公开(公告)号：US07551603B1

公开(公告)日：2009-06-23

申请号：US10765421

申请日：2004-01-27

Applicant: Shmuel Shaffer ,  Joseph F. Khouri ,  Michael E. Knappe

Inventor： Shmuel Shaffer ,  Joseph F. Khouri ,  Michael E. Knappe

IPC: H04L12/26 ,  H04J3/14 ,  G08C15/00 ,  G06F11/00 ,  G01R31/08

CPC classification number: H04L47/2441 ,  H04L47/2416 ,  H04L47/245 ,  H04L47/50

Abstract: A system and method are disclosed for interleaving time-critical packets and lower-priority packets onto a common data link. A packet arrival prediction mechanism predicts when a time-critical packet is expected to arrive. When transmission of a waiting lower-priority packet might cause a substantial delay in the expected time-critical packet's transmission, the lower-priority packet is parked until it can be transmitted without interfering with a time-critical packet.

Abstract translation: 公开了一种用于将时间关键分组和较低优先级分组交织到公共数据链路上的系统和方法。 分组到达预测机制预测何时预计到达时间关键分组。 当等待的较低优先级分组的传输可能导致预期的时间关键分组的传输中的实质性延迟时，优先级较低的分组被停放，直到其可以被传输而不干扰时间关键分组。

公开(公告)号：US07200214B2

公开(公告)日：2007-04-03

申请号：US11279983

申请日：2006-04-17

Applicant: Michael E. Knappe ,  Shmuel Shaffer

Inventor： Michael E. Knappe ,  Shmuel Shaffer

IPC: H04M3/42 ,  H04L12/16 ,  H04Q11/00 ,  G06F15/16

CPC classification number: H04L29/06027 ,  H04L12/1822 ,  H04L65/4038 ,  H04M3/56 ,  H04M3/564 ,  H04Q3/0025

Abstract: A method and system for participant control of privacy during a multiparty communication session includes receiving a request from a first participant to a multiparty communication connection for a sidebar between the first participant and a second participant to the multiparty communication connection. The sidebar is provided by at least substantially eliminating voice streams generated by the first participant and the second participant from conference output streams generated for a set of remaining participants to the multiparty communication connection.

Abstract translation: 一种用于在多方通信会话期间参与者控制隐私的方法和系统包括从第一参与者接收针对第一参与者和第二参与者之间的侧边栏的多方通信连接到多方通信连接的请求。 侧栏通过至少基本上消除由第一参与者和第二参与者产生的语音流从为一组剩余参与者生成的会议输出流提供到多方通信连接。

公开(公告)号：US07170886B1

公开(公告)日：2007-01-30

申请号：US09843596

申请日：2001-04-26

Applicant: Luke K. Surazski ,  Pascal Huart ,  Michael E. Knappe

Inventor： Luke K. Surazski ,  Pascal Huart ,  Michael E. Knappe

IPC: H04L12/66 ,  H04L12/28

CPC classification number: H04L29/06027 ,  H04L65/1083 ,  H04L65/403 ,  H04L65/605

Abstract: Devices, softwares and methods generate, in real time, indexing metadata for select portions of a telephone conversation or conference. The indexing metadata is generated responsive to inputs received while the conversation is being recorded live. The inputs are either by a user pressing a soft key on a telephone device, or by a voice conference bridge determining who is the dominant speaker in a multi-party conference.

Abstract translation: 设备，软件和方法实时地为电话对话或会议的选择部分生成索引元数据。 索引元数据是响应于在现场记录对话期间接收到的输入而生成的。 输入是通过用户按下电话设备上的软键，或通过语音会议桥来确定谁是多方会议中的主要扬声器。

公开(公告)号：US07099280B1

公开(公告)日：2006-08-29

申请号：US09821536

申请日：2001-03-28

Applicant: Shmuel Shaffer ,  Joseph F. Khouri ,  Michael E. Knappe ,  Robert T. Bell

Inventor： Shmuel Shaffer ,  Joseph F. Khouri ,  Michael E. Knappe ,  Robert T. Bell

IPC: G06F11/00

CPC classification number: H04M7/006 ,  G10L25/69 ,  H04M3/2236 ,  H04M3/42221 ,  H04M7/123 ,  H04M2201/36

Abstract: A method and system for logging voice quality issues for a communication connection includes receiving a signal for logging quality information for a voice connection at an endpoint of the voice connection. Voice samples are collected from the voice connection at the endpoint. The voice samples are stored in an error log at the endpoint.

Abstract translation: 用于记录通信连接的语音质量问题的方法和系统包括：在语音连接的端点处接收用于记录用于语音连接的质量信息的信号。 语音采样从终端的语音连接收集。 语音样本存储在端点的错误日志中。

公开(公告)号：US07002992B1

公开(公告)日：2006-02-21

申请号：US09802098

申请日：2001-03-07

Applicant: Shmuel Shaffer ,  Michael E. Knappe ,  Joseph F. Khouri ,  Robert T. Bell ,  Robert E. Gleichauf

Inventor： Shmuel Shaffer ,  Michael E. Knappe ,  Joseph F. Khouri ,  Robert T. Bell ,  Robert E. Gleichauf

IPC: H04J3/16

CPC classification number: H04M7/0072

Abstract: A communication system includes an endpoint that performs codec selection based on at least one network parameter. In a particular embodiment, a communication session exchanges voice information, and the codec selection improves the overall voice quality of the communication session.

Abstract translation: 通信系统包括基于至少一个网络参数执行编解码器选择的端点。 在特定实施例中，通信会话交换语音信息，并且编解码器选择改善了通信会话的总体语音质量。

公开(公告)号：US06904037B2

公开(公告)日：2005-06-07

申请号：US08743837

申请日：1996-11-05

Applicant: David R. Oran ,  Cary W. FitzGerald ,  Michael E. Knappe

Inventor： David R. Oran ,  Cary W. FitzGerald ,  Michael E. Knappe

IPC: H04L12/28 ,  H04L29/06 ,  H04L12/66

CPC classification number: H04L65/4076 ,  H04L12/5692 ,  H04L29/06 ,  H04L29/06027 ,  H04L65/607 ,  H04L69/22

Abstract: Voice and data streams transmitted from a conventional DSVD modem are interfaced directly to a network access server through a modified DSVD modem according to the invention. The voice and data is formatted into network data packets that are then routed directly to different endpoints through the network access server. The modified DSVD modem includes a packet framer that removes conventionally transmitted V.76 DSVD framing formats and stuffs bytes into the voice and data to form network packets. The network access server then routes the voice and data packets to the different endpoints identified in a packet header. Since the voice and data are output from the DSVD modem in data packets, the voice and data can be routed more efficiently to different network endpoints.

Abstract translation: 从传统DSVD调制解调器发送的语音和数据流通过根据本发明的经修改的DSVD调制解调器直接连接到网络接入服务器。 语音和数据被格式化成网络数据包，然后通过网络访问服务器直接路由到不同的端点。 经修改的DSVD调制解调器包括一个分组成帧器，其将常规传输的V.76 DSVD成帧格式和填充字节移除到语音和数据中以形成网络分组。 然后，网络接入服务器将语音和数据分组路由到分组报头中标识的不同端点。 由于语音和数据在数据分组中从DSVD调制解调器输出，所以语音和数据可以更有效地路由到不同的网络端点。

公开(公告)号：US06785267B1

公开(公告)日：2004-08-31

申请号：US09228031

申请日：1999-01-04

Applicant: Michael E. Knappe

Inventor： Michael E. Knappe

IPC: H04L1266

CPC classification number: H04L12/6418

Abstract: A telephone to digital data network interface 118 and method for its operation are disclosed. Interface 118 provides a connection point 52 and a standard line interface 50 for a POTS phone. Audio quality selector 86 interprets signaling received from a POTS phone (e.g., hookswitch signaling, DTMF signaling, or voice recognition commands) to select between multiple audio paths within interface 118. A first audio path includes the telephone line interface 50 and provides telephone-grade audio. Preferably, a second audio path includes a wideband audio output device (speaker 114) and a wideband audio input device (microphone 116). Audio quality selector 86 selects one of these paths using switches 106 and 108. It also communicates with digital audio processor 90, which preferably contains both a telephone-grade and a wideband audio codec and is capable of call-in-progress reconfiguration between the two. The present invention offers both a standard POTS phone interface and an option for higher-quality telephony when the other end of the conversation can provide this quality also.

Abstract translation: 公开了一种电话到数字数据网络接口118及其操作的方法。 接口118为POTS电话提供连接点52和标准线路接口50。 音频质量选择器86解释从POTS电话接收的信号（例如，钩开关信令，DTMF信令或语音识别命令）以在接口118内的多个音频路径之间进行选择。第一音频路径包括电话线接口50并提供电话级 音频。 优选地，第二音频路径包括宽带音频输出设备（扬声器114）和宽带音频输入设备（麦克风116）。 音频质量选择器86使用开关106和108来选择这些路径之一。它还与数字音频处理器90进行通信，数字音频处理器90优选地包含电话级和宽带音频编解码器，并且能够在两者之间进行呼叫进行中的重新配置 本发明提供标准POTS电话接口和用于更高质量电话的选项，当对话的另一端也可以提供该质量时。

公开(公告)号：US06603774B1

公开(公告)日：2003-08-05

申请号：US09169560

申请日：1998-10-09

Applicant: Michael E. Knappe ,  David R. Oran

Inventor： Michael E. Knappe ,  David R. Oran

IPC: H04J316

CPC classification number: H04M7/0072 ,  G10L19/167 ,  G10L19/173 ,  H04M7/006

Abstract: Voice packets are redirected in packet telephony applications to a codec proxy system that makes voice endpoints involved in an end-to-end call appear to be using the voice codec required of it by the other endpoint, even if the endpoints do not possess the required codec capability. The codec proxy system acts as a broker during initial capability negotiations, and as a real-time transcoding facility between disparate codec capabilities once voice traffic begins. The resulting system allows non-standard, cost-optimized and/or feature specific packet voice endpoints to interoperate in a standards-based network.

Abstract translation: 语音数据包在分组电话应用程序中被重定向到编解码器代理系统，使得端到端呼叫中涉及的语音端点似乎正在使用另一端点所需的语音编解码器，即使端点不具有所需的 编解码能力。 编解码器代理系统在初始功能协商期间充当代理，并且在语音流量开始之后作为不同编解码器功能之间的实时转码功能。 所得到的系统允许非标准，成本优化和/或特征特定分组语音端点在基于标准的网络中互操作。

公开(公告)号：US5485514A

公开(公告)日：1996-01-16

申请号：US220653

申请日：1994-03-31

Applicant: Michael E. Knappe ,  Brian R. Shelton

Inventor： Michael E. Knappe ,  Brian R. Shelton

IPC: H04M1/60 ,  H04R29/00 ,  H04S1/00 ,  H04S5/00 ,  H04M11/00

CPC classification number: H04S5/00 ,  H04M1/6016 ,  H04S7/30

Abstract: A telephone instrument creates spatially simulated sound signals from signals received from a telephone line. The received signals are directed to left and right channels. In each channel, the signals are processed via a direct path, an early reflection path including a finite impulse response filter and a reverberant decay path including all-pass filter. In each channel, the outputs from the three paths are summed with different weights.

Abstract translation: 电话乐器根据从电话线接收的信号产生空间模拟的声音信号。 接收的信号被引导到左右声道。 在每个通道中，通过直接路径，包括有限脉冲响应滤波器的早期反射路径和包括全通滤波器的混响衰减路径来处理信号。 在每个通道中，来自三个路径的输出与不同的权重相加。