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    • 2. 发明申请
    • Method and system for deterring SPam over Internet Protocol telephony and SPam Instant Messaging
    • 通过互联网协议电话和垃圾邮件即时消息来阻止垃圾邮件的方法和系统
    • US20070041372A1
    • 2007-02-22
    • US11203449
    • 2005-08-12
    • Anup RaoMatthew McRaeKendra HarringtonAllen Huotari
    • Anup RaoMatthew McRaeKendra HarringtonAllen Huotari
    • H04L12/66
    • H04L63/14H04L29/06027H04L51/12H04L65/1069H04L65/1079
    • Methods and systems for deterring spam on a communication medium are disclosed. Call traffic on the communication medium includes IP telephone calls, IM™ messages and other IP calls. The method includes detecting a call with a user device for accessing the communication medium, such as an IP telephone, a computer, etc. A signaling message associated with the call is accessed and compared with information relating to identities that correspond to originators of spam calls sent over said communication medium, which is stored in a repository associated with the communication medium, such as a dynamically updatable database. Where no match is found between the signaling message and the stored spam call identity information, the call is routed to the user. Where the signaling message matches any said spam call originator identity information however, the call is deterred from being routed to the user.
    • 公开了用于阻止通信介质上的垃圾邮件的方法和系统。 通信媒体上的呼叫业务包括IP电话呼叫,IM(TM)消息和其他IP呼叫。 该方法包括用用户设备检测用于访问诸如IP电话,计算机等的通信介质的呼叫。与该呼叫相关联的信令消息被访问并与与与垃圾邮件呼叫的发起者相对应的身份相关的信息进行比较 通过所述通信介质发送,所述通信介质存储在与通信介质相关联的存储库中,诸如动态可更新数据库。 在信令消息和存储的垃圾邮件呼叫身份信息之间没有匹配的情况下,呼叫被路由到用户。 在信令消息与任何所述垃圾呼叫发起者身份信息匹配的情况下,呼叫被阻止被路由到用户。
    • 6. 发明申请
    • Upstream data rate estimation
    • 上行数据速率估计
    • US20060256775A1
    • 2006-11-16
    • US11130333
    • 2005-05-16
    • Matthew McRaeAllen Huotari
    • Matthew McRaeAllen Huotari
    • H04L12/66
    • H04L29/06027H04L65/80
    • In one embodiment, a VoIP device operable to estimate an upstream data rate for a network device is provided. The VoIP device includes a transceiver operable to transmit VoIP packets to and receive VoIP packets from the network device; and a logic engine configured to initiate a series of simulated VoIP streams through the network device to a VoIP call destination, the logic engine being further configured to determine when at least one of the simulated VoIP streams in the series is unsuccessful, the logic engine being further configured to estimate the upstream data rate for the network device based upon a data rate for those simulated VoIP streams preceding the unsuccessful simulated VoIP stream.
    • 在一个实施例中,提供了可操作以估计网络设备的上行数据速率的VoIP设备。 所述VoIP设备包括:收发器,用于向所述网络设备发送VoIP分组并从所述网络设备接收VoIP分组; 以及逻辑引擎,被配置为启动通过所述网络设备到VoIP呼叫目的地的一系列模拟VoIP流,所述逻辑引擎还被配置为确定所述串联中的所述模拟VoIP流中的至少一个何时不成功,所述逻辑引擎是 还被配置为基于所述不成功的模拟VoIP流之前的那些模拟VoIP流的数据速率来估计所述网络设备的上行数据速率。
    • 9. 发明申请
    • Distributed codec for packet-based communications
    • 用于基于分组通信的分布式编解码器
    • US20070076697A1
    • 2007-04-05
    • US11243881
    • 2005-10-04
    • Allen HuotariKendra HarringtonMatthew McRae
    • Allen HuotariKendra HarringtonMatthew McRae
    • H04L12/66H04L12/56
    • H04L29/06027H04L12/2803H04L12/2834H04L12/2836H04L12/2838H04L65/1026H04L65/103H04L65/1036H04L65/104H04L2012/2841H04L2012/2845H04L2012/2849H04M7/0069
    • A packet-based communications system is provided, including: a client communications module for transmitting digital audio signals to a user interface device over a local area network (LAN) and for receiving digital audio signals from the user interface device, a voice packet module, and a network protocol module for transmitting data packets between the voice packet module and an Internet Protocol (IP) network. The voice packet module is configured to: receive audio signals from the client communications module and encapsulate the audio signals into data packets; convert data packets into audio signals and transmit the audio signals to the client communications module. A communications device is provided, including: a microphone; a speaker; a user input device; a network interface for coupling the communications device to a local area network (LAN); an audio processing module comprising an analog-to-digital converter (ADC) for converting analog audio signals from the microphone into a digital audio signal and a digital-to-analog converter (DAC) for converting a digital audio signal into an analog audio signal, said analog signal being output by the speaker; and an endpoint communications module for connecting to a telephony adapter on the LAN via the network interface, said endpoint communications module communicating digital audio signals with the telephony adapter over the LAN.
    • 提供了一种基于分组的通信系统,包括:客户端通信模块,用于通过局域网(LAN)向用户接口设备发送数字音频信号,并用于从用户接口设备接收数字音频信号,语音分组模块, 以及用于在语音分组模块和因特网协议(IP)网络之间传输数据分组的网络协议模块。 语音分组模块被配置为:从客户端通信模块接收音频信号,并将音频信号封装成数据分组; 将数据分组转换成音频信号,并将音频信号发送到客户端通信模块。 提供一种通信设备,包括:麦克风; 演讲者 用户输入设备; 用于将通信设备耦合到局域网(LAN)的网络接口; 音频处理模块,包括用于将来自麦克风的模拟音频信号转换为数字音频信号的模数转换器(ADC)和用于将数字音频信号转换成模拟音频信号的数模转换器(DAC) 所述模拟信号由扬声器输出; 以及端点通信模块,用于经由网络接口​​连接到LAN上的电话适配器,所述端点通信模块通过LAN与电话适配器通信数字音频信号。