An ad hoc conferencing system and methods are provided. Information is stored to associate a particular meeting link with an endpoint device. A first request to initiate a communication session is received from the endpoint device, wherein the first request is generated by a local participant interacting with the endpoint device. A second request to join the communication session is received from a remote device, wherein the second request is transmitted in response to a remote participant inputting, on the remote device, the particular meeting link. The communication session is established between the endpoint device and the remote device based on receiving the first request and receiving the second request.

Disclosed embodiments provide a framework for provisioning a virtual browser application environment accessible using a native browser application. The virtual browser application environment is implemented on a virtual machine instance, where the virtual environment is isolated from other functionality of the virtual machine instance. In response to a request to access the environment, a set of access control policies are identified. The access control policies define a level of access to actions that can be performed within the environment. The environment is presented through a website executed on a user's native browser application. The user's interactions with the environment and with any other users within the environment are monitored in real-time according to the access control policies.

Embodiments of the present disclosure relates to the field of communication technology, which provide a method for real-time communication connection, a server, and a storage medium. The server is configured to have service processes in one-to-one correspondence with UDP ports. After receiving each respective SDP request sent by each of at least one client for resource negotiation from a TCP port of the server, the service processes send each respective SDP response to each of the at least one client based on the each respective SDP request. The each respective SDP response carries information of a UDP port corresponding to a service process that processes the each respective SDP request. The service process receives a UDP connection request sent by a client from the UDP port corresponding to the service process, and establishes a UDP connection by calling a Connect function to implement RTC connection.

In one embodiment, a device obtains traffic for an application to be sent to a plurality of hybrid Information-Centric Networking clients. The device divides the plurality of hybrid Information-Centric Networking clients into active hybrid Information-Centric Networking clients and passive hybrid Information-Centric Networking clients, based in part on whether a given hybrid Information-Centric Networking client is sharing content with the plurality of hybrid Information-Centric Networking clients via the application. The device sends, to the active hybrid Information-Centric Networking clients, the traffic using a first transport mechanism that guarantees a level of end-to-end latency between the device and the active hybrid Information-Centric Networking clients. The device sends, to the passive hybrid Information-Centric Networking clients, the traffic using a second transport mechanism that does not guarantee the level of end-to-end latency between the device and the passive hybrid Information-Centric Networking clients.

A method for managing the HTTP adaptive streaming of an item of digital content within a multimedia stream player terminal in a local communication network is disclosed, the multimedia stream player terminal being associated with a terminal for rendering the item of digital content. On detection of an active state of a screen saver mode of the rendering terminal, such a management method implements a progressive reduction of a video rendering quality of the streamed item of digital content, and maintains an audio rendering quality of the item of content.

A method for managing the HTTP adaptive streaming of an item of digital content within a multimedia stream player terminal in a local communication network is disclosed, the multimedia stream player terminal being associated with a terminal for rendering the item of digital content. On detection of an active state of a screen saver mode of the rendering terminal, such a management method implements a progressive reduction of a video rendering quality of the streamed item of digital content, and maintains an audio rendering quality of the item of content.

Provided are a method for realizing a video conference, and a terminal and an SIP gateway. The method for realizing a video conference is applied to a WebRTC terminal, and comprises: performing interaction of SIP signaling with an SIP gateway by means of an SIP account, so as to establish a video conference connection with an SIP terminal, wherein SIP signaling between the WebRTC terminal and the SIP gateway is transmitted by means of a WebSocket protocol, and the WebRTC terminal can parse the received SIP signaling transmitted by means of the WebSocket protocol; and sending a locally collected video stream, and/or receiving a video stream of the SIP terminal, and playing same by means of a browser.

Systems, methods, and computer readable media are directed in various embodiments for providing multiuser sessions for coordinated electronic transactions. A technical solution is directed to coordinating the electronic transactions across a plurality of instances, where the underlying users of the instances can include at least two users. Access to sensitive information can be restricted using a trusted execution environment and access can be given in accordance with the coordinated electronic transactions.

Media, methods, and systems are provided for a shim to enable dynamic selection of RTC during virtual events. A virtual event may be created, and an RTC provider may be selected for the virtual event. A connection request to connect to the RTC provider may be received from a client device. Thereafter, connection data may be signaled to the client device. A signaling message may be received from the client device. The signaling message may be intercepted and translated by the shim. The signaling message may then be transmitted to a signaling server for connecting to the RTC provider. Changes of state of the virtual event may be monitored to determine if a new RTC provider should be leveraged.

Systems and methods for interconnecting point-to-point (P2P) (i.e. SIP/H323) and web browser compatible video conferencing services. The conference platform gatewaying service may use a virtual web browser participant to send and/or receive video and/or audio over VoIP/Video standards, such as SIP/H323 or other point-to-point protocols into a web browser compatible conference by means of a virtual web browser participant. The conference platform gateway service may create a binding between a SIP address and a web meeting URL. When communication is initiated from a compatible peer device to the gateway/server by means of a point-to-point protocol, the gateway/server establishes a connection to the web-based conference using the binding between URI and URL, and establishes a connection between the Point-to-point and Web browser compatible meeting service.