Abstract: A Data Over Cable Service Interface Specification (DOCSIS) cable modem system is coupled to: i) via a local area internet protocol (IP) network, a voice over internet protocol (VoIP) device operating Session Initiation Protocol (SIP) for signaling a VoIP session; and ii) via a DOCSIS network, a cable modem termination system (CMTS) via a network. The cable modem system comprises instructions stored in a memory and executed by a processor. The instructions comprise: i) in response to receiving a frame via the local area IP network, determining if the frame is a Session Initiation Protocol (SIP) invite message signaling a VoIP session with a remote endpoint; and ii) in response to determining that the frame is a SIP invite message, generating a DOCSIS message to the CMTS to request an addition of reserved bandwidth on the DOCSIS network for the VoIP session.

Abstract: A Data-Over-Cable Service Interface Specification (DOCSIS) cable modem system is coupled to: i) via a local area internet protocol (IP) network, a voice over internet protocol (VoIP) device operating Session Initiation Protocol (SIP) for signaling a VoIP media session; and ii) via a DOCSIS network, a cable modem termination system (CMTS) via a network. The cable modem system comprises instructions stored in a memory and executed by a processor. The instructions comprise: i) in response to receiving a frame via the local area IP network, determining if the frame is a Session Initiation Protocol (SIP) invite message signaling a VoIP session with a remote endpoint; and ii) in response to determining that the frame is a SIP invite message, generating a DOCSIS message to the CMTS to request an addition of reserved bandwidth on the DOCSIS network for the VoIP session.

Abstract: A cable modem integrated session border control circuit operates as a point of demarcation between a local area network (LAN) and a DOCSIS network and, in response to receiving a Session Initiation Protocol (SIP) message, which includes Session Description Protocol (SDP), from a VoIP device coupled to the LAN, communicates with a Cable Modem Termination System (CMTS) to take advantage of DOCSIS Dynamic Quality of Service (DQoS) if a VoIP session between the VoIP device and a remote endpoint includes use of the DOCSIS network. The cable modem integrated session border controller further determines required service flow attributes. If required service flow attributes, as determined from the SDP of the SIP message, matches service flow attributes of an existing UGS service flow with a CMTS, a DOCSIS Dynamic Service Chance (DSC) request is used to add an additional sub flow to the existing UGS service flow.

Abstract: An electronic device with improved heat dissipation properties comprises metallic housing. A base circuit board may be positioned within the housing in a plane parallel to the backside. Multiple connectors for coupling cassettes to the base circuit board may be positioned about the periphery of the base circuit board such that multiple cassettes may be positioned adjacent to each side of the housing. Each cassette includes a printed circuit board with a plurality of heat generating components coupled to the printed circuit board and having a thickness extending towards the side of the housing. The cassette includes a metallic housing with a heat dissipation structure contacting: i) the side of the housing; and ii) each heat generating component. Each of at least two heat generating components have different thicknesses and the heat dissipation structure had different thicknesses between the side of the housing and each heat generating component.

Abstract: A VoIP telephone operates in an IP environment wherein at least a portion of the signaling from a remote gateway is included within a media session real time protocol stream. The VoIP telephone comprises a network communication system for encapsulating data into IP frames for exchange with remote devices over a frame switched network. A system client application is coupled to the network communication system and exchanges call set up messages with a remote VoIP gateway to establish a media channel for the exchange of the real time protocol streams. The system client also provides VoIP status signals to a presentation module. A dialog system is coupled to the network communication system and: i) translates frames of compressed digital audio data originated from a remote device to recreate remote voice band; ii) detects and compresses local voice band for transmission to the VoIP gateway; iii) detects in band signaling within the voice band; and iv) generates in band status signals to the presentation module.

Abstract: A system and method for establishing a shared secret media key between each of a caller endpoint and a callee endpoint for securing a real time media channel comprises: i) establishing a caller authenticated signaling session with the caller endpoint using a caller shared secret authentication key and a symmetric encryption algorithm; and ii) establishing a callee authenticated signaling session with the callee endpoint using a callee shared secret authentication key and the symmetric encryption algorithm. A caller public value is received from the caller endpoint through the caller authenticated signaling session and sent to the callee endpoint through the callee authenticated signaling session. The caller public value is a public value of a pair of values generated by the caller endpoint and useful for calculating a shared secret media key.

Abstract: A media gateway for coupling to both an Internet service provider frame switched network and a PSTN provider circuit switched network comprises an FXS port to which a telephony device may be coupled to initiate a connection to a destination. A programmable DSP includes a time-division multiplexing bus coupled to the FXS port and a PSTN port which couples to a subscriber loop of the circuit switched network. Compression instruction code is programmed into the digital signal processor in response to determining that the connection to the destination is to be established using a real time communication channel over the frame switched network. Time slot switching instruction code is programmed into the digital signal processor in response to determining that the connection to the destination is to be established over the circuit switched network.

Abstract: A system for securing communications between a client and an application server comprises a session key management server and the application server. The system enables network address translation firewall traversal. The session key management server comprises a key management application, a session key database, and a notification services application. The key management application receives a first transport layer security connection request from the client and negotiates a device session master key with the client as part of the transport layer security exchange. The session key database is coupled to the key management application for storing the device session master key in conjunction with an identification of the client. The notification services application coupled to the session key database and provides a notification message to subscribing application servers. The notification message comprises the device session master key in conjunction with an identification of the client.

Abstract: A method for securely provisioning a device for operation within a service provider infrastructure over an open network comprises the device establishing physical and data link layer network connections for communication on at least a subnet of the open network and obtaining a network configuration data such as an IP address and a subnet mask from a provisioning server that responds to a network configuration broadcast request. A device establishes a secure hypertext transport protocol connection to a first provisioning server that corresponds to one of: i) and IP address and port number; and ii) a fully qualified domain name stored in a non-volatile memory of the device. After mutual authentication, the first provisioning server provides at least one of: i) a configuration file; and ii) identification of a second provisioning server and a cipher key through the secure connection.

Abstract: A system and method for providing a Voice-over Internet Protocol (VoIP) system configured to allow voice data to be transmitted and received over a network. The VoIP system includes a network including at least two VoIP proxy servers configured to allow voice data to be transmitted and received over the network. The VoIP system also includes at least two VoIP clients operatively coupled to the network to transmit and receive voice data over the network and a load balancing proxy server device configured to determine an identity of a VoIP proxy server with a lowest workload. The load balancing proxy server device is also configured to provide the identity of the VoIP proxy server with the lowest workload to one of the at least two VoIP clients so that the one of the at least two VoIP clients can connect to the VoIP proxy server with the lowest workload in order to transmit and receive voice data in relation to another of the at least two VoIP clients.

Abstract: A system and method for providing a Voice-over Internet Protocol (VoIP) system is disclosed. The VoIP system includes a network including at least two VoIP proxy servers configured to allow voice data to be transmitted and received over the network. The VoIP system also includes at least one VoIP client operatively coupled to the network to transmit and receive voice data over the network and a load monitor device configured to monitor workload on the at least two VoIP proxy servers and determine an identity of the VoIP proxy server with a lowest workload. The load monitor device is also configured to provide the identity of the VoIP proxy server with the lowest workload to the at least one VoIP client so that the at least one VoIP client can connect to the VoIP proxy server with the lowest workload to transmit and receive voice data.

Abstract: A system and method for providing a Voice-over-Internet Protocol (VoIP) system is disclosed. The VoIP system includes a network including at least two VoIP proxy servers configured to shift workload automatically and to allow voice data to be transmitted and received over the network and at least one VoIP client operatively is coupled to the network to transmit and receive voice data over the network.

Abstract: A system and method for establishing a shared secret media key between each of a caller endpoint and a callee endpoint for securing a real time media channel comprises: i) establishing a caller authenticated signaling session with the caller endpoint using a caller shared secret authentication key and a symmetric encryption algorithm; and ii) establishing a callee authenticated signaling session with the callee endpoint using a callee shared secret authentication key and the symmetric encryption algorithm. A caller public value is received from the caller endpoint through the caller authenticated signaling session and sent to the callee endpoint through the callee authenticated signaling session. The caller public value is a public value of a pair of values generated by the caller endpoint and useful for calculating a shared secret media key.

Abstract: An element management system enables a network management server to provide a variable value to a management information base of a managed device independent of whether the managed device is served by a network address translation firewall. The element management system comprises an SNMP message manager which receives periodic heart beat frames from the managed device and stored identification of a heart beat channel in association with identification of the managed device in a registration table. The heart beat channel comprises the source socket and destination socket of the heart beat frame. The SNMP message manager further uniquely associates an assigned UDP port number with the managed device and provides the unique association of the assigned UDP port and the managed device to the network management server.

Abstract: A media gateway control protocol (MGCP) proxy server interfaces between a plurality of MGCP gateways and at least one MGCP call agent which may be coupled to a private network and served by a network address translation firewall. The proxy server comprises a private network interface for communicating over a private network with the call agent and a public network interface for communicating over the Internet. A translation module provides for receiving an MGCP message generated by a gateway and addressed to the public network interface. The MGCP message includes a message transaction ID assigned by the gateway and an endpoint ID identifying the gateway. The endpoint ID comprises a local endpoint name and a domain.

Abstract: A jitter buffer system for reducing jitter in a packet audio reception device comprises an output time stamp index, a delay calculation module, and a histogram module. The output time stamp index determines an initial output time stamp value upon receipt of a jitter buffer latency value and increments the output time stamp upon release of each of the frames by a jitter buffer. The delay calculation module calculates a delay value for each of the sequence of received frames. The delay value is equal to the time difference between the output time stamp value and a transmission time stamp assigned to the frame by the transmitting system.

Abstract: The invention is an echo canceller for reducing the magnitude of an echo occurring in a return channel of a communication device and a method of implementing the same. The echo canceller includes a processing unit for generating an echo attenuated signal. The processing unit includes a first input, a second input and an output. The first input is for receiving a first signal that is the precursor of the echo. The second input is for receiving a second signal that potentially contains the echo. The output is for outputting the echo attenuated signal. The echo attenuated signal is derived by combining the second signal and an adaptation signal. The adaptation signal is generated by normalizing the echo attenuated signal to produce a normalized adaptation signal, and combining the normalized adaptation signal with the first signal to generate an intermediate signal. Additionally, the intermediate signal is de-normalized to produce the adaptation signal.

Abstract: A VoIP telephone operates in an IP environment wherein at least a portion of the signaling from a remote gateway is included within a media session real time protocol stream. The VoIP telephone comprises a network communication system for encapsulating data into IP frames for exchange with remote devices over a frame switched network. A system client application is coupled to the network communication system and exchanges call set up messages with a remote VoIP gateway to establish a media channel for the exchange of the real time protocol streams. The system client also provides VoIP status signals to a presentation module. A dialog system is coupled to the network communication system and: i) translates frames of compressed digital audio data originated from a remote device to recreate remote voice band; ii) detects and compresses local voice band for transmission to the VoIP gateway; iii) detects in band signaling within the voice band; and iv) generates in band status signals to the presentation module.

Abstract: A gateway comprises a router module coupled between a local area network interface and a wide area network interface. The router module receives an outbound IP frame from the local area network interface and provides a corresponding translated outbound IP frame to the wide area network interface. The router module comprises a transport layer translation module for performing network address and port translation on an IP header of the outbound IP frame. The router module further comprises an application layer translation module for detecting the presence of media session signaling information within payload of the outbound IP frame and performing network address translation and port translation of source network address information identified in the media session signaling information.

Abstract: A sub-manager interfaces between a traditional SNMP network management system (NMS) a plurality of clients, each of which may be served by a network address and port translation firewall. The sub-manager operates a master management information base and receives master network management request messages from the network management server. The master network management request message includes at least one master object identifier which comprises a client identifier which identifies a particular one of the clients and a variable portion that identifies a variable value within a client management information base. The sub-manager, in response, generates one or more client network management request messages to identified clients over TCP/IP connections through the firewall. The client network management request message includes a client object identifier that identifies the variable within the client management information base.