摘要:
A computationally simple yet powerful forward error correction code scheme for transmission of real-time media signals, such as digitized voice, video or audio, in a packet switched network such as the Internet. An encoder at the sending end derives p redundancy blocks from each group of a k payload blocks and concatenates the redundancy blocks, respectively, with payload blocks in the next group of k payload blocks. At the receiving end, a decoder may recover up to p missing packets in a group of k packets, provided with the p redundancy blocks carried by the next group of k packets. The invention thereby enables correction from the loss of multiple packets in a row, without significantly increasing the data rate or otherwise delaying transmission.
摘要:
A method and apparatus for improving the speed and quality of end-to-end data or real-time media transmissions over an internet is disclosed. A data stream representing a media signal at a given level of compression is processed just before the data stream enters the internet. A less compressed data stream representing the same media signal is generated transmitted through the internet. Due to the lower level of compression, the underlying media signal is less sensitive to packet loss in the internet and, as a result, the media signal that arrives at the receiving end will tend to be more continuous and clear.
摘要:
A computationally simple yet powerful forward error correction code scheme for transmission of real-time media signals, such as digitized voice, video or audio, in a packet switched network such as the Internet. An encoder at the sending end derives p redundancy blocks from each group of a k payload blocks and concatenates the redundancy blocks, respectively, with payload blocks in the next group of k payload blocks. At the receiving end, a decoder may recover up to p missing packets in a group of k packets, provided with the p redundancy blocks carried by the next group of k packets. The invention thereby enables correction from the loss of multiple packets in a row, without significantly increasing the data rate or otherwise delaying transmission.
摘要:
A computationally simple yet powerful forward error correction code scheme for transmission of real-time media signals, such as digitized voice, video or audio, in a packet switched network, such as the Internet. For each window of k data packets, the invention generates and transmits at least one cross-wise parity packet taken as an index-shifted function over the k data packets. The invention thereby enables a receiving end to recover from packet loss.
摘要:
A computationally simple yet powerful forward error correction code scheme for transmission of real-time media signals, such as digitized voice, video or audio, in a packet switched network, such as the Internet. The invention appends to each of a series of payload packets a single forward error correction code that is defined by taking the XOR sum of a preceding specified number of payload packets. The invention thereby enables correction from the loss of multiple packets in a row, without significantly increasing the data rate or otherwise delaying transmission.
摘要:
A method and apparatus for improving the speed and quality of end-to-end data or real-time media transmissions over an internet is disclosed. A media stream being transmitted to the internet is channel coded at the edge of the internet in order to free upstream bit rate for use in source coding the media. The channel coded media stream may then be decoded at a remote edge of the internet to recover lost packets.
摘要:
A computationally simple yet powerful forward error correction code scheme for transmission of real-time media signals, such as digitized voice, video or audio, in a packet switched network, such as the Internet. The invention appends to each of a series of payload packets a single forward error correction code that is defined by taking the XOR sum of a preceding specified number of payload packets. The invention thereby enables correction from the loss of multiple packets in a row, without significantly increasing the data rate or otherwise delaying transmission.
摘要:
A method and apparatus for controlling the transmission of real-time media signals over a data network based on a triggering event associated with a participating teleconference station. This triggering event may be the presence or absence of motion, the presence or absence of sound, or any of a variety of other events that preferably indicate the presence or absence of a person at the station. When no person is present at the station and/or when an appropriate triggering event occurs, remote teleconference participants will not transmit media signals over the network to the station, thereby conserving valuable network bandwidth and simulating a physically common meeting between people. In turn, when a person is present at the station and/or when another appropriate triggering event occurs, remote teleconference participants may start transmitting media signals over the network to the station. Additionally, a station may control its own transmission of media signals to remote stations in response to local triggering events, thus further conserving network resources.
摘要:
A system for user-space packet modification, including a set of kernel code and a user-level application programming interface (API). The system facilitates creation of a special socket for passing packets between kernel space and user space. The system in turn facilitates creation and application of a packet filter associated with the socket, in order to trap incoming or outgoing packets being processed in the kernel at a designated point in a protocol stack. Once a packet is trapped, it is moved through the socket into user space, thereby at least temporarily preventing the protocol stack from further processing the packet. In user space, an application may operate on the packet, for instance, modifying aspects of the packet or deleting the packet altogether. The system in turn facilitates injection of a packet from user space into kernel space, and into a designated point in the protocol stack for desired stack processing.
摘要:
A high-fidelity voice/audio communication system including a high-fidelity SLIC (HSLIC) device that combines traditional BORSCHT functionality with high fidelity sampling and compression techniques. The HSLIC preferably resides on a single plug-in line card contained within a multi-cards chassis. The line card includes an analog interface that connects to a two-wire subscriber line, a high fidelity codec for sampling the analog signal at a high resolution and converting high rate digital signals to an analog signal, a voice processing client running on a microprocessor and associated digital memory. The high fidelity codec preferably has a sample rate of at least twenty thousand samples per second, and no less that 250 quantization levels. The voice processing client preferably includes an Internet Protocol (IP) processing client, Session Initiation Protocol (SIP) client, a Real Time Protocol (RTP) client, and other components of a communication protocol stack for establishing a connection over a packet based network by way of the network interface circuit. The line card establishes a high fidelity audio connection by sending an invite request to a proxy server; receiving an okay signal indicating that the request was received; sending an acknowledge signal; quantizing audio information at a sampling rate greater than twenty thousand samples per second with a resolution of no less than 4096 quantization levels; and, packetizing the quantized data for transmission to a remote device.