308 results about "WebRTC" patented technology

Distributed application of enterprise policies to WebRTC interactive sessions, and related methods, systems, and computer-readable media are disclosed. In this regard, in one embodiment, a method for applying an enterprise policy to a WebRTC interactive session comprises receiving, by a distributed policy enforcement agent of a recipient device, a WebRTC session description object directed to the recipient device originating from a sender device via a secure network connection. The method further comprises determining, by the distributed policy enforcement agent, one or more enterprise policies based on the WebRTC session description object. The method additionally comprises applying the one or more enterprise policies to the WebRTC session description object. In this manner, an enterprise may permit establishment of a WebRTC interactive session that crosses an enterprise network boundary, while at the same time ensuring that the WebRTC interactive session complies with the one or more enterprise policies.

Scalable Web Real-Time Communication (WebRTC) media engines, and related methods, systems, and computer-readable media, are disclosed herein. In one embodiment, a method for providing a scalable WebRTC media engine comprises instantiating one or more virtual WebRTC agents, each corresponding to one or more of a plurality of WebRTC clients. The method further comprises establishing a plurality of WebRTC interactive flows, each connecting one of the one or more virtual WebRTC agents with the corresponding one or more of the plurality of WebRTC clients. The method also comprises receiving contents of the plurality of WebRTC interactive flows as input from the one or more virtual WebRTC agents, and synthesizing the contents of the plurality of WebRTC interactive flows. The method additionally comprises directing the synthesized contents as output to one of more of the plurality of WebRTC interactive flows via the corresponding one or more virtual WebRTC agents.

Remotely controlling Web Real-Time Communications (WebRTC) client functionality via WebRTC data channels, and related methods, systems, and computer-readable media are disclosed. In this regard, in one embodiment, a method for remotely controlling WebRTC client functionality comprises establishing, by a first WebRTC client executing on a first computing device and a second WebRTC client executing on a second computing device, a WebRTC media channel between the first WebRTC client and the second WebRTC client. The method further comprises establishing, between the first WebRTC client and the second WebRTC client, a WebRTC data channel affiliated with the WebRTC media channel. The method also comprises receiving, by the second WebRTC client, a client control signal originating from the first WebRTC client via the WebRTC data channel. The method additionally comprises, responsive to receiving the client control signal via the WebRTC data channel, modifying a functionality associated with the second WebRTC client.

Verification of privacy of Web Real-Time Communications (WebRTC) media channels via corresponding WebRTC data channels, and related methods, systems, and computer-readable media are disclosed. In this regard, in one embodiment, a method for verifying privacy of a WebRTC media channel comprises establishing the WebRTC media channel between first and second WebRTC clients using a keying material. The method further comprises establishing a corresponding WebRTC data channel between the first and second WebRTC clients using the keying material, and negotiating, in the WebRTC data channel, a cryptographic key exchange. The method also comprises generating a first and a second Short Authentication String (SAS) based on the cryptographic key exchange in the WebRTC data channel. The method further comprises displaying the first SAS and the second SAS, such that a mismatch between the first SAS and the second SAS indicates an existence of a man-in-the-middle (MitM) attacker.

Embodiments include managing Identity Provider (IdP) identifiers for Web Real-Time Communications (WebRTC) interactive flows, and related methods, systems, and computer-readable media. In one embodiment, a method for managing IdPs comprises selecting, by a WebRTC client executing on a computing device, one or more preferred IdP identifiers indicated by one or more preferences from a plurality of IdP identifiers corresponding to a plurality of IdPs for providing identity assertions during an establishment of a WebRTC interactive flow. The method further comprises obtaining one or more identity assertions from respective ones of the plurality of IdPs corresponding to the one or more preferred IdP identifiers. The method also comprises providing, during the establishment of the WebRTC interactive flow, the one or more identity assertions. In this manner, an entity may specify the IdP used for identity authentication, and the number of identity assertions provided during initiation of the WebRTC interactive flow.

A WebRTC system, device and method enabling a P2P communication when both ends of a communication are WebRTC enabled devices. The system and devices also enable a WebRTC client to SIP device communication. A SIP interworking function is configured to receive a SDP1 from an originating WebRTC and obtain local media information from a media interworking function. The first SIP interworking function is configured to create a SDP2 based on the SDP1 and the local media information, create a SIP message comprising a message-body field including the SDP2 and an SIP extension header field including the SDP1, and send the SIP message to an IMS or SIP server.

The invention provides a method for achieving call center video seating through the WebRTC technology. The method includes the steps of achieving the video seating method supporting a smart mobile terminal and application in a call center service by combining the WebRTC technology and the WebSocket technology. The method is characterized in that the WebSocket protocol is adopted to achieve dual-way real-time communication between a browser and a server, and the WebRTC technology is adopted to achieve audio and video collecting, coding, decoding, network transmission and displaying. The method is used for achieving the aim that the video call center seating supports the smart mobile terminal.

The invention discloses a real-time video plugin-free previewing method based on a RTSP and a real-time video plugin-free previewing system based on the RTSP, the method comprises the following steps:acquiring a real-time code stream from a front-end device; transmitting the real time code stream of the front-end device to a streaming media server through the RTSP; receiving and decoding the real-time code stream in a H.264 format into an original code stream by the streaming media server; by the streaming media server, coding the original code stream into the code stream in a VP8 format, andforwarding the code stream in the VP8 format to each client side browser; and by the client side browser, decoding the code stream in the VP8 format through a Web API based on a WebRTC technology, and directly playing the decoded code stream through HTML5. The system comprises the front-end device, the streaming media server and the client side browsers. The method and the system provided by theinvention have no need of installing plugins or extensions, support crossing browsers, also have no need of performing related setting operations of the browsers, and can preview no-delay or low-delaymonitoring videos in real time.

Back-to-back Web Real-Time Communication (WebRTC) virtual agents, and related methods, systems, and computer-readable media are disclosed herein. In one embodiment, a method for providing back-to-back virtual WebRTC agents comprises receiving, by a WebRTC server executing on a computing device, a WebRTC offer/answer exchange between first and second WebRTC clients. The method further comprises instantiating one or more virtual WebRTC agents. The method also comprises establishing a first WebRTC interactive flow between the first WebRTC client and one of the one or more virtual WebRTC agents, and a second WebRTC interactive flow between the second WebRTC client and one of the one or more virtual WebRTC agents. The method additionally comprises directing a content of the first WebRTC interactive flow to the second WebRTC interactive flow, and a content of the second WebRTC interactive flow to the first WebRTC interactive flow, via the one or more virtual WebRTC agents.

The invention relates to a video anchor method and system based on an HTML5 browser, a live video method and system, and a terminal, and belongs to the technical field of the Internet. In the invention, the browser obtains video and audio data of an anchor terminal and transmits the video and audio data to a gateway server of a live broadcast website based on the WebRTC protocol; the gateway server packages the video and audio data into streaming media data, and then transmits the video and audio data to a content distribution network based on the RTMP protocol; and then an audience terminal obtains the streaming media data from the content distribution network and plays the streaming media data. Therefore, no live broadcast software needs to be installed in the anchor terminal, and the audience can conveniently view the video and audio data by using the client software or the browser just by using the live broadcast method of the browser. In this way, a more convenient and quick network live broadcast platform is built, which is easy for users to use and enhances the experience of live broadcast. The application mode of the video anchor method and system is simple, the applicationcost is low, and the application range is also very wide.

The embodiments of the present invention relate to a transmission method for a media stream in a video conference. The method comprises: receiving a first media stream sent by a conference server, the first media stream comprising video and audio information about all conference participants in a video conference; receiving a selection instruction, the selection instruction specifically indicating to select conference participants corresponding to required video and audio information from the video and audio information about all the conference participants; according to the selection instruction, sending a transmission request message to the conference server via a WebRTC application, the transmission request message comprising identification information about the selected conference participants; and receiving a second media stream sent by the conference server according to the identification information, the second media stream comprising the video and audio information about the selected conference participants.

The invention discloses a method for previewing a GB/T28181 standard monitoring video of a cross-browser based on a WebRTC protocol. According to the method, on the premise of ensuring the real-time requirement of video monitoring on demand, the software application adaptation environment of the GB/T28181 standard can be expanded to the multi-operating system platforms, such as windows, android, etc., and the multi-browser environments, such as Chrome, Firefox, etc., meanwhile, the compatibility with a GB/T 28181 protocol and a large concurrent audio on-demand request media stream function ina standard end-to-end WebRTC video and audio live broadcast scene are also realized. According to the method, the GB/T28181 standard video monitoring on-demand real-time performance can be guaranteed,and meanwhile the software application environment based on the GB/T28181 standard is wider in use environment and higher in environment compatibility.

Compensating for user sensory impairment in Web Real-Time Communications (WebRTC) interactive sessions, and related methods, systems, and computer-readable media are disclosed. In this regard, in one embodiment, a method for compensating for a user sensory impairment in a WebRTC interactive session is provided. The method comprises receiving, by a computing device, an indication of user sensory impairment. The method further comprises receiving a content of a WebRTC interactive flow directed to the computing device. The method also comprises modifying, by the computing device, the content of the WebRTC interactive flow based on the indication of user sensory impairment. The method additionally comprises rendering the modified content of the WebRTC interactive flow. In this manner, a WebRTC interactive flow may be enhanced to compensate for a user sensory impairment, and thus the user's comprehension of the WebRTC interactive session may be improved.

A system and method to leverage Web Real-Time Communication (WebRTC) for implementing Push-to-Talk (PTT) solutions. One or more servers interface to a communications network to perform advanced voice services for one or more wireless or wired user devices, wherein the advanced voice services include a two-way half-duplex voice call within a group of the user devices comprising a PTT call session. At least one of the user devices communicates with at least one of the servers during the PTT call session using a WebRTC connection, and at least the media streams for the PTT call session are transmitted between the server and the user device using the WebRTC connection.

The invention discloses a method and a system for intercommunication between a first system and a second system, and a media gateway, and belongs to the field of the mobile Internet. According to the invention, client agents of a heterogeneous network are arranged in the media gateway; through a relay function of the media gateway, clients of the heterogeneous network and the client agents are enabled to obtain candidate address information of the opposite side and to perform connection detection, so as to establish connection; by the relay function of the media gateway, the media gateway is enabled to negotiate self media coding and decoding information with the clients of the heterogeneous network; then according to established connection in a segment manner and the negotiated media coding and decoding information, the relay transmission of media streams is performed; and finally, intercommunication between media streams of the clients of the heterogeneous network is achieved. As a result, the media intercommunication of the heterogeneous network is achieved, such as media communication between WebRTC (Web-based Real-Time Communications) and an IMS (IP Multimedia Subsystem).

A WebRTC (Web Real-Time Communication) communication method, related device and system are applied to improving the real-time performance of WebRTC communication, The method comprising: a WebRTC server receiving a call request transmitted by a calling terminal, the call request being Web signaling; obtaining the telecommunication account information of a called terminal according to the call request, and establishing on the WebRTC server session resources for the connection between the calling terminal and the called terminal according to calling terminal information, calling route information and calling type information carried in the call request; generating a WebRTC connection request comprising a WebRTC server address and session resource parameters of the session resources; transmitting the WebRTC connection request and the telecommunication account information of the called terminal to a telecommunication gateway to enable the telecommunication gateway to forward the WebRTC connection request to the called terminal, and establishing the connection from the called terminal to the session resources so as to establish the connection between the calling terminal and the called terminal, wherein the connection from the called terminal to the session resources is initiated by the called terminal according to the WebRTC connection request.

The invention discloses a multi-person voice video call method based on WebRTC, and the method comprises the following steps: enabling users to send multiple call requests at the same time through specifying connection room number Room IDs and room sizes; finally achieving the multi-person voice video call through the building of P2P connection between each two users. A communication mechanism based on WebRTC enables a system to be flexible in operation, to be quick in response and to be low in delay. A series of signal exchange and SDP negotiation is needed in the P2P connection building process. Faced with a complex network environment, the mature NAT penetrating technology can enable the users in different local area networks to communicate with each other directly. Based on the development of an Android platform, the method can be applied to various types of mobile equipment more widely, improves the applicability and flexibility, and is suitable for a small-scale multi-person voice video call.

The invention provides a method for realizing media intercommunication between a WebRTC terminal and an SIP terminal and a media gateway. Connection between the WebRTC terminal and the SIP terminal is established through a media relaying function of the media gateway; respective media coding and decoding information and SRTP secret key information of the WebRTC terminal are respectively negotiated with the WebRTC terminal and the SIP terminal through the media relaying function of the media gateway; then, transmission and conversion of a media stream are carried out through the negotiated media coding and decoding information and SRTP secrete key information; and thus, media intercommunication between the WebRTC terminal and the SIP terminal is realized.

The invention discloses a remote operation method based on WebRTC and a system. According to the method, WebRTC connection between a control end and a controlled end is established; the screen content data of the controlled end is acquired by the controlled end and is sent to the control end through WebRTC connection; the screen content data is analyzed and displayed by the control end; an inputted remote operation command is received by the control end and is sent to the controlled end through WebRTC connection; the remote operation command is received by the controlled end and is analyzed and performed. Through the method, a platform-independent, simple and effective screen sharing and remote operation mode is provided, a point-to-point, stable, efficient, high throughput, low time-delay data transmission channel between the control end and the controlled end is established, more timely screen transmission is realized, and great convenience for users is realized.

The invention provides a firewall traversal method, equipment and system based on web page browser communication. A source terminal sends signaling messages including port limited signs to a WebRTC server, sign information of a first port and a media stream corresponding to communication interaction are sent to a corresponding second port of a websocket server according to the address information of the websocket server in signaling response messages fed back by the WebRTC server and the sign information of the first port, the second port belongs to an open port corresponding to the source terminal, the websocket server forwards the media stream to a target terminal through a port corresponding to the sign information of the first port according to the address mapping relation which is stored in the WebRTC server and corresponds to the sign information of the first port, therefore, firewall traversal based on the web page browser communication in enterprise networks is achieved, communication safety of enterprise intranets is guaranteed, meanwhile, and flexibility of communication services and the compatibility and coupling degree of existing communication services are improved.

Intelligently managing Web Real-Time Communications (WebRTC) interactive flows, and related systems, methods, and computer-readable media are disclosed herein. In one embodiment, a system for intelligently managing WebRTC interactive flows comprises at least one communications interface, and an associated computing device comprising a WebRTC client. The WebRTC client is configured to receive a user input gesture directed to one or more visual representations corresponding to one or more WebRTC users, and determine a context for the WebRTC client based on a current state of the WebRTC client. The WebRTC client is further configured to obtain one or more identity attributes associated with the one or more WebRTC users, and provide one or more WebRTC interactive flows including the one or more WebRTC users based on the context, the user input gesture, and the one or more identity attributes.

The invention provides a plug-in-free live broadcast system and method based on a browser. The live broadcast system comprises an anchor end, an audience end, a media server, a STUN server and a TURNserver. Identities of the anchor end and the audience end are confirmed through the STUN. A default connection mode is preferentially employed between the anchor end and the audience end. If the default connection mode is unavailable, two parties query the STUN server and employ the TURN as a relay server. After the connection is successful, a signal server, the STUN server and the TURN server transmit and exchange SDP data and establish RTCPeerConnection prepared for WebRTC. A series of events are triggered between the anchor end and the audience end, so the anchor end can transmit audio/video data to the audience end. The system and the method have the advantages that through adoption of a WebRTC technology and a series of protocol modes, the live broadcast system can be employed withoutinstalling a plug-in or a client in a local system.