A networked system is provided for transporting digital media packets, such as audio and video. The network includes network devices interconnected to send and receive packets. Each network device can receive and transmit media signals from media devices. A master clock generates a system time signal that the network devices use, together with a network time protocol to generate a local clock signal synchronised to the system time signal for both rate and offset. The local clock signal governs both the rate and offset of the received or transmitted media signals. The system, which can be implemented using conventional network equipment enables media signals to be transported to meet quality and timing requirements for high quality audio and video reproduction.

Method for transporting digital media

A network monitoring device includes a flow processing element, disposed to receive flow information relating to network flows, and to generate a set of virtual packets, each representing a portion of a network flow. The virtual packets are maintained in a time-sequential order, and read by elements of the network monitoring device to generate information relating to network traffic, such as symptoms affecting the communication network, problems affecting the communication network, and otherwise. The network monitoring device randomly samples virtual packets, with at least one of two effects: (1) flow information from traffic reporting devices that are themselves sampling at differing rates can be equalized, with the effect of standardizing information from all of them; (2) the network monitoring device itself can restrict its attention to a fraction of all virtual packets, with the effect of keeping up with a relatively large number of virtual packets.

Network monitoring using virtual packets

The present disclosure is related to transmitting and receiving media channels, such as audio and video channels These channels may be transmitted as packets from one or more transmitting devices to one or more receiving devices for playout Certain embodiments of the present disclosure include systems, methods, and computer-readable media for determining latency of a data network for synchronized playout of received signals Additionally, certain embodiments of the present disclosure include a systems, methods, and computer-readable media for synchronizing playout among devices connected to a data network

Systems, Methods and Computer-Readable Media for Configuring Receiver Latency

A real-time transport protocol (RTP) describes a payload format for transporting IEC 61883-1 CIP compliant IEEE 1394-2000 isochronous transport data. The transport data includes a stream format, such as DV (Digital Video), AM824 (Audio/Music data format with an 8-bit header and 24 bits of audio), or MPEG, that has been packetized for isochronous transport by a source. The payload format is opaque to the transport mechanism. The isochronous transport clock is derived from the IEEE 1394-2000 cycle timer clock. The RTP is used to transport IEEE 1394-2000, IEC 61883 compliant data streams between IEEE 1394-2000 buses using IP (Internet Protocol), specifically, Ethernet/IP. Alternatively, other IP formats are used.

Real-time transport protocol

Techniques are described for encoding an audio video stream that is transmitted over a network, for example a wireless or IP network, such that an entire frame of audio and an entire frame of video are transmitted simultaneously within a period required to render the audio video stream frames by an application in a receiver. Aspects of the techniques include receiving audio and video RTP streams and assigning an entire frame of RTP video data to communication channel packets that occupy the same period, or less, as the video frame rate. Also an entire frame of RTP audio data is assigned to communication channel packets that occupy the same period, or less, as the audio frame rate. The video and audio communication channel packets are transmitted simultaneously. Receiving and assigning RTP streams can be performed in a remote station, or a base station.

Synchronization of audio and video data in a wireless communication system

A method and system for avoiding variance in transmission times of real-time, packet-based communications is disclosed. Transmissions such as voice and videoconferencing generally require a substantially constant delivery rate in order to be satisfactory to end users. When occurring via a packet-based network, such transmissions often experience unacceptable levels of variation in delay in their transmission times, and this variation in delay is known as jitter. One cause of jitter relates to collisions between packets generated by a first source and sent via a packet transport medium and other packets, generated by one or more other sources and transmitted by the same transport medium. The method and system disclosed allow data sources to independently establish their respective transmission timings so as to avoid these collisions, without a need for a master clock shared between all of the data sources. By operating in such a peer-to-peer manner, the data sources establish a reduction in jitter in a manner that is reliable, efficient, inexpensive, scaleable, and easy to implement.

Luis Ortiz

Papers

Methods and systems for jitter minimization in streaming media