H04Q3/625 ,  H04M1/57 ,  H04M3/22 ,  H04M3/26 ,  H04M3/42 ,  H04M3/42059 ,  H04M3/42127 ,  H04M3/42195 ,  H04M3/42314 ,  H04M3/42323 ,  H04M3/428 ,  H04M3/4283 ,  H04M3/533 ,  H04M3/58 ,  H04M9/00 ,  H04M9/002 ,  H04M2201/38 ,  H04M2207/08 ,  H04M2242/22 ,  H04Q1/028 ,  H04Q1/10 ,  H04Q3/545 ,  H04Q3/58 ,  H04Q11/0407 ,  H04Q2201/10 ,  H04Q2201/806 ,  H04Q2213/13003 ,  H04Q2213/13034 ,  H04Q2213/13036 ,  H04Q2213/13076 ,  H04Q2213/1308 ,  H04Q2213/13091 ,  H04Q2213/13093 ,  H04Q2213/13096 ,  H04Q2213/13097 ,  H04Q2213/13102 ,  H04Q2213/13103 ,  H04Q2213/13106 ,  H04Q2213/1315 ,  H04Q2213/13152 ,  H04Q2213/1316 ,  H04Q2213/13162 ,  H04Q2213/13173 ,  H04Q2213/13174 ,  H04Q2213/13175 ,  H04Q2213/1318 ,  H04Q2213/13196 ,  H04Q2213/13199 ,  H04Q2213/13201 ,  H04Q2213/13203 ,  H04Q2213/13204 ,  H04Q2213/13209 ,  H04Q2213/13213 ,  H04Q2213/13216 ,  H04Q2213/1322 ,  H04Q2213/13242 ,  H04Q2213/13251 ,  H04Q2213/1326 ,  H04Q2213/13282 ,  H04Q2213/13292 ,  H04Q2213/13299 ,  H04Q2213/13302 ,  H04Q2213/13305 ,  H04Q2213/1332 ,  H04Q2213/13322 ,  H04Q2213/13334 ,  H04Q2213/1334 ,  H04Q2213/13362 ,  H04Q2213/13377 ,  H04Q2213/13392 ,  H04Q2213/13405

Abstract: According to this invention, a digital key telephone system connected to an analog public network NW having a function of transmitting a ringing signal including identification information of a calling line through a subscriber line (CO line), accommodating a plurality of extension lines each connected to a digital key telephone (DKT) 2 or a standard telephone (STT) 4 as an extension terminal, and having a function of switching and connecting the subscriber line to the plurality of extension lines or the extension lines to each other includes a called party storage means storing, in advance, information representing the correlation between the calling line and the extension terminals 2 and 4 as a called terminal. When a ringing signal arrives from the analog communication network NW, calling line identification information (caller ID) contained in the ringing signal is detected by a calling line identification information interface unit (RCIU) 12. A control unit (RCTU) 16 determines the called extension terminal on the basis of the detected caller ID and the information stored in the storage means, so the extension terminal receives the call from the digital key telephone interface unit (RDKU) 13 or a standard telephone interface unit (RSTU) 15.

公开(公告)号：US6122294A

公开(公告)日：2000-09-19

申请号：US852619

申请日：1997-05-07

Applicant: Danny K. Hsu ,  Wei Lin

Inventor： Danny K. Hsu ,  Wei Lin

IPC: H04J3/00 ,  H04J3/16 ,  H04M3/00 ,  H04Q11/04 ,  H04J3/18

CPC classification number: H04J3/1647 ,  H04J3/1635 ,  H04J3/1688 ,  H04Q11/0407 ,  H04Q2213/13034 ,  H04Q2213/1309 ,  H04Q2213/13093 ,  H04Q2213/13106 ,  H04Q2213/13176 ,  H04Q2213/13179 ,  H04Q2213/13201 ,  H04Q2213/13204 ,  H04Q2213/13216 ,  H04Q2213/13292 ,  H04Q2213/1334 ,  H04Q2213/13349 ,  H04Q2213/13396

Abstract: Adjunct apparatus for increasing network transmission capacity provides a low-cost, efficient solution for increasing the network transmission capacity of the existing telephone circuit switched network, while keeping the current network equipment unchanged. A Local Switch Network (LSN) Adjunct (LSNA) and a Tandem/Toll Switch Network (TSN) Adjunct (TSNA) interface with standard network elements, such as switches and cross connect equipment. These network adjuncts, comprising a set of low-bit rate speech coders, a dynamic timeslot manager and other supporting functions, advantageously transmit to and receive from a T1/T3/OC3/E1 trunk. More than one channel of voice is carried on one 64 Kbps DS0 timeslot, while still maintaining the voice at toll quality. A sub-timeslot and sub-timeslot bundling are introduced in the standard T1 or E1 frame where each sub-timeslot is analogous to a single bit of the typical eight bit word to provide more than 24 or 30 voice channels respectively. Inband control information is generated for carrying over a T1 or E1 trunk, for example, for mapping sub-timeslot bundles to channels. Conventional out-of-band signaling is provided via SS-7 or other out-of-band signaling system.

Abstract translation: 用于增加网络传输容量的辅助设备为保持现有网络设备不变而提供现有电话交换网络的网络传输容量的低成本，高效解决方案。 具有标准网元（如交换机和交叉连接设备）的本地交换机网络（LSN）辅助（LSNA）和串联/收费交换网络（TSN）辅助（TSNA）接口。 这些网络附件，包括一组低比特率语音编码器，动态时隙管理器和其他支持功能，有利地发送到T1 / T3 / OC3 / E1中继线并从T1 / T3 / OC3 / E1中继线接收。 一个64Kbps的DS0时隙上携带多个语音通道，同时仍然保持语音通话质量。 在标准T1或E1帧中引入子时隙和子时隙捆绑，其中每个子时隙类似于典型8位字的单个比特，以分别提供超过24或30个语音信道。 生成用于携带T1或E1中继线的带内控制信息，例如用于将子时隙束映射到信道。 通过SS-7或其他带外信令系统提供传统的带外信令。

公开(公告)号：US5619252A

公开(公告)日：1997-04-08

申请号：US636890

申请日：1996-04-24

Applicant: Misao Nakano

Inventor： Misao Nakano

IPC: H04N7/14 ,  H04M3/12 ,  H04M11/00 ,  H04M19/08 ,  H04Q11/04

CPC classification number: H04M19/08 ,  H04N7/148 ,  H04Q11/0428 ,  H04M3/12 ,  H04Q2213/13034 ,  H04Q2213/1308 ,  H04Q2213/13166 ,  H04Q2213/13167 ,  H04Q2213/13179 ,  H04Q2213/13201 ,  H04Q2213/13204 ,  H04Q2213/13209 ,  H04Q2213/1322 ,  H04Q2213/13292 ,  H04Q2213/13337

Abstract: A video telephone system which automatically switches during a normally operating communicating function to a minimum voice signal and a sub-power source when a master power source fails, and automatically switches back to the normally operating communicating function when the failure is remedied. The system includes first CODEC equipment which encodes a voice signal input from voice input-output equipment, and outputs various signals. Image pickup equipment receives an image signal and display equipment displays the image signal. The first CODEC equipment, the image pickup equipment and the display equipment function as voice and image signal encoding and decoding equipment. Second CODEC equipment encodes an input voice signal from the voice input-output equipment and outputs various signals. Line control equipment selectively outputs a signal input from the first or second CODECs for controlling switching of the first and second CODECs to the transmission path. Switching control equipment inputs a signal indicating the condition of the master power source, outputs a control signal in accordance with the input signal, and automatically controls voice communication by switching the first and second CODECs if the master power source fails.

Abstract translation: 一种视频电话系统，当主电源发生故障时，在正常操作的通信功能期间自动切换到最小语音信号和子电源，并且在故障得到补救时自动切换回正常操作的通信功能。 该系统包括对语音输入输出设备输入的语音信号进行编码的第一个CODEC设备，并输出各种信号。 摄像设备接收图像信号，显示设备显示图像信号。 第一个CODEC设备，摄像设备和显示设备作为语音和图像信号编码和解码设备。 第二个CODEC设备对来自语音输入输出设备的输入语音信号进行编码，并输出各种信号。 线路控制设备选择性地输出从第一或第二CODEC输入的信号，以控制第一和第二CODEC到传输路径的切换。 切换控制设备输入指示主电源状态的信号，根据输入信号输出控制信号，并且如果主电源发生故障，则通过切换第一和第二CODEC来自动控制话音通信。

公开(公告)号：US5563882A

公开(公告)日：1996-10-08

申请号：US507868

申请日：1995-07-27

Applicant: Richard F. Bruno ,  Robert E. Markowitz ,  Roy P. Weber

Inventor： Richard F. Bruno ,  Robert E. Markowitz ,  Roy P. Weber

IPC: H04M3/42 ,  H04M3/56 ,  H04M11/06 ,  H04N7/14 ,  H04N7/15 ,  H04Q11/04

CPC classification number: H04N7/152 ,  H04M3/567 ,  H04N7/147 ,  H04N7/148 ,  H04N7/15 ,  H04Q11/0428 ,  H04M2203/5018 ,  H04Q2213/1307 ,  H04Q2213/13093 ,  H04Q2213/13106 ,  H04Q2213/13174 ,  H04Q2213/13201 ,  H04Q2213/13208 ,  H04Q2213/13209 ,  H04Q2213/1324 ,  H04Q2213/13337

Abstract: A process for converting a point-to-point multimedia call into a bridged multimedia call for accommodating multi-party conferencing. The process utilizes the bearer channel configuration of the standard BRI format to disconnect, one at a time, each bearer channel from the point-m-point call and to reconnect each line to a multipoint control unit, whereupon the reconnected lines are individually reformatted in accordance with standard video phone protocol. The process facilitates the continuous and uninterrupted exchange of audio information and data between the point-to-point participants throughout the disconnection and reconfiguration stages of the conversion.

Abstract translation: 用于将点对多点呼叫转换成用于容纳多方会议的桥接多媒体呼叫的过程。 该过程利用标准BRI格式的承载信道配置从点到点呼叫断开每个承载信道，并将每个线路重新连接到多点控制单元，从而将重新连接的线路单独重新格式化 按照标准的视频电话协议。 该过程有助于在转换的断开和重新配置阶段期间，点对点参与者之间的音频信息和数据的连续且不间断的交换。

公开(公告)号：US5187591A

公开(公告)日：1993-02-16

申请号：US645113

申请日：1991-01-24

Applicant: Kenneth R. Guy ,  Andrew M. Kameya ,  Herbert M. Wildfeuer

Inventor： Kenneth R. Guy ,  Andrew M. Kameya ,  Herbert M. Wildfeuer

IPC: H04L29/10 ,  H04L27/00 ,  H04M11/00 ,  H04M11/06 ,  H04N1/00 ,  H04Q11/04

CPC classification number: H04M11/068 ,  H04L27/00 ,  H04Q2213/13034 ,  H04Q2213/13093 ,  H04Q2213/13106 ,  H04Q2213/13174 ,  H04Q2213/13199 ,  H04Q2213/13201 ,  H04Q2213/13292

Abstract: Modulated data information having an individual one of different baud rates and aural information are converted at a transmitter from analog to digital form. A controller separately identifies the digitized aural and modulated data information and, if modulated data, identifies the baud rate of such modulated data. The aural information is separately transformed. For modulated data, the information is separately processed in accordance with the different baud rates. The individual ones of aural and modulated data are then introduced to a common line for packetizing. The beginning of each packet is labelled to identify whether the packet contains aural information (such as voice or music) or modulated data (such as modem or facsimile), and, if modulated data, the particular baud rate of such information. An individual code identifies the end of each packet. The packetized information is then multiplexed in a common bus with other packetized aural and data information. At a receiver connected to the common bus, the multiplexed information is separated into the different packets. The packetized information representing individual ones of aural information (such as voice or music) and modulated data (such as music or facsimile information) is separated, in accordance with the packet labels, into aural information and modulated data and, if modulated data, is introduced to an individual one of different modulators each operative at an individual one of the different baud rates. The aural information is then transformed substantially to its original form at the transmitter and the modulated data is separately processed in accordance with the different baud rates.

公开(公告)号：US4996685A

公开(公告)日：1991-02-26

申请号：US335542

申请日：1989-04-10

Applicant: Thomas R. Farese ,  Kenneth E. Ferrere

Inventor： Thomas R. Farese ,  Kenneth E. Ferrere

IPC: H04L12/28 ,  H04L12/64 ,  H04Q11/04

CPC classification number: H04L12/5692 ,  H04L12/6418 ,  H04Q11/0457 ,  H04L2012/6443 ,  H04L2012/647 ,  H04L2012/6475 ,  H04Q2213/13093 ,  H04Q2213/13138 ,  H04Q2213/13141 ,  H04Q2213/13176 ,  H04Q2213/13201 ,  H04Q2213/13202 ,  H04Q2213/13204 ,  H04Q2213/13209 ,  H04Q2213/13296 ,  H04Q2213/13389 ,  H04Q2213/174 ,  H04Q2213/201 ,  H04Q2213/292 ,  H04Q2213/296

Abstract: A technique is disclosed for use in conjunction with an ISDN communications system for permitting a host computer, that is executing a host session with a user and is connected through the system, to dynamically change an ISDN access path, that connects the user to the host computer and carries the host session therebetween, between a packet switched connection and a circuit switched connection during the occurrence of the session in order to provide the particular connection that is most suited to the communication requirements of a task currently being executed during the session by the host computer. Any such dynamic change of the ISDN access path is invoked by the host computer, does not disrupt the host session and is substantially transparent to the user.

Abstract translation: 公开了一种与ISDN通信系统结合使用的技术，用于允许主机与用户正在执行主机会话并通过系统连接，以动态地改变将用户连接到主机的ISDN接入路径 计算机并且在会话发生期间在分组交换连接和电路交换连接之间携带主机会话，以便提供最适合于在会话期间当前正在执行的任务的通信要求的特定连接 主机。 主机计算机调用ISDN访问路径的任何此类动态变化，不会中断主机会话，并且对用户基本上是透明的。