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Anantha R Mekala

from Shrewsbury, MA
Age ~57

Anantha Mekala Phones & Addresses

  • 9 Birch Brush Rd, Shrewsbury, MA 01545 (508) 845-6618
  • 122 Water St, Leominster, MA 01453 (978) 534-1109
  • Nashua, NH
  • 20 River Rd, Manchester, NH 03104 (603) 623-4015
  • 9 Birch Brush Rd, Shrewsbury, MA 01545

Work

Company: Cisco systems, inc 2013 Position: Senior technical leader

Education

School / High School: George Washington University 2005 Specialities: Master in Project Management

Public records

Vehicle Records

Anantha Mekala

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Address:
9 Birch Brush Rd, Shrewsbury, MA 01545
Phone:
(508) 845-6618
VIN:
5TDZK23C07S015933
Make:
TOYOTA
Model:
SIENNA
Year:
2007

Resumes

Resumes

Anantha Mekala Photo 1

Anantha Mekala Shrewsbury, MA

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Work:
CISCO SYSTEMS, INC

2013 to 2000
Senior Technical Leader

CISCO SYSTEMS, INC

1998 to 2000

CISCO SYSTEMS, INC

2007 to 2012
Technical leader

CISCO SYSTEMS, INC

1998 to 2006
Software Engineer

Alcatel Europe

1993 to 1997
Software Engineer

Indian Telephone Industries Ltd
Bangalore, Karnataka
1988 to 1993
Software Engineer

Education:
George Washington University
2005
Master in Project Management

S.V University
1988
M.TECH in Electronic Communications

Publications

Us Patents

Transport Of Dtmf Tones Over Voatm/Voip Networks

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US Patent:
7486665, Feb 3, 2009
Filed:
Nov 22, 2004
Appl. No.:
10/994559
Inventors:
Anantha R. Mekala - Leominster MA,
Subrahmanyam V. Kunapuli - Cupertino CA,
Jason A. Kuhne - Acton MA,
Salman Haider - Nashua NH,
Assignee:
Cisco Technology, Inc. - San Jose CA
International Classification:
H04L 12/66
H04J 3/12
US Classification:
370352, 370526, 379 9326
Abstract:
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.

Transporting Synchronization Channel Information Across A Packet Network

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US Patent:
7627004, Dec 1, 2009
Filed:
May 26, 2005
Appl. No.:
11/138599
Inventors:
Anantha R. Mekala - Leominster MA,
George O. Ogagan - Nashua NH,
Assignee:
CISCO Technology, Inc. - San Jose CA
International Classification:
H04J 3/06
US Classification:
370509
Abstract:
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.

Voice Over Internet Protocol (Voip) Subcell Multiplexing

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US Patent:
7680105, Mar 16, 2010
Filed:
Dec 3, 2004
Appl. No.:
11/003636
Inventors:
Anantha R. Mekala - Leominster MA,
Assignee:
Cisco Technology, Inc. - San Jose CA
International Classification:
H04L 12/56
US Classification:
370389
Abstract:
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.

Technique For Transferring Data From A Time Division Multiplexing Network Onto A Packet Network

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US Patent:
7729332, Jun 1, 2010
Filed:
Jul 19, 2005
Appl. No.:
11/184483
Inventors:
Anantha R. Mekala - Leominster MA,
Mehryar K. Garakani - Westlake Village CA,
Assignee:
Cisco Technology, Inc. - San Jose CA
International Classification:
H04B 7/212
US Classification:
370347, 370474
Abstract:
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.

Generating And Signaling Tones In A Communications Network

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US Patent:
7881712, Feb 1, 2011
Filed:
Apr 25, 2005
Appl. No.:
11/113679
Inventors:
Chengsheng Luo - Chelmsford MA,
David A. Houghton - Danville NH,
Anantha R. Mekala - Leominster MA,
Assignee:
Cisco Technology, Inc. - San Jose CA
International Classification:
H04W 4/00
US Classification:
4554221, 4554141, 455433, 455450, 455560, 370352, 370390, 370392, 370396, 370464, 37922102, 37940601
Abstract:
A technique for providing a flexible approach to generating and signaling tones, such as call progress tones, in a communications network is described. Parameters associated with tones generated and signaled in a communication network are dynamically configured. A tone to be generated and signaled onto the network is specified in a request that identifies the tone. Parameters associated with the specified tone are identified. The information representing the tone is then generated in accordance with the identified parameters and signaled onto the communications network.

Adaptive Call Admission Control For Calls Handled Over A Compressed Clear Channel

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US Patent:
7369559, May 6, 2008
Filed:
Oct 6, 2003
Appl. No.:
10/680601
Inventors:
Mehryar Khalili Garakani - Westlake Village CA,
Prasad Miriyala - Union City CA,
Anantha R. Mekala - Leominster MA,
Jianping Huang - San Jose CA,
Assignee:
Cisco Technology Inc. - San Jose CA
International Classification:
H04L 12/28
H04L 12/56
H04L 12/26
H04J 3/18
G06F 15/16
US Classification:
3703952, 370252, 370468, 370477, 370521, 709205, 709247
Abstract:
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.
Anantha R Mekala from Shrewsbury, MA, age ~57 Get Report