Systems, methods, and apparatus for providing real-time voice communication between devices connected to an Internet Protocol (IP) network and devices connected to a public switched telephone network (PSTN). In one implementation, the system includes a computer-controlled switch for connection to a local PSTN, for receiving calls from the IP network and the PSTN, and for routing calls to the PSTN and the IP network; and gate interface circuitry connected to the switch and adapted for connection to the IP network. The gate interface circuitry includes gateway circuitry for interfacing the IP network and PSTN voice circuits, and gatekeeper circuitry for performing address translation, admission control, bandwidth management, and zone management. The switch stores at least one PSTN destination address and at least one IP network destination address for each subscriber. The switch routes an incoming call to any one of the destination addresses stored for the subscriber.

The media stream delivery system encodes and fragments media streams into numerous media stream fragments maintained on fragment servers. Devices obtain fragments to reconstruct media streams including live real-time media streams for playback on the devices. A device may perform caching of media stream fragments so that particular fragments need not be accessed again from a fragment server. A fragment server or even a content provider can analyze and monitor characteristics of media streams, viewing behavior, content popularity, etc., to identify fragments for caching at the playback devices. Caching indicators along with time period indicators may be included in the media stream fragments.

The media stream delivery system encodes and fragments media streams into numerous media stream fragments maintained on fragment servers. Devices obtain fragments to reconstruct media streams including live real-time media streams for playback on the devices. A device may perform caching of media stream fragments so that particular fragments need not be accessed again from a fragment server. A fragment server or even a content provider can analyze and monitor characteristics of media streams, viewing behavior, content popularity, etc., to identify fragments for caching at the playback devices. Caching indicators along with time period indicators may be included in the media stream fragments.

A source communication device comprises a cluster manager to group destination device connections into a plurality of performance clusters. The cluster manager then causes a synchronization mechanism to be generated for each performance cluster. Each synchronization cluster can then allow data to be optimally shared between the source device and destination devices within a given cluster.

An adaptive bit rate system for use in transmitting data at adaptable bit rates. The bit rate may be adjusted according to historical behaviors and/or past operational settings, such as but not limited to adjusting a requested bit rate to a different bit rate depending the historical behavior.

In one aspect, the invention provides an interactive, multi-platform video network. In another aspect, the invention provides a dynamic scheduling method.

Provided are a method and device for sharing resources between stations in a Wireless Local Area Network (WLAN). A primary node receives a radio frame sent by a secondary node, the secondary node being a secondary node that gains a Transmission Opportunity (TXOP), and Uplink Multi-User (UL MU) shared information is carried in the radio frame; and the primary node sends a feedback frame to the secondary node in response to the radio frame, the feedback frame carrying multi-user information of shared secondary node(s), and the shared secondary node(s) being one or more secondary nodes that perform UL MU shared transmission with the secondary node which gains a TXOP.

The disclosure relates to a method performed in a first device for negotiating session descriptor parameters with a second device over an application layer protocol. The method comprises converting a value of a text based session descriptor parameter of a session description protocol, SDP, message into a binary format; encapsulating the converted SDP message into an application layer protocol message; and sending the application layer protocol message to the second device.

A data transmission device mediates transmission/reception of data when delivering encrypted data between a server and a terminal device. The data transmission device includes a receiver for receiving encrypted data, a transmitter for transmitting data received by the receiver, and a controller for controlling data transmission speed depending on the amount of received data. The controller determines whether or not a series of data are subjected to initial transmission with respect to each series of data received by the receiver. Upon determining initial transmission data, the controller precludes the initial transmission data, among a series of data, from a transmission-speed limitation range so as to control the transmitter to transmit the initial transmission data. The controller limits the data transmission speed of the transmitter by way of pacing, and therefore the controller transmits data by switching between a pacing mode and a non-pacing mode.

Embodiments include providing reliable Session Initiation Protocol (SIP) signaling for Web Real Time Communications (WebRTC) interactive flows, and related methods, systems, and computer-readable media. In one embodiment, a method for providing reliable SIP signaling for WebRTC interactive flows comprises establishing, via a stateless SIP user agent executing on a first web server, a WebRTC interactive flow between a WebRTC client executing on a computing device and a remote endpoint. The method further comprises receiving, by the WebRTC client, a call state from the stateless SIP user agent, the call state indicating a current state of the WebRTC interactive flow. The method also comprises storing, by the WebRTC client, the call state. In this manner, the call state of the WebRTC interactive flow may be accessible for restoring the WebRTC interactive flow in the event of an unexpected termination of the WebRTC client and/or the stateless SIP user agent.