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1. 发明申请

US20050074012A1 Adaptive call admission control for calls handled over a compressed clear channel 有权
标题翻译：通过压缩清除信道处理的呼叫的自适应呼叫准入控制
公开(公告)号：US20050074012A1
公开(公告)日：2005-04-07
申请号：US10680601
申请日：2003-10-06
申请人： Mehryar Garakani , Prasad Miriyala , Anantha Mekala , Jianping Huang
发明人： Mehryar Garakani , Prasad Miriyala , Anantha Mekala , Jianping Huang
IPC分类号： H04L12/56 , H04W24/00 , H04W28/06 , H04W28/10 , H04W28/20 , H04W48/06 , H04W72/04 , H04W74/00
CPC分类号： H04L47/15 , H04L47/11 , H04L47/14 , H04L47/70 , H04L47/745 , H04L47/801 , H04L47/805 , H04L47/822 , H04L47/824 , H04W24/00 , H04W28/06 , H04W28/10 , H04W28/20 , H04W48/06 , H04W72/0486 , H04W74/00
摘要： A system that packetizes telephone calls using loss-less codecs for transmission over a communication link. The system includes a call admission control mechanism that admits calls for transmission over the communication link. An adaptive system which monitors bandwidth usage and which dynamically provides a feedback loop to the CAC system. The bandwidth allocation for transmission of the calls and the admission of calls is based upon actual bandwidth usage conditions in the system.
摘要翻译：一种使用无损编解码器通过通信链路传输来打电话的系统。该系统包括允许通过通信链路传输的呼叫的呼叫许可控制机制。一种自适应系统，用于监控带宽使用情况，并为CAC系统动态提供反馈回路。用于传输呼叫和接入呼叫的带宽分配基于系统中的实际带宽使用条件。

2. 发明申请

US20070019581A1 Technique for transferring data from a time division multiplexing network onto a packet network 有权
标题翻译：用于将数据从时分复用网络传送到分组网络的技术
公开(公告)号：US20070019581A1
公开(公告)日：2007-01-25
申请号：US11184483
申请日：2005-07-19
申请人： Anantha Mekala , Mehryar Garakani
发明人： Anantha Mekala , Mehryar Garakani
IPC分类号： H04B7/212 , H04Q7/00
CPC分类号： H04M7/1205
摘要： A technique for processing data carried by a Time Division Multiplexing (TDM) channel in a TDM network. A data stream associated with a TDM channel is monitored to determine if it contains data frames. If so, the channel associated with the data stream enters a clear channel mode which causes data in the data stream to be placed in packets for transfer over the packet network without being modified. Provided the data stream continues to carry data frames, the data stream stays in clear channel mode. After the data stream data no longer carries data frames, the channel exits clear channel mode.
摘要翻译：用于处理TDM网络中的时分复用（TDM）信道携带的数据的技术。监视与TDM信道相关联的数据流，以确定其是否包含数据帧。如果是这样，则与数据流相关联的信道进入清除信道模式，这导致数据流中的数据被放置在分组中以便在分组网络上传送而不被修改。如果数据流继续携带数据帧，则数据流保持在清除信道模式。在数据流数据不再携带数据帧之后，通道退出清除通道模式。

3. 发明申请

US20060120347A1 Voice over internet protocol (VOIP) subcell multiplexing 有权
标题翻译：语音互联网协议（VOIP）子信元复用
公开(公告)号：US20060120347A1
公开(公告)日：2006-06-08
申请号：US11003636
申请日：2004-12-03
申请人： Anantha Mekala
发明人： Anantha Mekala
IPC分类号： H04L12/66 , H04L12/56
CPC分类号： H04M7/125 , H04L12/66 , H04L29/06 , H04L65/1026 , H04L65/103 , H04L69/08
摘要： A communication device such as a Voice over Internet Protocol (VoIP) gateway multiplexes data intended for multiple voice connections within a single IP packet. If it is known in advance that packets for multiple connections between a given Originating Exchange (OEX) and Terminating Exchange (TEX) will travel between the Originating Gateway and Terminating Gateway, voice samples are multiplexed into the same VoIP packet. This “cell multiplexing” is accomplished by adding a cell header field to each cell payload portion. The cell header field indicates at least a connection identifier, so that the terminating gateway can route the payload to the correct TEX trunk. The scheme permits greatly improved efficiency in the carrying of VoIP traffic, especially where efficient voice coders are used.
摘要翻译：诸如互联网语音协议（VoIP）网关之类的通信设备复用旨在用于单个IP分组内的多个语音连接的数据。如果预先知道给定始发交换机（OEX）和终止交换机（TEX）之间的多个连接的分组将在始发网关和终止网关之间传播，则语音样本被复用到相同的VoIP分组中。这种“小区多路复用”是通过向每个小区有效负载部分添加小区头部字段来实现的。小区报头字段至少指示一个连接标识符，使得终端网关可以将有效载荷路由到正确的TEX中继线。该方案允许大大提高VoIP业务携带的效率，特别是在使用高效语音编码器的情况下。

4. 发明申请

US20060268832A1 Transporting synchronization channel information across a packet network 有权
公开(公告)号：US20060268832A1
公开(公告)日：2006-11-30
申请号：US11138599
申请日：2005-05-26
申请人： Anantha Mekala , George Ogagan
发明人： Anantha Mekala , George Ogagan
IPC分类号： H04L12/66 , H04L12/56 , H04J3/06
CPC分类号： H04Q3/0045 , H04J3/14 , H04J3/1623 , H04L12/4633 , H04L49/90 , H04L49/9042
摘要： A technique for preserving information contained in a synchronization channel of a Time Division Multiplexing (TDM) frame across a packet network. Information contained in the synchronization channel of TDM frames is transferred over the packet network by a first gateway device that received the TDM frames. A second gateway device receives the synchronization channel information and places the information in one or more TDM frames. The TDM frames are transferred onto the TDM network.

5. 发明申请

US20060083220A1 Transport of DTMF tones over VOATM/VOIP networks 有权
公开(公告)号：US20060083220A1
公开(公告)日：2006-04-20
申请号：US10994559
申请日：2004-11-22
申请人： Anantha Mekala , Subrahmanyam Kunapuli , Jason Kuhne , Salman Haider
发明人： Anantha Mekala , Subrahmanyam Kunapuli , Jason Kuhne , Salman Haider
IPC分类号： H04L12/66 , H04L12/16 , H04Q11/00
CPC分类号： H04Q11/04 , H04L12/66 , H04L65/1026 , H04L65/103 , H04M7/1295 , H04Q2213/13034 , H04Q2213/13196 , H04Q2213/13214 , H04Q2213/13216 , H04Q2213/1329 , H04Q2213/13296 , H04Q2213/13389 , H04Q2213/13405
摘要： A communication device, such as a Voice over Internet Protocol (VoIP) gateway, determines a duration for Dual Tone Multi-Frequency (DTMF) tone portions of telephony signal. If the duration is less than a pre-determined amount, a minimum duration is enforced during DTMF playback at a remote end of a network connection connected to a destination gateway. Minimum playback duration can be enforced at the terminating gateway—however, the originatinggateway can also encode a DTMF packet with a minimum duration value. At the terminating receiver it is not always possible to playback exactly what happened at the originating point in the same time frame. One solution to is to, at the terminating gateway, drop the first portion of voice packets that overlap with the end portion of played back DTMF tones.

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