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1.发明申请TECHNIQUE FOR PROVIDING TRANSLATION BETWEEN THE PACKET ENVIRONMENT AND THE PSTN ENVIRONMENT 审中-公开提供分组环境与PSTN环境之间的转换的技术公开(公告)号：US20100220715A1
公开(公告)日：2010-09-02
申请号：US12632728
申请日：2009-12-07
申请人： Ali Cherchali , James Ehlinger , Paul J. Fellingham , Marius Jonas Gudelis , Steven M. Michelson , James Yatsko
发明人： Ali Cherchali , James Ehlinger , Paul J. Fellingham , Marius Jonas Gudelis , Steven M. Michelson , James Yatsko
IPC分类号： H04L12/66
CPC分类号： H04L12/66
摘要： Voice over Internet Protocol (VoIP) calls received in a Hybrid Fiber Coax (HFC) network (12) maintained by a provider of HFC VoIP telephony services are advantageously translated into a Time-Division Multiplex (TDM) format by an Internet Protocol Device Terminal (IPDT) (26) in the HFC network. Once translated into a TDM format, the call passes to the Public Switched Telephone Network (PSTN) (28) for call processing to afford the call features subscribed to by the called party, such as caller-ID and call waiting. Once processed, the PSTN routes the call to the destination. Likewise, a call destined for an HFC VoIP customer can be processed within the PSTN to afford the call features subscribed to by the HFC VoIP customer. In this way, the HFC VoIP telephony service can offer full-featured VoIP cable telephony without the need to perform call processing in its own network.
摘要翻译： 由HFC VoIP电话服务提供商维护的混合光纤同轴（HFC）网络（12）中接收到的因特网协议语音（VoIP）呼叫有利地被互联网协议设备终端（TDM）转换为时分复用（TDM）格式 IPDT）（26）。 一旦转换成TDM格式，呼叫转到公共交换电话网（PSTN）（28）进行呼叫处理，以承担被叫方订阅的呼叫功能，如呼叫者ID和呼叫等待。 一旦处理，PSTN将呼叫路由到目的地。 同样，可以在PSTN内处理发往HFC VoIP客户的呼叫，以承担由HFC VoIP客户订购的呼叫功能。 以这种方式，HFC VoIP电话服务可以提供全功能的VoIP电话电话，而无需在其自己的网络中执行呼叫处理。
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2.发明授权Technique for providing translation between the packet environment and the PSTN environment 有权在分组环境和PSTN环境之间提供翻译的技术公开(公告)号：US07630359B1
公开(公告)日：2009-12-08
申请号：US09966492
申请日：2001-09-28
申请人： Ali Cherchali , James Ehlinger , Paul J Fellingham , Marius Jonas Gudelis , Steven M Michelson , James Yatsko
发明人： Ali Cherchali , James Ehlinger , Paul J Fellingham , Marius Jonas Gudelis , Steven M Michelson , James Yatsko
IPC分类号： H04L12/66
CPC分类号： H04L12/66
摘要： Voice over Internet Protocol (VoIP) calls received in a Hybrid Fiber Coax (HFC) network (12) maintained by a provider of HFC VoIP telephony services are advantageously translated into a Time-Division Multiplex (TDM) format by an Internet Protocol Device Terminal (IPDT) (26) in the HFC network. Once translated into a TDM format, the call passes to the Public Switched Telephone Network (PSTN) (28) for call processing to afford the call features subscribed to by the called party, such as caller-ID and call waiting. Once processed, the PSTN routes the call to the destination. Likewise, a call destined for an HFC VoIP customer can be processed within the PSTN to afford the call features subscribed to by the HFC VoIP customer. In this way, the HFC VoIP telephony service can offer full-featured VoIP cable telephony without the need to perform call processing in its own network.
摘要翻译： 由HFC VoIP电话服务提供商维护的混合光纤同轴（HFC）网络（12）中接收到的因特网协议语音（VoIP）呼叫有利地被互联网协议设备终端（TDM）转换为时分复用（TDM）格式 IPDT）（26）。 一旦转换成TDM格式，呼叫转到公共交换电话网（PSTN）（28）进行呼叫处理，以承担被叫方订阅的呼叫功能，如呼叫者ID和呼叫等待。 一旦处理，PSTN将呼叫路由到目的地。 同样，可以在PSTN内处理发往HFC VoIP客户的呼叫，以承担由HFC VoIP客户订购的呼叫功能。 以这种方式，HFC VoIP电话服务可以提供全功能的VoIP电话电话，而无需在其自己的网络中执行呼叫处理。
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公开(公告)号：US20090201920A1
公开(公告)日：2009-08-13
申请号：US12386148
申请日：2009-04-14
IPC分类号： H04L12/66
CPC分类号： H04L12/66
摘要： An improved telephony adapter compresses voice data, creates IP packets, and prioritizes the voice IP packets over the data IP packets. Preferably, the compression and packetization interval is such that the bandwidth occupied by the voice IP packets is approximately half of the minimum average available bandwidth in the upstream direction, thereby maintaining acceptable latency and voice quality of the speech. Further enhancement is achieved by causing the ISP to also give priority to voice packets that are destined to the telephony adapter, over the data packets that are destined to the telephony adapter.
摘要翻译： 改进的电话适配器压缩语音数据，创建IP分组，并通过数据IP分组对语音IP分组进行优先级排序。 优选地，压缩和打包间隔使得语音IP分组占用的带宽大约是上行方向上的最小平均可用带宽的一半，从而保持语音的可接受的等待时间和语音质量。 通过使ISP还优先考虑目的地为电话适配器的语音分组，通过去往电话适配器的数据分组来实现进一步的增强。
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公开(公告)号：US08526424B2
公开(公告)日：2013-09-03
申请号：US12386148
申请日：2009-04-14
IPC分类号： H04L12/66
CPC分类号： H04L12/66
摘要： An improved telephony adapter compresses voice data, creates IP packets, and prioritizes the voice IP packets over the data IP packets. Preferably, the compression and packetization interval is such that the bandwidth occupied by the voice IP packets is approximately half of the minimum average available bandwidth in the upstream direction, thereby maintaining acceptable latency and voice quality of the speech. Further enhancement is achieved by causing the ISP to also give priority to voice packets that are destined to the telephony adapter, over the data packets that are destined to the telephony adapter.
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5.发明申请Method and Apparatus for Controlling the Quality of Service of Voice and Data Services Over Variable Bandwidth Access Networks 有权用于控制可变带宽接入网络中语音和数据业务服务质量的方法和装置公开(公告)号：US20090290579A1
公开(公告)日：2009-11-26
申请号：US12535162
申请日：2009-08-04
IPC分类号： H04L12/66
CPC分类号： H04L47/10 , H04L47/2416 , H04L47/2441 , H04L47/245 , H04L47/29 , H04L47/30 , H04L47/32
摘要： A terminal adapter for guaranteeing the quality of service of both voice and data packets is disclosed. When a data packet is received in a first data input queue of a terminal adapter, a determination is made whether a voice packet is present in a voice input queue. Another determination is made as to whether the sum of the size of the data packet and the size of all packets in a terminal adapter output queue would exceed a first size threshold established for the output queue. If voice packets are present in the voice input queue, or if the aforementioned sum exceeds the size threshold, the data packet is not forwarded to the output queue. If no voice packets are present in the voice input queue and if the aforementioned sum is below the first size threshold, then the data packet is forwarded to the output queue.
摘要翻译： 公开了用于保证语音和数据分组的服务质量的终端适配器。 当在终端适配器的第一数据输入队列中接收到数据分组时，确定语音分组是否存在于语音输入队列中。 另外确定数据分组的大小和终端适配器输出队列中的所有分组的大小之和是否超过为输出队列建立的第一大小阈值。 如果语音输入队列中存在语音分组，或者如果上述的和超过了大小阈值，则数据分组不会转发到输出队列。 如果语音输入队列中没有语音分组，并且如果上述和低于第一大小阈值，则将数据分组转发到输出队列。
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6.发明授权Method and apparatus for controlling the quality of service of voice and data services over variable bandwidth access networks 有权用于通过可变带宽接入网络控制语音和数据业务的服务质量的方法和装置公开(公告)号：US07590058B1
公开(公告)日：2009-09-15
申请号：US11000677
申请日：2004-12-01
IPC分类号： G01R31/08
CPC分类号： H04L47/10 , H04L47/2416 , H04L47/2441 , H04L47/245 , H04L47/29 , H04L47/30 , H04L47/32
摘要： A terminal adapter for guaranteeing the quality of service of both voice and data packets is disclosed. When a data packet is received in a first data input queue of a terminal adapter, a determination is made whether a voice packet is present in a voice input queue. Another determination is made as to whether the sum of the size of the data packet and the size of all packets in a terminal adapter output queue would exceed a first size threshold established for the output queue. If voice packets are present in the voice input queue, or if the aforementioned sum exceeds the size threshold, the data packet is not forwarded to the output queue. If no voice packets are present in the voice input queue and if the aforementioned sum is below the first size threshold, then the data packet is forwarded to the output queue.
摘要翻译： 公开了用于保证语音和数据分组的服务质量的终端适配器。 当在终端适配器的第一数据输入队列中接收到数据分组时，确定语音分组是否存在于语音输入队列中。 另外确定数据分组的大小和终端适配器输出队列中的所有分组的大小之和是否超过为输出队列建立的第一大小阈值。 如果语音输入队列中存在语音分组，或者如果上述的和超过了大小阈值，则数据分组不会转发到输出队列。 如果语音输入队列中没有语音分组，并且如果上述和低于第一大小阈值，则将数据分组转发到输出队列。
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7.发明授权Method and apparatus for controlling the quality of service of voice and data services over variable bandwidth access networks 有权用于通过可变带宽接入网络控制语音和数据业务的服务质量的方法和装置公开(公告)号：US07545745B1
公开(公告)日：2009-06-09
申请号：US11036861
申请日：2005-01-14
申请人： Ali M. Cherchali , Gagan Lal Choudhury , Gerald Murray Ezrol , Marius Jonas Gudelis , Thomas Joseph Killian , Jerry A. Leger , Norman L. Schryer
发明人： Ali M. Cherchali , Gagan Lal Choudhury , Gerald Murray Ezrol , Marius Jonas Gudelis , Thomas Joseph Killian , Jerry A. Leger , Norman L. Schryer
IPC分类号： G08C15/00
CPC分类号： H04L43/0882 , H04L41/5003 , H04L41/5087 , H04L47/225 , H04L47/2416 , H04L47/32
摘要： A terminal adapter for guaranteeing the quality of service of both voice and data packets is disclosed. Such quality is ensured by inserting gaps between successive data packets in a stream of multiplexed data and/or voice packets. A gap after a particular data packet is proportional to the size of that particular data packet. In this way, bandwidth is preserved for any voice packets that may have arrived during the transfer of the data packet as well as for any voice packets that arrive during the gap. The unconstrained upstream data bandwidth and the bandwidth used by voice calls may each be estimated by taking a plurality of instantaneous measurements of the available bandwidth and/or taking individual direct measurements. The size of data packets may be limited to a maximum size in order to ensure that time-sensitive voice packets experience only an acceptable delay in queue for transmission.
摘要翻译： 公开了用于保证语音和数据分组的服务质量的终端适配器。 通过在多路复用数据流和/或语音分组流中的连续数据分组之间插入间隙来确保这样的质量。 特定数据包之后的间隙与该特定数据包的大小成比例。 以这种方式，对于可能在数据分组的传输期间以及在间隙期间到达的任何语音分组可能已经到达的任何语音分组来保留带宽。 每个可以通过对可用带宽进行多个瞬时测量和/或进行各个直接测量来估计无约束上行数据带宽和语音呼叫所使用的带宽。 数据分组的大小可以被限制为最大大小，以便确保时间敏感的语音分组在传输队列中仅经历可接受的延迟。
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8.发明授权Method and apparatus for controlling the quality of service of voice and data services over variable bandwidth access networks 有权用于通过可变带宽接入网络控制语音和数据业务的服务质量的方法和装置公开(公告)号：US07953005B2
公开(公告)日：2011-05-31
申请号：US12535162
申请日：2009-08-04
IPC分类号： H04L12/26
CPC分类号： H04L47/10 , H04L47/2416 , H04L47/2441 , H04L47/245 , H04L47/29 , H04L47/30 , H04L47/32
摘要： A terminal adapter for guaranteeing the quality of service of both voice and data packets is disclosed. When a data packet is received in a first data input queue of a terminal adapter, a determination is made whether a voice packet is present in a voice input queue. Another determination is made as to whether the sum of the size of the data packet and the size of all packets in a terminal adapter output queue would exceed a first size threshold established for the output queue. If voice packets are present in the voice input queue, or if the aforementioned sum exceeds the size threshold, the data packet is not forwarded to the output queue. If no voice packets are present in the voice input queue and if the aforementioned sum is below the first size threshold, then the data packet is forwarded to the output queue.
摘要翻译： 公开了用于保证语音和数据分组的服务质量的终端适配器。 当在终端适配器的第一数据输入队列中接收到数据分组时，确定语音分组是否存在于语音输入队列中。 另外确定数据分组的大小和终端适配器输出队列中的所有分组的大小之和是否超过为输出队列建立的第一大小阈值。 如果语音输入队列中存在语音分组，或者如果上述的和超过了大小阈值，则数据分组不会转发到输出队列。 如果语音输入队列中没有语音分组，并且如果上述和低于第一大小阈值，则将数据分组转发到输出队列。
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公开(公告)号：US07613282B1
公开(公告)日：2009-11-03
申请号：US10411533
申请日：2003-04-10
CPC分类号： H04L12/66
摘要： An improved telephony adapter compresses voice data, creates IP packets, and prioritizes the voice IP packets over the data IP packets. Preferably, the compression and packetization interval is such that the bandwidth occupied by the voice IP packets is approximately half of the minimum average available bandwidth in the upstream direction, thereby maintaining acceptable latency and voice quality of the speech. Further enhancement is achieved by causing the ISP to also give priority to voice packets that are destined to the telephony adapter, over the data packets that are destined to the telephony adapter.
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公开(公告)号：US06847704B1
公开(公告)日：2005-01-25
申请号：US10328577
申请日：2002-12-23
申请人： Ali M. Cherchali , James C. Ehlinger , Marius Jonas Gudelis , William G. Lester , Robert J. McLaughlin
发明人： Ali M. Cherchali , James C. Ehlinger , Marius Jonas Gudelis , William G. Lester , Robert J. McLaughlin
CPC分类号： H04M3/42153 , H04M3/005 , H04M3/42297 , H04M7/1255 , H04M7/128
摘要： An architecture and technique for creating self-installable and portable telephony (dial tone) service that can be moved between any two locations that has access to both a voice communication network and a data network. A telephony adapter is used as a subscriber premises device that is connected between a conventional telephone set and both a voice network and a data network. A provisioning server communicates with the telephony adapter through the data network and maintains a record of the subscriber's local telephone number and IP address of the telephony adapter. As the subscriber moves from one location to another, the telephony adapter (once turned “on”) will communicate with the provisioning server and re-establish phone service, always using the same local phone number of the subscriber.
摘要翻译： 一种用于创建可安装和便携式电话（拨号音）服务的架构和技术，可以在可访问语音通信网络和数据网络的任何两个位置之间移动。 电话适配器用作连接在常规电话机与语音网络和数据网络两者之间的用户驻地设备。 配置服务器通过数据网络与电话适配器通信，并维护用户的本地电话号码和电话适配器的IP地址的记录。 当用户从一个位置移动到另一个位置时，电话适配器（一旦打开）将与配置服务器进行通信，并重新建立电话服务，总是使用订户的相同的本地电话号码。
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