公开(公告)号：US20060171324A1

公开(公告)日：2006-08-03

申请号：US11050338

申请日：2005-02-03

申请人： Satish Kumar Mundra ,  Daniel Thomas ,  David Lide

发明人： Satish Kumar Mundra ,  Daniel Thomas ,  David Lide

IPC分类号： H04L12/66 ,  H04L12/26

CPC分类号： H04M7/0072 ,  H04B14/06 ,  H04L29/06027 ,  H04L65/1026 ,  H04L65/1036 ,  H04L65/1043 ,  H04L65/605

摘要： Providing interoperability on a voice communication network by detecting incompatible bit packing payload formats of ADPCM encoded voice data signals in an encoder/decoder. A mismatch of bit packing formats between little endian format and big endian format is determined through the use of a media gateway control protocol package called Real Time Protocol that allows a Call Agent to detect the presence of garbled audio and allows an intelligent gateway to detect the payload format being used by remote endpoints during call setup procedures and then switch the gateway's encoding for the payload to the payload format being used by the remote endpoint. The invention may be applied to an ITU G.726 encoder/decoder.

摘要翻译： 通过检测编码器/解码器中ADPCM编码语音数据信号的不兼容的位打包有效载荷格式，在语音通信网络上提供互操作性。 通过使用称为实时协议的媒体网关控制协议包来确定小端格式和大端格式之间的位打包格式的不匹配，该协议包允许呼叫代理检测到混乱音频的存在，并允许智能网关检测 有效载荷格式在呼叫建立过程期间由远程端点使用，然后将网关的有效负载编码切换到由远程端点使用的有效载荷格式。 本发明可以应用于ITU G.726编码器/解码器。

公开(公告)号：US20060171372A1

公开(公告)日：2006-08-03

申请号：US11047746

申请日：2005-02-02

申请人： Satish Kumar Mundra ,  Daniel Thomas

发明人： Satish Kumar Mundra ,  Daniel Thomas

IPC分类号： H04L12/66

CPC分类号： H04L65/605 ,  H04L29/06027 ,  H04L65/103 ,  H04L65/104 ,  H04L69/22

摘要： Providing interoperability on a voice communication network by detecting incompatible bit packing payload formats of ADPCM encoded voice data signals in an encoder/decoder. A mismatch of bit packing formats between little endian format and big endian format is determined by detecting the occurrence of ADPCM code words of interest in the decoded voice data stream out of the encoder/decoder. The invention may be applied to an ITU G.726 encoder/decoder.

摘要翻译： 通过检测编码器/解码器中ADPCM编码语音数据信号的不兼容的位打包有效载荷格式，在语音通信网络上提供互操作性。 通过检测编码器/解码器中解码的语音数据流中感兴趣的ADPCM码字的出现来确定小端格式和大端格式之间的比特打包格式的不匹配。 本发明可以应用于ITU G.726编码器/解码器。

公开(公告)号：US20060072522A1

公开(公告)日：2006-04-06

申请号：US10953683

申请日：2004-09-29

申请人： Praphul Chandra ,  David Lide ,  Satish Kumar Mundra ,  Manoj Sindhwani

发明人： Praphul Chandra ,  David Lide ,  Satish Kumar Mundra ,  Manoj Sindhwani

IPC分类号： H04Q7/24

CPC分类号： H04L29/06027 ,  H04L65/1069 ,  H04L65/4007 ,  H04L65/80 ,  H04W28/02 ,  H04W48/06 ,  H04W72/1242 ,  H04W84/12

摘要： A method for transmitting voice data traffic over a wireless medium using a decision process for the admission of a voice call, such as a call voice over Internet Protocol (VoIP) over a wireless local area network (WLAN). Admission of the call through the network is based upon the priority level of the call and a congestion level of the WLAN in order to increase the quality and the number of calls on a BSS. Paramters such as codec and packetization periods are selected for the call based upon the priority and congestion level. As the level of congestion in the network Increases, only calls above a certain Priority level are allowed into the network.

摘要翻译： 一种用于通过无线局域网（WLAN）的诸如通过因特网协议（VoIP）的呼叫语音接入语音呼叫的决定过程在无线介质上发送语音数据业务的方法。 通过网络接纳呼叫是基于呼叫的优先级和WLAN的拥塞级别，以便提高BSS上的呼叫的质量和数量。 基于优先级和拥塞级别，为呼叫选择编解码器和分组化周期等参数。 随着网络拥塞程度的增加，只有在某一优先级以上的呼叫被允许进入网络。

公开(公告)号：US20120033676A1

公开(公告)日：2012-02-09

申请号：US12849419

申请日：2010-08-03

申请人： Satish Kumar MUNDRA ,  Tinku MANNAN

发明人： Satish Kumar MUNDRA ,  Tinku MANNAN

IPC分类号： H04L12/56

CPC分类号： H04L65/1043 ,  G06F1/3212 ,  H04L65/102 ,  Y02D10/174

摘要： In one embodiment, a gateway includes an interface, and a processor cooperatively operable with the interface to transmit and receive packet communications. The processor receives, over the interface, a request-notification for a backup battery status, which is formatted according to a media gateway control protocol (MGCP) package protocol. The processor transmits, over the interface, a notify of an observed event, the observed event indicating the backup battery status which is formatted according to the MGCP package protocol. In another embodiment, a call agent includes an interface and a processor. The call agent processor transmits a request-notification for a backup battery status, which is formatted according to a media gateway control protocol (MGCP) package protocol. The call agent processor also receives a notify of an observed event over the interface, the observed event indicating the backup battery status, which is formatted according to the MGCP package protocol.

摘要翻译： 在一个实施例中，网关包括接口，以及与接口协作地可操作以发送和接收分组通信的处理器。 处理器通过接口接收根据媒体网关控制协议（MGCP）包协议格式化的备用电池状态的请求通知。 处理器通过接口发送观察事件的通知，观察到的事件指示根据MGCP包协议格式化的备用电池状态。 在另一个实施例中，呼叫代理包括接口和处理器。 呼叫代理处理器发送根据媒体网关控制协议（MGCP）包协议格式化的备用电池状态的请求通知。 呼叫代理处理器还通过接口接收观察到的事件的通知，观察到的事件指示备用电池状态，其根据MGCP包协议被格式化。

公开(公告)号：US20050190756A1

公开(公告)日：2005-09-01

申请号：US10788091

申请日：2004-02-26

申请人： Satish Kumar Mundra ,  David Lide

发明人： Satish Kumar Mundra ,  David Lide

IPC分类号： H04L12/56 ,  H04L29/06

CPC分类号： H04L29/06027 ,  H04L65/1096 ,  H04L65/607 ,  H04L65/608

摘要： A method for defining a payload format in Real Time Protocol (RTP) messages for voice band data transmission to be transmitted over a packet network to an Internet Protocol (IP) telephony endpoint, such as an IP phone. The invention extends RFC 2833 standards for Real Time Protocol (RTP) message payload format to include voice data transmission. The invention includes receiving voice transmission signals containing a voice band data from the public switched telephone network into a packet network and converting said voice transmission signals into data packets using real time protocol (RTP) and assigning an RTP event code in the RTP payload format for the voice band data applications. Applications include Caller Identification (Caller-ID), Visual Message Waiting Indication (VMWI), and Advanced Digital Subscriber Interface (ADSI).

摘要翻译： 用于定义用于语音频带数据传输的实时协议（RTP）消息中的有效载荷格式的方法，其通过分组网络传送到诸如IP电话的因特网协议（IP）电话端点。 本发明扩展了用于实时协议（RTP）消息有效载荷格式的RFC 2833标准，以包括语音数据传输。 本发明包括将包含来自公共交换电话网络的语音频带数据的语音传输信号接收到分组网络中，并使用实时协议（RTP）将所述语音传输信号转换成数据分组，并将RTP有效载荷格式的RTP事件代码分配给 语音数据应用。 应用程序包括来电显示（来电显示），可视消息等待指示（VMWI）和高级数字用户接口（ADSI）。