摘要:
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. If the first provisioning server provided identification of a second provisioning server, the device establishes a transport connection to the identified second provisioning server. The second provisioning server provides an encrypted file which, when decrypted using the cipher key yields the configuration information needed by the device for operation with the service provider infrastructure.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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. Both the network address and port translation of the IP header and the network address and port translation of the source network address information are recorded in a translation table such that inbound frames may be reverse translated.
摘要:
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. When the callee client generates a response message that includes the IP address and media port number of the callee client, the proxy server will replace the IP address and media port number of the callee client in the response message before forwarding the response message to the caller client.
摘要:
A network access module interconnects a stand alone multi-media terminal adapter with a network controller of a frame switched network. The network access module comprises a frame switched network interface coupled to the frame switched network for communicating with the network controller. The network access module further comprises a communication link interface for communicating with the stand alone-multi media terminal adapter. A service flow module is coupled to the frame switched network interface and coupled to the communication link interface. The service flow module receives a plurality of frames of IP traffic from the multi-media terminal adapter and sorts the frames such that each frame is delivered to the frame switched network interface at a time that corresponds to a time division logical channel which corresponds to the frame. A QoS module is coupled to the service flow module and coupled to communication link interface. The QoS module generates a quality of service request for transmission to the network controller in response to receipt of a bandwidth management instruction from the multi-media terminal adapter.
摘要:
A stand-alone multi-media terminal adapter controls a dynamic quality of service management system of a broad band network access module. The multi-media terminal adapter provides the dynamic quality of service management system with instructions to reserve, commit, and release time division logical channels on a broad band network as well as discrimination identification to be used by the network access module for identifying IP traffic that corresponds to a time division logical channel. The multi-media terminal adapter receives acknowledgement of a time division logical channel that comprises identification of a frame frequency and a frame size. The multi-media terminal adapter encapsulates compressed digital audio data representing a VoIP session into IP frames with a frame size, frame frequency, and discrimination identification that corresponds to the time division logical channel.
摘要:
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.