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 wherein the at least one VoIP client connects to one of the at least two VoIP proxy servers which has a lower workload in order to transmit and receive voice data; and a network connection coupled to the at least two VoIP proxy servers for sharing workload data therebetween and the workload data is used to determine which of the at least two VoIP proxy servers the at least one VoIP client connects to in order to transmit and receive voice data.

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 device provides for sending datagrams representing real time streaming media frames to a client independent of whether the client is served by a network address proxy. The device receives a datagram originated by the client that includes an indicated network address and an indicated port number for receipt of the datagrams representing real time streaming media frames, extracts a source network address and a source port number from the datagram originated by the client, and compares the indicated network address to the source network address. The device addresses the datagrams representing real time streaming media frames to the source network address and source port number if the indicated network address and the source network address are not the same.

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 method for communicating real time media data between a first client and a second client across a packet switched data network is provided. The method includes receiving an indication of a first client network address for use as a destination network address for sending media datagrams to the first client. A media datagram originated by the first client is also received. A comparison between the first client network address and a source network address extracted from the media datagram originated by the first client is made. A media datagram is sent to the first client using the source network address if the source network address and the first client network receiving address are not the same and the media datagram is sent to the first client using the first client network address if the source network address and the first client network address are the same.

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: The improved AES processing method provides an efficient alternative to both Mips intensive multiplication and to conventional table lookup, used to multiply terms over a Galois field (GF). The improved method takes advantage of the fact that in the GF, any non zero element X can be represented by a power of a primitive element P. The improved method thereby results in a 2 by 256 table. The log base P of the terms being multiplied are looked up and summed, and the anti-log of the sum is looked up in the same table.

Abstract: A method of audio communication between a first telephony client located behind a network address translation (NAT) server and a remote second telephony client is disclosed. A calibration datagram is sent from the first telephony client to the second telephony client on a user datagram protocol (UDP) channel identified for sending audio data. The second telephony client extracts the source address and port number to identify a reverse UDP channel for sending audio data to the first telephony client.

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 method of audio communication utilizing media datagrams between a first telephony client located behind a network address translation (NAT) server and a remote second telephony client is disclosed. Each client utilizes a single port number for both sending and receiving media datagrams. A media datagram is sent from the first telephony client to the second telephony client on a UDP/IP channel utilizing a destination IP address and port number provided by the second telephony client. The second telephony client extracts the source IP address and source port number from the received media datagram to determine if the first telephony client is located behind a NAT server. If the first telephony client is located behind a NAT server, the extracted source IP address and port number are stored and used to send media datagrams to the first telephony client located behind the NAT server.

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 identity of a remote speaker is verified by receiving compressed audio formants from a remote Internet telephony client and comparing the compressed audio formants with sample compressed audio formants known to represent the person the remote speaker purports to be. The compressed audio formants include energy and pitch data characterizing the residue of the speaker uttering a predetermined pass phrase and a plurality of formant coefficients characterizing the resonance of the speaker.

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.

Abstract: When a customer premises internet telephony device (CPE) is manufactured, the factory stores a unique CPE ID number and a contact for a pre-provisioning server in non volatile memory of the CPE. At the time a CPE is purchased by a customer, an internet telephony service provider is selected and a provisioning entity is assigned to the CPE. The unique CPE ID number and a provisioning contact are stored on the pre-provisioning server.

Abstract: A system for initiating and maintaining a real time audio or video media session between two clients, at least one of which has a private network IP address and is supported by a NAT firewall, comprises a proxy server serving each client and a relay server. The first proxy server may receive an invite message from a caller client to initiate a media session with a callee client. The invite message will identify the IP address and media port number of the caller client. The proxy server queries the relay server to obtain a port number of the relay server that may be used for relaying the media session between the caller client and the callee client. The proxy server will replace the IP address and port number of the caller client with the IP address and port number of the relay server in the invite message before forwarding to the callee client.