WO2015000356A1 - Webrtc communication method, related device and system - Google Patents
Webrtc communication method, related device and system Download PDFInfo
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- WO2015000356A1 WO2015000356A1 PCT/CN2014/079869 CN2014079869W WO2015000356A1 WO 2015000356 A1 WO2015000356 A1 WO 2015000356A1 CN 2014079869 W CN2014079869 W CN 2014079869W WO 2015000356 A1 WO2015000356 A1 WO 2015000356A1
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- webrtc
- terminal
- called terminal
- connection
- calling terminal
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/02—Details
- H04L12/16—Arrangements for providing special services to substations
- H04L12/18—Arrangements for providing special services to substations for broadcast or conference, e.g. multicast
- H04L12/189—Arrangements for providing special services to substations for broadcast or conference, e.g. multicast in combination with wireless systems
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/02—Details
- H04L12/16—Arrangements for providing special services to substations
- H04L12/18—Arrangements for providing special services to substations for broadcast or conference, e.g. multicast
- H04L12/1813—Arrangements for providing special services to substations for broadcast or conference, e.g. multicast for computer conferences, e.g. chat rooms
- H04L12/1818—Conference organisation arrangements, e.g. handling schedules, setting up parameters needed by nodes to attend a conference, booking network resources, notifying involved parties
Definitions
- the present invention relates to the field of communications technologies, and in particular, to a WebRTC communication method, related device, and system. Background technique
- WebRTC Web Real-Time Communication
- WebRTC can implement web-based video conferencing.
- WebRTC technology makes direct web communication between different terminal browsers possible, thus changing the network structure mode in which the terminal browser can only pull information through the server, which is a major change to the WEB technology.
- the main purpose of embodiments of the present invention is to provide a WebRTC communication method, related device, and system to ensure real-time performance of WebRTC communication.
- the present invention provides a WebRTC communication method, including:
- the webRTC server receives the call request sent by the calling terminal, and the call request is a web signaling; obtaining the telecommunication account information of the called terminal according to the call request, and according to the calling terminal information and the calling party carried in the call request Routing information, call type information, establishing, on the WebRTC server, a session resource that is connected between the calling terminal and the called terminal;
- the WebRTC connection request including the WebRTC server Address and session resource parameters of the session resource;
- the obtaining the telecommunication account information of the called terminal according to the call request includes:
- the call request includes the telecommunication account information of the called terminal, directly obtain the telecommunication account information of the called terminal;
- the call request includes the WebRTC account information of the called terminal, searching for a mapping relationship between the WebRTC account information of the called terminal and the telecommunication account information of the called terminal, to obtain the telecommunication of the called terminal. account information.
- a WebRTC connection request is generated, where the WebRTC connection request includes the WebRTC server address and the session resource session Resource parameters, including:
- a WebRTC connection request including a Uniform Resource Locator URL address is generated, the URL address representing the WebRTC server address and a session resource parameter of the session resource.
- the method further includes:
- a fourth possible implementation manner when the called terminal selects to establish a connection through a telecommunication network, the receiving the WebRTC initialization information of the calling terminal and Sending to the called terminal, receiving WebRTC initialization information of the called terminal, and transmitting the information to the calling terminal, so that the calling terminal and the called terminal complete WebRTC communication according to the WebRTC initialization information, Specifically:
- the TRT gateway completes the WebRTC communication.
- the present invention provides a WebRTC communication method, including:
- the called terminal receives the WebRTC connection request sent by the telecommunication gateway;
- the WebRTC connection request is generated by the WebRTC server according to the call request in the form of web signaling sent by the calling terminal, and is sent to the telecommunication gateway, including the address of the WebRTC server and Session resource parameters;
- the session resource is a call request sent by the WebRTC server according to the calling terminal as the primary
- the called terminal is allocated with the called terminal.
- the WebRTC connection request includes a uniform resource locator URL address, where the URL address represents the WebRTC server address and a session resource parameter of the session resource, or
- the WebRTC connection request includes a phone number that is obtained by the telecommunications gateway encoding the URL address.
- the method further includes:
- the telephone number is decoded to obtain a WebRTC connection request including a URL address, the URL address representing the WebRTC server address and a session resource parameter of the session resource.
- the session connected to the WebRTC server is requested according to the WebRTC connection request Resources, establish a connection with the calling terminal, including:
- the URL address is opened, and the session resource in the WebRTC server is connected to establish a connection with the calling terminal.
- the connection to the calling terminal is established according to the WebRTC connection request to connect to the session resource in the WebRTC server, Includes:
- the implementation manner, in the fifth possible implementation manner further includes: When the connection is established through the WebRTC, the WebRTC initialization information is sent to the WebRTC server, and the WebRTC initialization information of the calling terminal sent by the WebRTC server is received, and the WebRTC communication is completed with the calling terminal;
- the WebRTC initialization information is sent to the telecommunication gateway, the WebRTC initialization information of the calling terminal sent by the telecommunication gateway is received, and the WebRTC communication is completed with the calling terminal.
- the present invention provides a WebRTC server, including:
- a receiving unit configured to receive a call request sent by the calling terminal, where the call request is a web signaling
- an establishing unit configured to obtain, according to the call request received by the receiving unit, the telecommunication account information of the called terminal, according to The calling terminal information, the calling routing information, and the call type information carried in the call request received by the receiving unit, and the session resource that the calling terminal and the called terminal are connected to are established on the WebRTC server;
- Generating a unit generating a WebRTC connection request, where the WebRTC connection request includes a WebRTC server address and a session resource parameter of the session resource established by the establishing unit;
- a sending unit configured to send, to the telecommunication gateway, the WebRTC connection request generated by the generating unit and the telecommunication account information of the called terminal established by the establishing unit, so that the telecommunication gateway forwards the to the called terminal WebRTC connection request;
- connection unit establishing a connection of the called terminal to the session resource, thereby establishing a connection between the calling terminal and the called terminal, where the connection of the called terminal to the session resource is the
- the calling terminal is initiated according to the WebRTC connection request sent by the sending unit.
- the establishing unit is specifically configured to:
- the call request received by the receiving unit includes the telecommunication account information of the called terminal, directly obtain the telecommunication account information of the called terminal, and establish a session resource that is connected between the calling terminal and the called terminal;
- the receiving unit includes the WebRTC account information of the called terminal, searching for a mapping relationship between the WebRTC account information of the called terminal and the telecommunication account information of the called terminal,
- the telecom account information of the called terminal is used to establish a session resource that is connected between the calling terminal and the called terminal.
- the generating unit is specifically configured to:
- the table describes the WebRTC server address and the session resource parameter of the session resource established by the establishing unit.
- the method further includes:
- the initializing unit receives the WebRTC initialization information of the calling terminal and sends the information to the called terminal, receives the WebRTC initialization information of the called terminal, and sends the information to the calling terminal, so that the calling terminal and the calling terminal
- the called terminal completes the WebRTC communication according to the WebRTC initialization information.
- the initializing unit when the called terminal selects to establish a connection through the telecommunication network, is specifically configured to:
- the present invention provides a terminal, including:
- a receiving unit configured to receive a WebRTC connection request sent by the telecommunication gateway;
- the WebRTC connection request is generated by the WebRTC server according to a call request in the form of web signaling sent by the calling terminal, and sent to the telecommunication gateway, including the WebRTC server Address and session resource parameters;
- connection unit configured to establish, according to the WebRTC connection request received by the receiving unit, a session resource connected to the WebRTC server, to establish a connection with the calling terminal;
- the session resource is the WebRTC server according to the The call request sent by the calling terminal is allocated by the calling terminal and the called terminal.
- the WebRTC connection request includes a uniform resource locator URL address, where the URL address represents the WebRTC server address and a session resource parameter of the session resource, or
- the WebRTC connection request includes a phone number that is obtained by the telecommunications gateway encoding the URL address.
- the method further includes:
- a decoding unit configured to: when the WebRTC connection request includes a phone number, decode the phone number, obtain a WebRTC connection request including a URL address, where the URL address represents the WebRTC server address and a session resource of the session resource parameter.
- the connecting unit is specifically configured to:
- the URL address received by the receiving unit or decoded by the decoding unit is opened, and the session resource in the WebRTC server is connected to establish a connection with the calling terminal.
- the connecting unit is specifically configured to:
- An initialization unit configured to send WebRTC initialization information to the WebRTC server when the connection is established through the WebRTC, receive WebRTC initialization information of the calling terminal sent by the WebRTC server, complete WebRTC communication with the calling terminal, or select
- the WebRTC initialization information is sent to the telecommunication gateway, the WebRTC initialization information of the calling terminal sent by the telecommunication gateway is received, and the WebRTC communication is completed with the calling terminal.
- the present invention provides a WebRTC communication system, including:
- the WebRTC server is a WebRTC server provided by the foregoing invention.
- the called terminal is a terminal provided by the foregoing invention.
- the telecommunication gateway is configured to receive a WebRTC connection request sent by the WebRTC server and the telecommunication account information of the called terminal, and forward the WebRTC connection request to the called terminal.
- the WebRTC connection request is sent to the called terminal through the telecommunication gateway through the fusion communication between the WebRTC and the telecommunication network, and the information push is implemented by using the telecommunication gateway, so that the user terminal can receive the WebRTC connection without always maintaining the connection with the WebRTC server.
- Request to establish WebRTC communication thus ensuring the real-time nature of WebRTC communication.
- DRAWINGS 1 is a flowchart of Embodiment 1 of a WebRTC communication method according to an embodiment of the present invention
- Embodiment 2 is a flowchart of Embodiment 2 of a WebRTC communication method according to an embodiment of the present invention
- Embodiment 3 is a flowchart of Embodiment 3 of a WebRTC communication method according to an embodiment of the present invention
- Embodiment 4 is a flowchart of Embodiment 4 of a WebRTC communication method according to an embodiment of the present invention
- FIG. 5 is a flowchart of Embodiment 5 of a WebRTC communication method according to an embodiment of the present invention.
- Embodiment 6 is a flowchart of Embodiment 6 of a WebRTC communication method according to an embodiment of the present invention
- FIG. 7 is a schematic diagram of an embodiment of a WebRTC communication system according to an embodiment of the present invention.
- FIG. 8 is a schematic diagram of signaling interaction of an embodiment of a WebRTC communication method according to an embodiment of the present invention.
- FIG. 9 is a schematic diagram of an embodiment of a WebRTC server according to an embodiment of the present invention.
- FIG. 10 is a schematic diagram of an embodiment of a telecommunication gateway according to an embodiment of the present invention.
- FIG. 11 is a schematic diagram of an embodiment of a terminal according to an embodiment of the present invention.
- FIG. 12 is a schematic structural diagram of hardware of an embodiment of a WebRTC server according to an embodiment of the present invention.
- FIG. 13 is a schematic structural diagram of hardware of an embodiment of a telecommunication gateway according to an embodiment of the present invention.
- FIG. 14 is a schematic structural diagram of hardware of an embodiment of a terminal according to an embodiment of the present invention.
- the embodiments of the present invention will be further described in detail below with reference to the drawings and specific embodiments.
- WebRTC is a new technology in the HTML5 standard.
- the core of the WebRTC revolution is media standardization and signaling standardization. That is, in the WebRTC standard, the format of the service data transmitted after the connection between the two client browsers and the method of processing the business data are defined in detail. However, WebRTC does not define a signaling format for two client browsers to establish a connection.
- WebRTC pays attention to the transmission of audio and video media streams from client to client.
- the real-time nature makes WebRTC have high requirements for signaling, and the WebRTC service form is highly coincident with traditional telecommunication services.
- WebRTC combines with telephony signaling to form a complete service, which is a good choice in technology. Therefore, there is a need for convergence between the WebRTC and the telecommunication network, and how to implement the real-time communication between the WebRTC and the telecommunication network to ensure the real-time communication of the WebRTC. For this reason, the following embodiments of the present invention provide the following WebRTC communication method.
- FIG. 1 is a flowchart of Embodiment 1 of a WebRTC communication method according to an embodiment of the present invention.
- This embodiment may be implemented by a WebRTC server, and may include the following steps: Step 101: The WebRTC server receives a call request sent by the calling terminal, where the call request is Web signaling.
- the calling terminal that is, the WebRTC client that initiates the calling, can initiate a WebRTC connection call, and can send the call request to the WebRTC server via the Internet.
- the calling terminal can be a mobile phone, a computer or other terminal device equipped with a WebRTC-enabled browser.
- the WebRTC server receives a call request sent by the calling terminal, and the call request is web signaling.
- the calling terminal can establish a long-term two-way communication connection with the WebRTC server, such as a websocket connection, for signaling transmission.
- the call request received by the WebRTC server may carry the calling terminal information, the calling routing information, the call type information carried in the call request, and the calling terminal information, such as the related identity information of the calling terminal, for example, the calling terminal WebRTC account information, calling routing information such as related network information of the calling terminal, call type information such as the current call represents a WebRTC call request; the call request may also include related identity information of the called terminal, such as the telephone account of the called terminal Information or WebRTC account information of the called terminal.
- the WebRTC account information of the calling terminal and the WebRTC account information of the called terminal can be pre-registered in the WebRTC server, so that the WebRTC server can know the identity information of both parties that need to perform WebRTC communication.
- Step 102 Obtain the telecommunication account information of the called terminal according to the call request, and establish a connection between the calling terminal and the called terminal on the WebRTC server according to the calling terminal information, the calling routing information, and the call type information carried in the call request. Session resources.
- the WebRTC server can obtain the telecommunication account information of the called terminal according to the call request for notifying the telecommunication gateway of the telecommunication account information of the called terminal, so that the telecommunication gateway can call the called terminal.
- the process of obtaining the telecommunication account information of the called terminal according to the call request may include:
- the call request includes the telecommunication account information of the called terminal, directly obtain the telecommunication account information of the called terminal; or, if the call request includes the WebRTC account information of the called terminal, look up the WebRTC account information of the called terminal and the telecommunication of the called terminal. The mapping relationship of the account information obtains the telecommunication account information of the called terminal.
- the WebRTC server determines whether the telecom account information of the called terminal is directly included according to the call request. If not, the telecom account information of the called terminal needs to be searched by pre-storing the mapping relationship between the WebRTC account information and the telecom account information. For example, the WebRTC server can map different account information through a corporate address book or other communication management module. It should be noted that the mapping module of different account information in the WebRTC server is an optional module. When the WebRTC server does not support searching for the mapping relationship between the WebRTC account information of the called terminal and the telecom account information of the called terminal, the call request is It is necessary to directly include the telecommunication account information of the called terminal.
- the WebRTC server can allocate session resources to the connection between the calling terminal and the called terminal on the WebRTC server according to the calling terminal information, the calling routing information, and the call type information carried in the call request, that is, establish a session on the WebRTC server.
- the session resource may include a session resource parameter of the session resource, for example, the session resource is the fifth session resource in the WebRTC server, so that the calling terminal and the called terminal can connect to the session resource allocated to the called terminal.
- Step 103 Generate a WebRTC connection request, where the WebRTC connection request includes a WebRTC server address and a session resource parameter of the session resource.
- generating a WebRTC connection request, the WebRTC connection request including the WebRTC server address and the session resource parameter of the session resource may include: generating a WebRTC connection request including a URL address, the URL address representing the WebRTC server address, and Session resource parameters for session resources.
- the WebRTC server may specify a URL (Uniform Resource Locator) address for the session resource allocated on behalf of the calling terminal and the called terminal connection, and includes the WebRTC server address and the session resource parameter of the session resource, so that the WebRTC server can Place the URL address in the WebRTC connection request.
- URL Uniform Resource Locator
- Step 104 Send a WebRTC connection request to the telecommunication gateway and the telecommunication account information of the called terminal, so that the telecommunication gateway forwards the WebRTC connection request to the called terminal.
- the WebRTC server invokes the telecommunication gateway to send the WebRTC connection request to the called terminal, that is, the WebRTC server invokes the telecommunication gateway to forward the WebRTC server address and the session resource parameter of the session resource to the called terminal as the calling party information.
- the WebRTC server invokes the WebRTC connection request sent by the telecommunication gateway to the called terminal to be telecommunication signaling.
- the telecommunication signaling has the characteristics of strong real-time performance.
- the called terminal does not need to be connected to the WebRTC server in real time, and can also be called through the WebRTC server.
- the telecom gateway receives the WebRTC connection request in real time, thereby realizing the connection with the calling terminal and ensuring the real-time performance of the WebRTC communication.
- Step 105 Establish a connection between the called terminal and the session resource, thereby establishing a connection between the calling terminal and the called terminal, where the connection of the called terminal to the session resource is initiated by the called terminal according to the WebRTC connection request.
- the called terminal If the called terminal agrees to establish communication, it can obtain the WebRTC service according to the WebRTC connection request.
- the server address and the session resource parameter of the session resource may be connected to the corresponding session resource in the WebRTC server by opening the local browser, so that the calling terminal and the called terminal can establish a connection.
- the WebRTC server invokes the telecommunication gateway to send the WebRTC connection request to the called terminal through the fusion communication between the WebRTC and the telecommunication network, so that the called terminal can receive the WebRTC connection request without maintaining the connection with the WebRTC server.
- the connection between the calling terminal and the called terminal is established, thereby ensuring the real-time performance of the WebRTC communication.
- the WebRTC communication method may further include: receiving the WebRTC initialization information of the calling terminal and transmitting the information to the called terminal, and receiving the called terminal.
- the WebRTC initializes the information and sends it to the calling terminal, so that the calling terminal and the called terminal complete the WebRTC communication according to the WebRTC initialization information.
- the WebRTC initialization information of the calling terminal is received and sent to the called terminal, and the WebRTC initialization information of the called terminal is received and sent to the calling terminal, so that the calling terminal and the called terminal are according to WebRTC.
- the initialization information completes the WebRTC communication, which may be specifically: when the called terminal selects to establish a connection through the telecommunication network, receives the WebRTC initialization information of the calling terminal and sends the information to the called terminal through the telecommunication gateway, and receives the WebRTC initialization information of the called terminal through the telecommunication gateway. And sending to the calling terminal, so that the calling terminal and the called terminal complete the WebRTC communication through the telecommunication gateway according to the WebRTC initialization information.
- the WebRTC server can directly receive and forward the WebRTC initialization information of the calling terminal and the called terminal; and when the called terminal selects to establish a connection through the telecommunication network, the WebRTC server can pass the telecommunication gateway. Receive and forward WebRTC initialization information of the calling terminal and the called terminal.
- Embodiment 2 of the WebRTC communication method in the embodiment of the present invention is a flowchart of Embodiment 2 of the WebRTC communication method in the embodiment of the present invention.
- This embodiment may be implemented by a WebRTC server, and may include the following steps:
- Step 201 The WebRTC server receives a call request sent by the calling terminal, where the call request is Web signaling.
- Step 202 Determine whether the call request contains the telecom account information of the called terminal. If yes, go to step 203. If no, go to step 204.
- Step 203 directly obtain the telecommunication account information of the called terminal.
- Step 204 Find a mapping relationship between the WebRTC account information of the called terminal and the telecommunication account information of the called terminal, and obtain the telecommunication account information of the called terminal.
- Step 205 According to the calling terminal information, the calling routing information, and the call type information carried in the call request The session resource of the calling terminal and the called terminal is established on the WebRTC server.
- Step 206 Generate a WebRTC connection request, where the WebRTC connection request includes a WebRTC server address and a session resource parameter of the session resource.
- Step 207 Send a WebRTC connection request to the telecommunication gateway and the telecommunication account information of the called terminal, so that the telecommunication gateway forwards the WebRTC connection request to the called terminal.
- Step 208 Establish a connection between the called terminal and the session resource, thereby establishing a connection between the calling terminal and the called terminal, where the connection of the called terminal to the session resource is initiated by the called terminal according to the WebRTC connection request.
- Step 209 Receive the calling terminal when the called terminal selects to establish a connection through the WebRTC.
- the WebRTC initializes the information and sends it to the called terminal, receives the WebRTC initialization information of the called terminal and sends it to the calling terminal, so that the calling terminal and the called terminal complete according to the WebRTC initialization information.
- the initial process and information exchange required for the calling terminal and the called terminal to complete the WebRTC connection through the WebRTC server may include exchanging SDP (Session Description Protocol) licenses of the calling terminal and the called terminal (or other similar letters). Order), the IP address of the calling terminal and the called terminal, the list of devices participating in the communication (such as video, audio), media format, network penetration protocol (such as ice) and network penetration server information (such as google-ice) information.
- SDP Session Description Protocol
- the SDP information exchanged between the calling terminal and the called terminal can complete the communication of the WebRTC audio, video or data.
- the WebRTC server can also control calls, such as re-negotiating media information, or hanging up. That is, when the calling terminal establishes a connection with the called terminal and during the call, the WebRTC server can assist the communication party to transmit signaling information, such as connection permission, protocol media information, and on-hook signaling of the two parties. Both can be transmitted by the WebRTC server.
- Step 210 Receive the calling terminal when the called terminal selects to establish a connection through the telecommunication network.
- the WebRTC initialization information is sent to the called terminal through the telecommunication gateway, and the WebRTC initialization information of the called terminal is received by the telecommunication gateway and sent to the calling terminal, so that the calling terminal and the called terminal complete the WebRTC communication through the telecommunication gateway according to the WebRTC initialization information. .
- the telecom gateway can set the WebRTC client proxy module.
- the WebRTC server still completes the initialization process and information exchange function required for the WebRTC connection. The difference is that the WebRTC server is not directly connected to the called terminal. The information exchange is performed, and the information exchange is directly performed with the called terminal through the WebRTC client proxy module in the telecommunication gateway. Thereby, the calling terminal and the called terminal complete the WebRTC communication through the telecommunication gateway.
- the method embodiment further includes a WebRTC server.
- the WebRTC initializes the process of information exchange, so that the calling terminal and the called terminal complete the WebRTC communication, that is, complete the transmission of audio, video or data between the calling terminal and the called terminal.
- Embodiment 3 of the WebRTC communication method in the embodiment of the present invention is a flowchart of Embodiment 3 of the WebRTC communication method in the embodiment of the present invention.
- This embodiment may be implemented by a telecommunications gateway, and may include the following steps:
- Step 301 The telecommunication gateway receives the WebRTC connection request sent by the WebRTC server and the telecommunication account information of the called terminal.
- the telecommunication account information of the called terminal is obtained by the WebRTC server according to the call request sent by the calling terminal, and the WebRTC connection request is based on the WebRTC server.
- the call request generated by the calling terminal in the form of web signaling includes the WebRTC server address and the session resource parameter.
- the WebRTC server can invoke the telecommunication gateway through the open interface of the telecommunication gateway (such as the SIP interface), and the telecommunication gateway can receive the WebRTC connection request sent by the WebRTC server and the telecommunication account information of the called terminal.
- the open interface of the telecommunication gateway such as the SIP interface
- the WebRTC connection request includes a Uniform Resource Locator URL address representing the WebRTC server address and session resource parameters of the session resource.
- the generation of the WebRTC connection request can be performed by the WebRTC server.
- Step 302 Send a WebRTC connection request to the called terminal, so that the called terminal connects to the session resource in the WebRTC server, thereby establishing a connection between the calling terminal and the called terminal, where the connection of the called terminal to the session resource is called
- the terminal is initiated according to the WebRTC connection request, and the session resource is allocated by the WebRTC server according to the calling terminal information, the calling routing information, and the call type information carried in the call request, and is connected to the called terminal on the WebRTC server.
- the telecommunication gateway can complete the authentication and authentication, and push the WebRTC connection request to the called terminal through the signaling management channel in the telecommunication network, that is, the WebRTC connection request is telecommunication signaling.
- the method embodiment of the method uses the communication communication between the WebRTC and the telecommunication network, and utilizes the telecommunication gateway.
- the WebRTC connection request is sent to the called terminal, so that the called terminal can receive the WebRTC connection request without establishing the connection with the WebRTC server to establish the connection between the calling terminal and the called terminal, thereby ensuring the real-time performance of the WebRTC communication.
- the telecommunication gateway has a function similar to the caller ID, and the WebRTC connection request including the URL address (the WebRTC server address of the WebRTC server and the session resource parameter) can be sent to the called terminal through the signaling management channel.
- the Caller ID feature refers to all methods of transmitting caller information in the signaling channel, such as the BELL202 standard, which allows the transmission of caller information within 255 characters.
- the caller ID function of the telecommunication gateway can only transmit the electricity representing the calling party information.
- the phone number cannot transmit multiple characters (such as a WebRTC connection request including a URL address), so the telecommunication gateway can use the encoding method to map the phone number with the WebRTC connection request including the URL address.
- the WebRTC communication method of the embodiment of the present invention may further include: encoding the URL address into a phone number; and transmitting the WebRTC connection request to the called terminal, including: sending the included phone number to the called terminal WebRTC connection request. That is, the telecommunication gateway can push the WebRTC connection request to the called terminal.
- the WebRTC connection request can be in two different forms, one is a WebRTC connection request including a URL address, and the other is a WebRTC connection request including a phone number.
- the WebRTC client proxy module may be set at the telecommunications gateway.
- the WebRTC communication method of the embodiment of the present invention may further include: establishing a connection with the called terminal when the called terminal selects to complete the WebRTC communication through the telecommunication network, and connecting to the WebRTC according to the WebRTC connection request.
- the telecommunication gateway starts an analog WebRTC client, completes the connection with the WebRTC client of the calling terminal, starts an analog telecommunication client, and completes the connection with the telecommunication client of the called terminal.
- the data sent by the calling terminal to the called terminal is received by the analog WebRTC client of the telecommunication gateway, and after being converted by the protocol, is sent by the analog telecommunication gateway of the telecommunication gateway to the called terminal.
- the data sent by the called terminal to the calling terminal is also forwarded through the telecommunication gateway, and the telecommunication gateway can complete the conversion of the WebRTC protocol and the telecommunication protocol.
- Embodiment 4 of the WebRTC communication method in the embodiment of the present invention is a flowchart of Embodiment 4 of the WebRTC communication method in the embodiment of the present invention.
- This embodiment may be implemented by a telecommunication gateway, and may include the following steps:
- Step 401 The telecommunication gateway receives the WebRTC connection request sent by the WebRTC server and the telecommunication account information of the called terminal.
- Step 402 Send a WebRTC connection request to the called terminal.
- Step 403 When the called terminal completes the WebRTC communication through the telecommunication network, establish a connection with the called terminal, and connect to the session resource in the WebRTC server according to the WebRTC connection request.
- Step 404 Receive WebRTC initialization information of the calling terminal sent by the WebRTC server. It is sent to the called terminal, receives the WebRTC initialization information of the called terminal and sends it to the calling terminal through the WebRTC server.
- Step 405 Perform protocol conversion on the data sent between the calling terminal and the called terminal, so that the calling terminal and the called terminal complete the WebRTC communication according to the WebRTC initialization information.
- the method embodiment further includes a process of WebRTC initialization information exchange, in particular, when the called terminal selects to complete the WebRTC communication through the telecommunication network, the telecommunication gateway needs the data between the called terminal and the WebRTC server. The information is forwarded, so that the calling terminal and the called terminal complete the WebRTC communication, thereby completing the transmission of audio, video or data between the calling terminal and the called terminal.
- the method may be implemented by the called terminal, and may include the following steps:
- Step 501 The called terminal receives the WebRTC connection request sent by the telecommunication gateway.
- the WebRTC connection request is generated by the WebRTC server according to the call request in the form of Web signaling sent by the calling terminal, and is sent to the telecommunication gateway, including the WebRTC server address and the session resource parameter. .
- the WebRTC connection request includes a uniform resource locator URL address
- the URL address represents the WebRTC server address and the session resource parameter of the session resource, or the WebRTC connection request includes a phone number, which is obtained by the telecommunication gateway encoding the URL address.
- the telecommunications gateway encodes the WebRTC connection request, and the phone number needs to be decoded.
- the called terminal can store the correspondence between the phone number and the URL address, and the corresponding relationship can be called by the called party.
- the telecom client of the terminal comes with it, and can also be manually updated by the user.
- the WebRTC communication method of the embodiment of the present invention may further include: when the WebRTC connection request includes a phone number, decoding the phone number, obtaining a WebRTC connection request including a URL address, and the URL address represents a WebRTC server.
- the address and session resource parameters of the session resource may further include: when the WebRTC connection request includes a phone number, decoding the phone number, obtaining a WebRTC connection request including a URL address, and the URL address represents a WebRTC server.
- Step 502 Connect to the session resource in the WebRTC server according to the WebRTC connection request, and establish a connection with the calling terminal; the session resource is allocated by the WebRTC server according to the call request sent by the calling terminal to the calling terminal and the called terminal.
- the WebRTC connection request is sent to the called terminal through the telecommunication gateway through the fusion communication between the WebRTC and the telecommunication network, and the information push is implemented by using the telecommunication gateway, so that the user terminal can receive the WebRTC connection without always maintaining the connection with the WebRTC server.
- Request to establish WebRTC communication Thereby ensuring the real-time nature of WebRTC communication.
- the implementation process of establishing a connection with the calling terminal may include: when connecting through the WebRTC connection, opening the URL address and connecting to the WebRTC server The session resource establishes a connection with the calling terminal.
- the process of establishing a connection with the calling terminal according to the WebRTC connection request to connect to the session resource in the WebRTC server may also include: establishing a connection with the telecommunication gateway when establishing a connection through the telecommunication network. So that the telecommunication gateway connects to the session resource in the WebRTC server according to the WebRTC connection request, connects to the session resource in the WebRTC server, and establishes a connection with the calling terminal through the telecommunication gateway.
- the WebRTC communication method of the embodiment of the present invention may further include: when selecting to establish a connection through the WebRTC, sending the WebRTC initialization information to the WebRTC server, and receiving the WebRTC initialization information of the calling terminal sent by the WebRTC server, and the main The terminal is called to complete the WebRTC communication; or, when the connection is established through the telecommunication network, the WebRTC initialization information is sent to the telecommunication gateway, the WebRTC initialization information of the calling terminal sent by the telecommunication gateway is received, and the WebRTC communication is completed with the calling terminal.
- the call management software in the called terminal receives the WebRTC connection request, it can judge by the calling party information that this is a WebRTC connection request or a normal telephone call request. If it is a WebRTC connection request, the user can choose whether to directly connect with the calling terminal. If yes, the user can open a URL, open a URL address, connect to the session resource in the WebRTC server, and establish a connection with the calling terminal. And the WebRTC server can directly exchange the initialization information with the calling terminal to complete the WebRTC communication with the calling terminal; if not, it can connect with the telecommunication gateway, connect to the session resource in the WebRTC server through the telecommunication gateway, and pass the telecommunication gateway. Establishing a connection with the calling terminal, and performing initialization information exchange between the calling terminal and the called terminal through the telecommunication gateway and the WebRTC server, and completing the WebRTC communication with the calling terminal through the telecommunication gateway.
- the called terminal when the called terminal chooses to directly connect with the calling terminal, after the WebRTC initialization information exchange, the called terminal selects direct communication with the calling terminal, which is a true end-to-end communication mode; and the called terminal selects no When directly connecting with the calling terminal, after the WebRTC initialization information exchange, the called terminal selects to communicate with the calling terminal through the forwarding of the telecommunication gateway, and does not belong to the end-to-end communication in a strict sense.
- the called terminal when the called terminal does not have a browser supporting WebRTC, real-time communication with the calling terminal can also be completed.
- FIG. 6 which is a flowchart of Embodiment 6 of the WebRTC communication method in the embodiment of the present invention, the method may be implemented by the called terminal, and may include the following steps:
- Step 601 The called terminal receives a WebRTC connection request sent by the telecommunication gateway.
- Step 602 Identify the content of the WebRTC connection request.
- Step 603 Obtain the URL address when the WebRTC connection request content includes a URL address representing a WebRTC server address and a session resource parameter.
- Step 604 When the WebRTC connection request includes a phone number, the phone number is decoded to obtain a URL address including a WebRTC server address and a session resource parameter.
- Step 605 Determine whether to directly connect with the calling terminal. If yes, go to step 606. If no, go to step 608.
- Step 606 Open a URL address through a browser, connect to a session resource in the WebRTC server, and establish a connection with the calling terminal.
- Step 607 Send WebRTC initialization information to the WebRTC server, receive WebRTC initialization information of the calling terminal sent by the WebRTC server, and complete WebRTC communication with the calling terminal.
- Step 608 Establish a connection with the telecommunication gateway, so that the telecommunication gateway connects to the session resource in the WebRTC server according to the WebRTC connection request; establish a connection with the calling terminal through the telecommunication gateway.
- Step 609 Send WebRTC initialization information to the telecommunication gateway, receive the WebRTC initialization information of the calling terminal sent by the telecommunication gateway, and complete the WebRTC communication with the calling terminal through the telecommunication gateway.
- the method embodiment further includes the process that the called terminal obtains the URL address representing the WebRTC server address and the session resource parameter in the WebRTC connection request directly or by decoding, and selects to connect or select through the WebRTC.
- the method embodiment illustrates the process of establishing real-time WebRTC communication between the called terminal and the calling terminal from the perspective of the called terminal.
- the embodiment of the present invention further provides an embodiment of a WebRTC communication system, including a calling terminal 701, a WebRTC server 702, a telecommunications gateway 703, and a called terminal 704, corresponding to the foregoing embodiments of the WebRTC communication method. .
- the calling terminal 701 is configured to send a call request to the WebRTC server, where the call request may include the telecommunication account information of the called terminal; and connect to the session resource established by the WebRTC to connect the calling terminal and the called terminal to establish and call Terminal connection.
- the calling terminal is further configured to: send the WebRTC initialization information of the calling terminal to the WebRTC server, and receive the WebRTC initial of the called terminal sent by the WebRTC server. The information is completed, and the WebRTC communication with the called terminal is completed.
- the WebRTC server 702 is configured to receive a call request sent by the calling terminal, and the call request is a Web signaling; obtain the telecommunication account information of the called terminal according to the call request, and according to the calling terminal information and the calling routing information carried in the call request And calling type information, establishing a session resource connected between the calling terminal and the called terminal on the WebRTC server; generating a WebRTC connection request, the WebRTC connection request includes a WebRTC server address and a session resource parameter of the session resource; and sending a WebRTC connection request to the telecommunication gateway and The telecom account information of the called terminal, so that the telecommunication gateway forwards the WebRTC connection request to the called terminal; establishes a connection of the called terminal to the session resource, thereby establishing a connection between the calling terminal and the called terminal, wherein the called terminal reaches the session resource The connection is initiated by the called terminal according to the WebRTC connection request.
- obtaining the telecommunication account information of the called terminal according to the call request includes: if the call request includes the telecommunication account information of the called terminal, directly obtaining the telecommunication account information of the called terminal; or, if the call request The WebRTC account information of the called terminal is searched for, and the mapping relationship between the WebRTC account information of the called terminal and the telecom account information of the called terminal is obtained, and the telecom account information of the called terminal is obtained.
- a WebRTC connection request is generated, and the WebRTC connection request includes
- the WebRTC server address and the session resource parameters of the session resource include: generating a WebRTC connection request including a uniform resource locator URL address, the URL address representing a WebRTC server address, and a session resource parameter of the session resource.
- the WebRTC server is further configured to: receive the WebRTC initialization information of the calling terminal and send the information to the called terminal, receive the WebRTC initialization information of the called terminal, and send the information to the calling terminal, so that the calling terminal And the called terminal completes the WebRTC communication according to the WebRTC initialization information.
- the WebRTC initialization information of the calling terminal is received and sent to the called terminal, and the receiving is received.
- the WebRTC initialization information of the terminal is sent to the calling terminal, so that the calling terminal and the called terminal complete the WebRTC communication according to the WebRTC initialization information, specifically: receiving the WebRTC initialization information of the calling terminal and transmitting the information to the called terminal through the telecommunication gateway.
- the WebRTC initialization information of the called terminal is received by the telecommunication gateway and sent to the calling terminal, so that the calling terminal and the called terminal complete the WebRTC communication through the telecommunication gateway according to the WebRTC initialization information.
- the telecommunication gateway 703 is configured to receive the WebRTC connection request sent by the WebRTC server and the telecommunication account information of the called terminal.
- the telecommunication account information of the called terminal is obtained by the WebRTC server according to the call request sent by the calling terminal, and the WebRTC connection request is the WebRTC server.
- the connection of the terminal is called, wherein the connection of the called terminal to the session resource is initiated by the called terminal according to the WebRTC connection request, and the session resource is the calling terminal information, the calling routing information, and the call type information carried by the WebRTC server according to the call request.
- the call is connected to the called terminal on the WebRTC server.
- the WebRTC connection request includes a Uniform Resource Locator URL address representing the WebRTC server address and session resource parameters of the session resource.
- the telecommunication gateway is further configured to: encode the URL address into a phone number; and send the WebRTC connection request to the called terminal, including: sending a WebRTC connection request including the phone number to the called terminal.
- the telecommunication gateway is further configured to: when the called terminal selects to complete the WebRTC communication through the telecommunication network, establish a connection with the called terminal, and connect to the session resource in the WebRTC server according to the WebRTC connection request, To enable the called terminal to connect to the session resource in the WebRTC server; receive the WebRTC initialization information of the calling terminal sent by the WebRTC server and send it to the called terminal, receive the WebRTC initialization information of the called terminal, and send it to the calling terminal through the WebRTC server. Translating data transmitted between the calling terminal and the called terminal to enable the calling terminal and the called terminal to complete WebRTC communication according to the WebRTC initialization information.
- the called terminal 704 is configured to receive a WebRTC connection request sent by the telecommunication gateway; the WebRTC connection request is generated by the WebRTC server according to the call request in the form of Web signaling sent by the calling terminal, and sent to the telecommunication gateway, including the WebRTC server address and the session resource.
- the parameter is connected to the session resource in the WebRTC server according to the WebRTC connection request, and establishes a connection with the calling terminal; the session resource is allocated by the WebRTC server to the calling terminal and the called terminal according to the call request sent by the calling terminal.
- the WebRTC connection request includes a uniform resource locator URL address, the URL address represents a WebRTC server address, and a session resource parameter of the session resource, or the WebRTC connection request includes a phone number, and the phone number is a telecommunication gateway to the URL. Address code obtained.
- the called terminal is further configured to: when the WebRTC connection request includes a phone number, decode the phone number, obtain a WebRTC connection request including a URL address, the URL address represents a WebRTC server address, and a session resource session Resource parameters.
- a connection to a calling terminal is established according to a WebRTC connection request to connect to a session resource in the WebRTC server, including: selecting a URL to open when connecting via WebRTC
- the address, connected to the session resource in the WebRTC server establishes a connection with the calling terminal.
- the connection to the calling terminal is established according to the WebRTC connection request to connect to the session resource in the WebRTC server, including: when establishing a connection through the telecommunication network, establishing a connection with the telecommunication gateway to enable telecommunication The gateway connects to the session resource in the WebRTC server according to the WebRTC connection request; establishes a connection with the calling terminal through the telecommunication gateway.
- the called terminal is further configured to: when the connection is established through the WebRTC, send the WebRTC initialization information to the WebRTC server, receive the WebRTC initialization information of the calling terminal sent by the WebRTC server, and complete the WebRTC with the calling terminal. Or; when selecting a connection through the telecommunication network, sending WebRTC initialization information to the telecommunication gateway, receiving WebRTC initialization information of the calling terminal sent by the telecommunication gateway, and completing WebRTC communication with the calling terminal.
- Step 801 The calling terminal sends a call request to the WebRTC server, where the call request is a Web signaling.
- Step 802 The WebRTC server obtains the telecom account information of the called terminal according to the call request.
- Step 803 The WebRTC server establishes session resources connected between the calling terminal and the called terminal on the WebRTC server according to the calling terminal information, the calling routing information, and the call type information carried in the call request.
- Step 804 The WebRTC server generates a WebRTC connection request, invokes a telecommunication gateway, and sends a WebRTC connection request to the telecommunication gateway and the telecommunication account information of the called terminal.
- Step 805 The telecommunication gateway prepares to call the called terminal through authentication and authentication, and pushes the WebRTC connection request to the called terminal through the signaling channel.
- Step 806 The called terminal receives the WebRTC connection request, and determines whether to directly connect with the calling terminal according to the WebRTC connection request. If yes, the connection is established through WebRTC, and the process proceeds to step 807. If not, the connection is established through the telecommunication network. Go to step 809.
- Step 807 The called terminal opens the URL address through the browser, connects to the session resource in the WebRTC server, and establishes a connection with the calling terminal.
- Step 808 The WebRTC server exchanges the WebRTC initialization information of the calling terminal and the called terminal, and completes the WebRTC communication between the calling terminal and the called terminal.
- Step 809 The called terminal establishes a connection with the telecommunication gateway, and the telecommunication gateway connects to the session resource in the WebRTC server according to the WebRTC connection request; establishes the called terminal and the calling terminal through the telecommunication gateway. Connection.
- Step 810 The WebRTC server exchanges the WebRTC initialization information of the calling terminal and the called terminal through the telecommunication gateway, and completes the WebRTC communication between the calling terminal and the called terminal through the telecommunication gateway.
- FIG. 9 is a schematic diagram of an embodiment of a WebRTC server according to an embodiment of the present invention, which may include:
- the receiving unit 901 is configured to receive, by the WebRTC server, a call request sent by the calling terminal, where the call request is a Web signaling;
- the establishing unit 902 is configured to obtain the telecommunication account information of the called terminal according to the call request, and establish the calling terminal and the called party on the WebRTC server according to the calling terminal information, the calling routing information, and the call type information carried in the call request. Session resources connected by the terminal;
- a generating unit 903 configured to generate a WebRTC connection request, where the WebRTC connection request includes a WebRTC server address and a session resource parameter of the session resource;
- the sending unit 904 is configured to send, to the telecommunication gateway, the WebRTC connection request generated by the generating unit and the telecommunication account information of the called terminal established by the establishing unit, so that the telecommunication gateway forwards the WebRTC connection request to the called terminal;
- the connecting unit 905 establishes a connection between the called terminal and the session resource, thereby establishing a connection between the calling terminal and the called terminal, wherein the connection of the called terminal to the session resource is initiated by the called terminal according to the WebRTC connection request sent by the sending unit.
- the establishing unit may be specifically configured to:
- the receiving unit If the call request received by the receiving unit includes the telecommunication account information of the called terminal, directly obtain the telecommunication account information of the called terminal, and establish a session resource that the calling terminal connects with the called terminal;
- the mapping relationship between the WebRTC account information of the called terminal and the telecommunication account information of the called terminal is obtained, the telecommunication account information of the called terminal is obtained, and the calling party is established.
- the session resource that the terminal connects to the called terminal is established.
- the generating unit may be specifically configured to:
- a WebRTC connection request including a Uniform Resource Locator URL address is generated, and the URL address represents a WebRTC server address and a session resource parameter of the session resource established by the establishment unit.
- the WebRTC server of the embodiment of the present invention may further include: an initialization unit that receives the WebRTC initialization information of the calling terminal and sends the information to the called terminal, receives the WebRTC initialization information of the called terminal, and sends the information to the calling party.
- the terminal so that the calling terminal and the called terminal complete the WebRTC communication according to the WebRTC initialization information.
- the initializing unit when the called terminal selects to establish a connection through the telecommunication network, is specifically configured to: when the called terminal selects to establish a connection through the telecommunication network, receives the calling terminal.
- the WebRTC initialization information is sent to the called terminal through the telecommunication gateway, and the WebRTC initialization information of the called terminal is received by the telecommunication gateway and sent to the calling terminal, so that the calling terminal and the called terminal complete the WebRTC communication through the telecommunication gateway according to the WebRTC initialization information. .
- FIG. 10 it is a schematic diagram of an embodiment of a telecommunication gateway in an embodiment of the present invention, which may include: a receiving unit 1001, configured to receive a WebRTC connection request sent by a WebRTC server, and a telecommunication account information of the called terminal, and a telecommunication of the called terminal.
- a receiving unit 1001 configured to receive a WebRTC connection request sent by a WebRTC server, and a telecommunication account information of the called terminal, and a telecommunication of the called terminal.
- the account information is obtained by the WebRTC server according to the call request sent by the calling terminal, and the WebRTC connection request is generated by the WebRTC server according to the call request in the form of Web signaling sent by the calling terminal, including the WebRTC server address and the session resource parameter; the sending unit 1002 And sending, by the called terminal, a WebRTC connection request received by the receiving unit, so that the called terminal connects to the session resource in the WebRTC server, thereby establishing a connection between the calling terminal and the called terminal, where the called terminal reaches the session resource.
- connection is initiated by the called terminal according to the WebRTC connection request, and the session resource is the calling terminal information, the calling routing information, and the call type information carried by the WebRTC server according to the call request, and the calling terminal is connected to the called terminal on the WebRTC server. distributed.
- the WebRTC connection request may include a Uniform Resource Locator URL address representing the WebRTC server address and session resource parameters.
- the telecommunication gateway in the embodiment of the present invention may further include:
- a coding unit for encoding a URL address as a phone number
- the sending unit may be specifically configured to: send, to the called terminal, the coding unit code including the phone number
- the telecommunication gateway in the embodiment of the present invention may further include:
- the proxy unit is configured to establish a connection with the called terminal when the called terminal selects to complete the WebRTC communication through the telecommunication network, and connect to the session resource in the WebRTC server according to the WebRTC connection request, so that the called terminal is connected to the WebRTC server.
- Conversational resources ;
- the initialization unit is configured to receive the WebRTC initialization information of the calling terminal sent by the WebRTC server, and send the information to the called terminal, receive the WebRTC initialization information of the called terminal, and send the information to the calling terminal through the WebRTC server;
- the proxy unit may also be configured to perform protocol conversion on the data sent between the calling terminal and the called terminal, so that the calling terminal and the called terminal complete the WebRTC communication according to the WebRTC initialization information.
- FIG. 11 is a schematic diagram of an embodiment of a terminal in the embodiment of the present invention, where the terminal may be called The terminal can include:
- the receiving unit 1101 is configured to receive a WebRTC connection request sent by the telecommunication gateway; the WebRTC connection request is generated by the WebRTC server according to the call request in the form of Web signaling sent by the calling terminal, and sent to the telecommunication gateway, including the WebRTC server address and the session resource parameter. ;
- the connecting unit 1102 is configured to establish a connection with the calling terminal according to the WebRTC connection request received by the receiving unit and connect to the session resource in the WebRTC server; the session resource is the WebRTC server, the calling terminal and the called terminal according to the call request sent by the calling terminal. Called by the terminal.
- the WebRTC connection request may include a uniform resource locator URL address, the URL address represents a WebRTC server address and a session resource parameter of the session resource, or the WebRTC connection request includes a phone number, and the phone number is a telecommunication gateway pair. URL address encoding obtained.
- the terminal in the embodiment of the present invention may further include:
- the decoding unit is configured to: when the WebRTC connection request includes a phone number, decode the phone number, obtain a WebRTC connection request including a URL address, the URL address represents a WebRTC server address, and a session resource parameter of the session resource.
- the connecting unit may be specifically configured to:
- the URL address received by the receiving unit or decoded by the decoding unit is opened, and the session resource in the WebRTC server is connected to establish a connection with the calling terminal.
- the connecting unit may be specifically configured to:
- connection When the connection is established through the telecommunication network, a connection with the telecommunication gateway is established, so that the telecommunication gateway connects to the session resource in the WebRTC server according to the WebRTC connection request; and establishes a connection with the calling terminal through the telecommunication gateway.
- the terminal in the embodiment of the present invention may further include:
- the initializing unit is configured to: when the connection is established through the WebRTC, send the WebRTC initialization information to the WebRTC server, receive the WebRTC initialization information of the calling terminal sent by the WebRTC server, complete the WebRTC communication with the calling terminal; or select the connection established through the telecommunication network. Sending WebRTC initialization information to the telecommunication gateway, receiving the WebRTC initialization information of the calling terminal sent by the telecommunication gateway, and completing the WebRTC communication with the calling terminal.
- the embodiments of the present invention further provide hardware configurations of the WebRTC server, the telecommunication gateway, and the terminal, respectively.
- At least one processor e.g., a CPU
- the processor is for executing an executable module, such as a computer program, stored in the memory.
- Memory may contain high
- RAM random access memory
- the communication connection between the system gateway and at least one other network element may be implemented through at least one network interface (which may be wired or wireless), and may use an Internet, a wide area network, a local network, a metropolitan area network, or the like.
- program instructions are stored in the memory, and the program instructions may be executed by the processor, wherein the program instructions may include a receiving unit 901, an establishing unit 902, and a generating unit 903.
- the sending unit 904, the connecting unit 905, or the program instructions may further include an initializing unit.
- each unit refer to the corresponding unit disclosed in FIG. 9, and details are not described herein again.
- program instructions are stored in the memory, and the program instructions may be executed by the processor, wherein the program instructions may include the receiving unit 1001, the sending unit 1002, or the program instructions. It may also include a coding unit, a proxy unit, and an initialization unit. For the specific implementation of each unit, refer to the corresponding unit disclosed in FIG. 10, and details are not described herein again.
- program instructions are stored in the memory, and the program instructions may be executed by the processor, wherein the program instructions may include the receiving unit 1 101, the connecting unit 1 102, or the program
- the instructions may also include a decoding unit, an initialization unit.
- each unit reference may be made to the corresponding unit disclosed in FIG. 11 , and details are not described herein again.
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Abstract
Description
一种 WebRTC通信方法、 相关设备及系统 WebRTC communication method, related device and system
技术领域 本发明涉及通信技术领域,具体涉及一种 WebRTC通信方法、相关设备及系 统。 背景技术 The present invention relates to the field of communications technologies, and in particular, to a WebRTC communication method, related device, and system. Background technique
WebRTC ( Web Real-Time Communication , Web实时通信)是一项在浏览器 内部进行实时视频和音频通信的技术,例如 WebRTC可以实现基于网页的视频会 议。 这样 WebRTC技术使得不同终端浏览器之间的直接 web通信成为可能,从而 改变了终端浏览器只能通过服务器拉取信息的网络结构模式,是对 WEB技术的 一大变革。 WebRTC (Web Real-Time Communication) is a technology for real-time video and audio communication inside a browser. For example, WebRTC can implement web-based video conferencing. In this way, WebRTC technology makes direct web communication between different terminal browsers possible, thus changing the network structure mode in which the terminal browser can only pull information through the server, which is a major change to the WEB technology.
在现有技术中 , 由于 WebRTC是实时通信,需要通信双方同时在线才能 进行,因此需要通信的双方必须一直保持与 WebRTC服务器的连接才能及时 收到 WebRTC呼叫请求, 以建立通信双方的 WebRTC通信。 而 WebRTC作为 HTML5标准的一部分,可能会被各种网站使用 ,因此,用户终端需要保持与 各个 WebRTC服务器的连接, 才能收到 WebRTC呼叫请求,这将耗费用户终 端的大量使用资源,从而无法保证 WebRTC的实时通信。 发明内容 有鉴于此,本发明实施例的主要目的是提供一种 WebRTC通信方法、 相关设 备及系统,以保证 WebRTC通信的实时性。 In the prior art, since WebRTC is real-time communication, both parties need to be online at the same time. Therefore, both parties that need to communicate must maintain the connection with the WebRTC server in time to receive the WebRTC call request in time to establish WebRTC communication between the two parties. WebRTC, as part of the HTML5 standard, may be used by various websites. Therefore, the user terminal needs to maintain a connection with each WebRTC server in order to receive the WebRTC call request, which will consume a large amount of resources of the user terminal, thereby failing to guarantee WebRTC. Real-time communication. SUMMARY OF THE INVENTION In view of this, the main purpose of embodiments of the present invention is to provide a WebRTC communication method, related device, and system to ensure real-time performance of WebRTC communication.
为解决上述问题,本发明提供的技术方案如下: In order to solve the above problems, the technical solution provided by the present invention is as follows:
第一方面,本发明提供了一种 WebRTC通信方法,包括: In a first aspect, the present invention provides a WebRTC communication method, including:
WebRTC服务器接收主叫终端发送的呼叫请求,所述呼叫请求为 Web信令; 根据所述呼叫请求获得被叫终端的电信账号信息,并根据所述呼叫请求中携 带的主叫终端信息、 主叫路由信息、 呼叫类型信息,在所述 WebRTC服务器上建 立所述主叫终端与被叫终端连接的会话资源; The webRTC server receives the call request sent by the calling terminal, and the call request is a web signaling; obtaining the telecommunication account information of the called terminal according to the call request, and according to the calling terminal information and the calling party carried in the call request Routing information, call type information, establishing, on the WebRTC server, a session resource that is connected between the calling terminal and the called terminal;
生成 WebRTC连接请求,所述 WebRTC连接请求包括所述 WebRTC服务器地 址以及所述会话资源的会话资源参数; Generating a WebRTC connection request, the WebRTC connection request including the WebRTC server Address and session resource parameters of the session resource;
向电信网关发送所述 WebRTC连接请求以及所述被叫终端的电信账号信息, 以使所述电信网关向所述被叫终端转发所述 WebRTC连接请求; Transmitting, by the telecommunications gateway, the WebRTC connection request and the telecom account information of the called terminal, so that the telecommunication gateway forwards the WebRTC connection request to the called terminal;
建立所述被叫终端到所述会话资源的连接,从而建立所述主叫终端与所述被 叫终端的连接,其中所述被叫终端到所述会话资源的连接是所述被叫终端根据所 述 WebRTC连接请求发起的。 Establishing a connection of the called terminal to the session resource, so as to establish a connection between the calling terminal and the called terminal, where the connection of the called terminal to the session resource is the called terminal according to the called terminal The WebRTC connection request is initiated.
在第一方面的第一种可能的实现方式中 ,所述根据所述呼叫请求获得被叫终 端的电信账号信息,包括: In a first possible implementation manner of the first aspect, the obtaining the telecommunication account information of the called terminal according to the call request includes:
如果所述呼叫请求包含所述被叫终端的电信账号信息,直接获得所述被叫终 端的电信账号信息; If the call request includes the telecommunication account information of the called terminal, directly obtain the telecommunication account information of the called terminal;
或者,如果所述呼叫请求包含所述被叫终端的 WebRTC账号信息,査找所述 被叫终端的 WebRTC账号信息与所述被叫终端的电信账号信息的映射关系 ,获得 所述被叫终端的电信账号信息。 Or, if the call request includes the WebRTC account information of the called terminal, searching for a mapping relationship between the WebRTC account information of the called terminal and the telecommunication account information of the called terminal, to obtain the telecommunication of the called terminal. account information.
结合第一方面或者第一方面的第一种可能的实现方式,在第二种可能的实现 方式中 ,生成 WebRTC连接请求,所述 WebRTC连接请求包括所述 WebRTC服务 器地址以及所述会话资源的会话资源参数,包括: With reference to the first aspect or the first possible implementation manner of the first aspect, in a second possible implementation, a WebRTC connection request is generated, where the WebRTC connection request includes the WebRTC server address and the session resource session Resource parameters, including:
生成包括统一资源定位符 URL地址的 WebRTC连接请求,所述 URL地址代 表所述 WebRTC服务器地址以及所述会话资源的会话资源参数。 A WebRTC connection request including a Uniform Resource Locator URL address is generated, the URL address representing the WebRTC server address and a session resource parameter of the session resource.
结合第一方面或者第一方面的第一种可能的实现方式或者第一方面的第二种 可能的实现方式,在第三种可能的实现方式中 ,还包括: With reference to the first aspect, or the first possible implementation manner of the first aspect, or the second possible implementation manner of the first aspect, in a third possible implementation manner, the method further includes:
接收所述主叫终端的 WebRTC初始化信息并发送给所述被叫终端,接收所述 被叫终端的 WebRTC初始化信息并发送给所述主叫终端, 以使所述主叫终端与所 述被叫终端根据所述 WebRTC初始化信息完成 WebRTC通信。 Receiving the WebRTC initialization information of the calling terminal and transmitting the information to the called terminal, receiving the WebRTC initialization information of the called terminal, and transmitting the information to the calling terminal, so that the calling terminal and the called terminal The terminal completes the WebRTC communication according to the WebRTC initialization information.
结合第一方面的第三种可能的实现方式,在第四种可能的实现方式中 ,在所 述被叫终端选择通过电信网络建立连接时,所述接收所述主叫终端的 WebRTC初 始化信息并发送给所述被叫终端,接收所述被叫终端的 WebRTC初始化信息并发 送给所述主叫终端, 以使所述主叫终端与所述被叫终端根据所述 WebRTC初始化 信息完成 WebRTC通信,具体为 : With reference to the third possible implementation manner of the first aspect, in a fourth possible implementation manner, when the called terminal selects to establish a connection through a telecommunication network, the receiving the WebRTC initialization information of the calling terminal and Sending to the called terminal, receiving WebRTC initialization information of the called terminal, and transmitting the information to the calling terminal, so that the calling terminal and the called terminal complete WebRTC communication according to the WebRTC initialization information, Specifically:
接收所述主叫终端的 WebRTC初始化信息并通过所述电信网关发送给所述被 叫终端,通过所述电信网关接收所述被叫终端的 WebRTC初始化信息并发送给所 述主叫终端, 以使所述主叫终端与所述被叫终端根据所述 WebRTC初始化信息通 过所述电信网关完成 WebRTC通信。 Receiving WebRTC initialization information of the calling terminal, and transmitting the information to the called terminal through the telecommunication gateway, receiving, by the telecommunication gateway, WebRTC initialization information of the called terminal, and sending the information to the calling terminal, so that The calling terminal and the called terminal are connected according to the WebRTC initialization information. The TRT gateway completes the WebRTC communication.
第二方面,本发明提供了一种 WebRTC通信方法,包括: In a second aspect, the present invention provides a WebRTC communication method, including:
被叫终端接收电信网关发送的 WebRTC连接请求;所述 WebRTC连接请求是 WebRTC服务器根据主叫终端发送的 Web信令形式的呼叫请求生成并发送给所述 电信网关的,包括所述 WebRTC服务器地址以及会话资源参数; The called terminal receives the WebRTC connection request sent by the telecommunication gateway; the WebRTC connection request is generated by the WebRTC server according to the call request in the form of web signaling sent by the calling terminal, and is sent to the telecommunication gateway, including the address of the WebRTC server and Session resource parameters;
根据所述 WebRTC连接请求连接到所述 WebRTC服务器中的会话资源,建立 与所述主叫终端的连接;所述会话资源是所述 WebRTC服务器根据所述主叫终端 发送的呼叫请求为所述主叫终端与所述被叫终端分配的。 Establishing a connection with the calling terminal according to the WebRTC connection request to connect to a session resource in the WebRTC server; the session resource is a call request sent by the WebRTC server according to the calling terminal as the primary The called terminal is allocated with the called terminal.
在第二方面的第一种可能的实现方式中 ,所述 WebRTC连接请求包括统一资 源定位符 URL地址,所述 URL地址代表所述 WebRTC服务器地址以及所述会话 资源的会话资源参数,或者,所述 WebRTC连接请求包括电话号码,所述电话号 码是所述电信网关对所述 URL地址编码获得的。 In a first possible implementation manner of the second aspect, the WebRTC connection request includes a uniform resource locator URL address, where the URL address represents the WebRTC server address and a session resource parameter of the session resource, or The WebRTC connection request includes a phone number that is obtained by the telecommunications gateway encoding the URL address.
结合第二方面的第一种可能的实现方式,在第二种可能的实现方式中 ,还包 括: In conjunction with the first possible implementation of the second aspect, in a second possible implementation manner, the method further includes:
当所述 WebRTC连接请求包括电话号码,对所述电话号码进行解码,获得包 括 URL地址的 WebRTC连接请求,所述 URL地址代表所述 WebRTC服务器地址 以及所述会话资源的会话资源参数。 When the WebRTC connection request includes a telephone number, the telephone number is decoded to obtain a WebRTC connection request including a URL address, the URL address representing the WebRTC server address and a session resource parameter of the session resource.
结合第二方面的第一种可能的实现方式或者第二方面的第二种可能的实现方 式,在第三种可能的实现方式中 ,根据所述 WebRTC连接请求连接到所述 WebRTC 服务器中的会话资源,建立与主叫终端的连接,包括: In conjunction with the first possible implementation of the second aspect or the second possible implementation of the second aspect, in a third possible implementation, the session connected to the WebRTC server is requested according to the WebRTC connection request Resources, establish a connection with the calling terminal, including:
选择通过 WebRTC连接时,打开所述 URL地址,连接到所述 WebRTC服务 器中的会话资源,建立与主叫终端的连接。 When the connection through WebRTC is selected, the URL address is opened, and the session resource in the WebRTC server is connected to establish a connection with the calling terminal.
结合第二方面的第三种可能的实现方式,在第四种可能的实现方式中 ,根据 所述 WebRTC连接请求连接到所述 WebRTC服务器中的会话资源,建立与所述主 叫终端的连接,包括: With reference to the third possible implementation of the second aspect, in a fourth possible implementation, the connection to the calling terminal is established according to the WebRTC connection request to connect to the session resource in the WebRTC server, Includes:
选择通过电信网络建立连接时,建立与所述电信网关的连接, 以使所述电信 网关根据所述 WebRTC连接请求连接到所述 WebRTC服务器中的会话资源;通过 所述电信网关,建立与主叫终端的连接。 When establishing a connection through the telecommunication network, establishing a connection with the telecommunication gateway, so that the telecommunication gateway connects to the session resource in the WebRTC server according to the WebRTC connection request; establishing and calling through the telecommunication gateway Terminal connection.
结合第二方面或者第二方面的第一种可能的实现方式或者第二方面的第二种 可能的实现方式或者第二方面的第三种可能的实现方式或者第二方面的第四种可 能的实现方式,在第五种可能的实现方式中 ,还包括: 选择通过 WebRTC建立连接时,向所述 WebRTC服务器发送 WebRTC初始化 信息,接收所述 WebRTC服务器发送的主叫终端的 WebRTC初始化信息,与所述 主叫终端完成 WebRTC通信; Combining the second aspect or the first possible implementation of the second aspect or the second possible implementation of the second aspect or the third possible implementation of the second aspect or the fourth possible implementation of the second aspect The implementation manner, in the fifth possible implementation manner, further includes: When the connection is established through the WebRTC, the WebRTC initialization information is sent to the WebRTC server, and the WebRTC initialization information of the calling terminal sent by the WebRTC server is received, and the WebRTC communication is completed with the calling terminal;
或者,选择通过电信网络建立连接时, 向所述电信网关发送 WebRTC初始化 信息,接收所述电信网关发送的主叫终端的 WebRTC初始化信息,与所述主叫终 端完成 WebRTC通信。 Alternatively, when the connection is established through the telecommunication network, the WebRTC initialization information is sent to the telecommunication gateway, the WebRTC initialization information of the calling terminal sent by the telecommunication gateway is received, and the WebRTC communication is completed with the calling terminal.
第三方面,本发明提供了一种 WebRTC服务器,包括: In a third aspect, the present invention provides a WebRTC server, including:
接收单元,用于接收主叫终端发送的呼叫请求,所述呼叫请求为 Web信令; 建立单元,用于根据所述接收单元接收的所述呼叫请求获得被叫终端的电信 账号信息,并根据所述接收单元接收的所述呼叫请求中携带的主叫终端信息、 主 叫路由信息、 呼叫类型信息,在所述 WebRTC服务器上建立所述主叫终端与被叫 终端连接的会话资源; a receiving unit, configured to receive a call request sent by the calling terminal, where the call request is a web signaling, and an establishing unit, configured to obtain, according to the call request received by the receiving unit, the telecommunication account information of the called terminal, according to The calling terminal information, the calling routing information, and the call type information carried in the call request received by the receiving unit, and the session resource that the calling terminal and the called terminal are connected to are established on the WebRTC server;
生成单元,生成 WebRTC连接请求,所述 WebRTC连接请求包括 WebRTC服 务器地址以及所述建立单元建立的所述会话资源的会话资源参数; Generating a unit, generating a WebRTC connection request, where the WebRTC connection request includes a WebRTC server address and a session resource parameter of the session resource established by the establishing unit;
发送单元,用于向电信网关发送所述生成单元生成的所述 WebRTC连接请求 以及所述建立单元建立的所述被叫终端的电信账号信息, 以使所述电信网关向被 叫终端转发所述 WebRTC连接请求; a sending unit, configured to send, to the telecommunication gateway, the WebRTC connection request generated by the generating unit and the telecommunication account information of the called terminal established by the establishing unit, so that the telecommunication gateway forwards the to the called terminal WebRTC connection request;
连接单元,建立所述被叫终端到所述会话资源的连接,从而建立所述主叫终 端与所述被叫终端的连接,其中所述被叫终端到所述会话资源的连接是所述被叫 终端根据所述发送单元发送的所述 WebRTC连接请求发起的。 a connection unit, establishing a connection of the called terminal to the session resource, thereby establishing a connection between the calling terminal and the called terminal, where the connection of the called terminal to the session resource is the The calling terminal is initiated according to the WebRTC connection request sent by the sending unit.
在第三方面的第一种可能的实现方式中 ,所述建立单元具体用于: In a first possible implementation manner of the third aspect, the establishing unit is specifically configured to:
如果所述接收单元接收的所述呼叫请求包含所述被叫终端的电信账号信息, 直接获得所述被叫终端的电信账号信息,建立所述主叫终端与被叫终端连接的会 话资源; If the call request received by the receiving unit includes the telecommunication account information of the called terminal, directly obtain the telecommunication account information of the called terminal, and establish a session resource that is connected between the calling terminal and the called terminal;
或者,如果所述接收单元接收的所述呼叫请求包含所述被叫终端的 WebRTC 账号信息,査找所述被叫终端的 WebRTC账号信息与所述被叫终端的电信账号信 息的映射关系,获得所述被叫终端的电信账号信息,建立所述主叫终端与被叫终 端连接的会话资源。 Or, if the call request received by the receiving unit includes the WebRTC account information of the called terminal, searching for a mapping relationship between the WebRTC account information of the called terminal and the telecommunication account information of the called terminal, The telecom account information of the called terminal is used to establish a session resource that is connected between the calling terminal and the called terminal.
结合第三方面或者第三方面的第一种可能的实现方式,在第二种可能的实现 方式中 ,所述生成单元具体用于: With reference to the third aspect or the first possible implementation manner of the third aspect, in a second possible implementation manner, the generating unit is specifically configured to:
生成包括统一资源定位符 URL地址的 WebRTC连接请求,所述 URL地址代 表所述 WebRTC服务器地址以及所述建立单元建立的所述会话资源的会话资源参 数。 Generating a WebRTC connection request including a uniform resource locator URL address, the URL address generation The table describes the WebRTC server address and the session resource parameter of the session resource established by the establishing unit.
结合第三方面或者第三方面的第一种可能的实现方式或者第三方面的第二种 可能的实现方式,在第三种可能的实现方式中 ,还包括: With reference to the third aspect, or the first possible implementation manner of the third aspect, or the second possible implementation manner of the third aspect, in a third possible implementation manner, the method further includes:
初始化单元,接收所述主叫终端的 WebRTC初始化信息并发送给所述被叫终 端,接收所述被叫终端的 WebRTC初始化信息并发送给所述主叫终端, 以使所述 主叫终端与所述被叫终端根据所述 WebRTC初始化信息完成 WebRTC通信。 The initializing unit receives the WebRTC initialization information of the calling terminal and sends the information to the called terminal, receives the WebRTC initialization information of the called terminal, and sends the information to the calling terminal, so that the calling terminal and the calling terminal The called terminal completes the WebRTC communication according to the WebRTC initialization information.
结合第三方面的第三种可能的实现方式,在第四种可能的实现方式中 ,在所 述被叫终端选择通过电信网络建立连接时,所述初始化单元具体用于: In conjunction with the third possible implementation of the third aspect, in a fourth possible implementation, when the called terminal selects to establish a connection through the telecommunication network, the initializing unit is specifically configured to:
接收所述主叫终端的 WebRTC初始化信息并通过所述电信网关发送给所述被 叫终端,通过所述电信网关接收所述被叫终端的 WebRTC初始化信息并发送给所 述主叫终端, 以使所述主叫终端与所述被叫终端根据所述 WebRTC初始化信息通 过所述电信网关完成 WebRTC通信。 Receiving WebRTC initialization information of the calling terminal, and transmitting the information to the called terminal through the telecommunication gateway, receiving, by the telecommunication gateway, WebRTC initialization information of the called terminal, and sending the information to the calling terminal, so that The calling terminal and the called terminal complete WebRTC communication through the telecommunication gateway according to the WebRTC initialization information.
第四方面,本发明提供了一种终端,包括: In a fourth aspect, the present invention provides a terminal, including:
接收单元,用于接收电信网关发送的 WebRTC连接请求;所述 WebRTC连接 请求是 WebRTC服务器根据主叫终端发送的 Web信令形式的呼叫请求生成并发送 给所述电信网关的,包括所述 WebRTC服务器地址以及会话资源参数; a receiving unit, configured to receive a WebRTC connection request sent by the telecommunication gateway; the WebRTC connection request is generated by the WebRTC server according to a call request in the form of web signaling sent by the calling terminal, and sent to the telecommunication gateway, including the WebRTC server Address and session resource parameters;
连接单元,用于根据所述接收单元接收的所述 WebRTC连接请求连接到所述 WebRTC服务器中的会话资源,建立与所述主叫终端的连接;所述会话资源是所 述 WebRTC服务器根据所述主叫终端发送的呼叫请求为所述主叫终端与所述被叫 终端分配的。 a connection unit, configured to establish, according to the WebRTC connection request received by the receiving unit, a session resource connected to the WebRTC server, to establish a connection with the calling terminal; the session resource is the WebRTC server according to the The call request sent by the calling terminal is allocated by the calling terminal and the called terminal.
在第四方面的第一种可能的实现方式中 ,所述 WebRTC连接请求包括统一资 源定位符 URL地址,所述 URL地址代表所述 WebRTC服务器地址以及所述会话 资源的会话资源参数,或者,所述 WebRTC连接请求包括电话号码,所述电话号 码是所述电信网关对所述 URL地址编码获得的。 In a first possible implementation manner of the fourth aspect, the WebRTC connection request includes a uniform resource locator URL address, where the URL address represents the WebRTC server address and a session resource parameter of the session resource, or The WebRTC connection request includes a phone number that is obtained by the telecommunications gateway encoding the URL address.
结合第四方面的第一种可能的实现方式,在第二种可能的实现方式中 ,还包 括: In conjunction with the first possible implementation of the fourth aspect, in a second possible implementation manner, the method further includes:
解码单元,用于当所述 WebRTC连接请求包括电话号码,对所述电话号码进 行解码,获得包括 URL地址的 WebRTC连接请求,所述 URL地址代表所述 WebRTC 服务器地址以及所述会话资源的会话资源参数。 a decoding unit, configured to: when the WebRTC connection request includes a phone number, decode the phone number, obtain a WebRTC connection request including a URL address, where the URL address represents the WebRTC server address and a session resource of the session resource parameter.
结合第四方面的第一种可能的实现方式或者第四方面的第二种可能的实现方 式,在第三种可能的实现方式中 ,所述连接单元具体用于: Combining the first possible implementation of the fourth aspect or the second possible implementation of the fourth aspect In a third possible implementation manner, the connecting unit is specifically configured to:
选择通过 WebRTC连接时,打开所述接收单元接收的或所述解码单元解码的 所述 URL地址,连接到所述 WebRTC服务器中的会话资源,建立与主叫终端的连 接。 When the connection through the WebRTC is selected, the URL address received by the receiving unit or decoded by the decoding unit is opened, and the session resource in the WebRTC server is connected to establish a connection with the calling terminal.
结合第四方面的第三种可能的实现方式,在第四种可能的实现方式中 ,所述 连接单元具体用于: In conjunction with the third possible implementation of the fourth aspect, in a fourth possible implementation, the connecting unit is specifically configured to:
选择通过电信网络建立连接时,建立与所述电信网关的连接, 以使所述电信 网关根据所述 WebRTC连接请求连接到所述 WebRTC服务器中的会话资源;通过 所述电信网关,建立与主叫终端的连接。 When establishing a connection through the telecommunication network, establishing a connection with the telecommunication gateway, so that the telecommunication gateway connects to the session resource in the WebRTC server according to the WebRTC connection request; establishing and calling through the telecommunication gateway Terminal connection.
结合第四方面或者第四方面的第一种可能的实现方式或者第四方面的第二种 可能的实现方式或者第四方面的第三种可能的实现方式或者第四方面的第四种可 能的实现方式,在第五种可能的实现方式中 ,还包括: Combining the fourth aspect or the first possible implementation of the fourth aspect or the second possible implementation of the fourth aspect or the third possible implementation of the fourth aspect or the fourth possible implementation of the fourth aspect The implementation manner, in the fifth possible implementation manner, further includes:
初始化单元,用于选择通过 WebRTC建立连接时,向所述 WebRTC服务器发 送 WebRTC初始化信息,接收所述 WebRTC服务器发送的主叫终端的 WebRTC初 始化信息,与所述主叫终端完成 WebRTC通信;或者,选择通过电信网络建立连 接时, 向所述电信网关发送 WebRTC初始化信息,接收所述电信网关发送的主叫 终端的 WebRTC初始化信息,与所述主叫终端完成 WebRTC通信。 An initialization unit, configured to send WebRTC initialization information to the WebRTC server when the connection is established through the WebRTC, receive WebRTC initialization information of the calling terminal sent by the WebRTC server, complete WebRTC communication with the calling terminal, or select When the connection is established through the telecommunication network, the WebRTC initialization information is sent to the telecommunication gateway, the WebRTC initialization information of the calling terminal sent by the telecommunication gateway is received, and the WebRTC communication is completed with the calling terminal.
第五方面,本发明提供了一种 WebRTC通信系统,包括: In a fifth aspect, the present invention provides a WebRTC communication system, including:
主叫终端、 WebRTC服务器、 电信网关以及被叫终端; Calling terminal, WebRTC server, telecommunication gateway and called terminal;
所述 WebRTC服务器是上述本发明提供的一种 WebRTC服务器; The WebRTC server is a WebRTC server provided by the foregoing invention;
所述被叫终端是上述本发明提供的一种终端; The called terminal is a terminal provided by the foregoing invention;
所述电信网关,用于接收所述 WebRTC服务器发送的 WebRTC连接请求以及 所述被叫终端的电信账号信息, 向所述被叫终端转发所述 WebRTC连接请求。 The telecommunication gateway is configured to receive a WebRTC connection request sent by the WebRTC server and the telecommunication account information of the called terminal, and forward the WebRTC connection request to the called terminal.
由此可见,本发明实施例具有如下有益效果: It can be seen that the embodiments of the present invention have the following beneficial effects:
本发明实施例通过 WebRTC与电信网络的融合通信,将 WebRTC连接请求通 过电信网关发送给被叫终端,利用电信网关实现信息推送,使用户终端不用一直 保持与 WebRTC服务器的连接也能收到 WebRTC连接请求,以建立 WebRTC通信, 从而保证了 WebRTC通信的实时性。 附图说明 图 1为本发明实施例 WebRTC通信方法实施例 1的流程图; In the embodiment of the present invention, the WebRTC connection request is sent to the called terminal through the telecommunication gateway through the fusion communication between the WebRTC and the telecommunication network, and the information push is implemented by using the telecommunication gateway, so that the user terminal can receive the WebRTC connection without always maintaining the connection with the WebRTC server. Request to establish WebRTC communication, thus ensuring the real-time nature of WebRTC communication. DRAWINGS 1 is a flowchart of Embodiment 1 of a WebRTC communication method according to an embodiment of the present invention;
图 2为本发明实施例 WebRTC通信方法实施例 2的流程图; 2 is a flowchart of Embodiment 2 of a WebRTC communication method according to an embodiment of the present invention;
图 3为本发明实施例 WebRTC通信方法实施例 3的流程图; 3 is a flowchart of Embodiment 3 of a WebRTC communication method according to an embodiment of the present invention;
图 4为本发明实施例 WebRTC通信方法实施例 4的流程图; 4 is a flowchart of Embodiment 4 of a WebRTC communication method according to an embodiment of the present invention;
图 5为本发明实施例 WebRTC通信方法实施例 5的流程图; FIG. 5 is a flowchart of Embodiment 5 of a WebRTC communication method according to an embodiment of the present invention;
图 6为本发明实施例 WebRTC通信方法实施例 6的流程图; 6 is a flowchart of Embodiment 6 of a WebRTC communication method according to an embodiment of the present invention;
图 7为本发明实施例 WebRTC通信系统实施例的示意图; 7 is a schematic diagram of an embodiment of a WebRTC communication system according to an embodiment of the present invention;
图 8为本发明实施例 WebRTC通信方法实施例的信令交互示意图; FIG. 8 is a schematic diagram of signaling interaction of an embodiment of a WebRTC communication method according to an embodiment of the present invention;
图 9为本发明实施例 WebRTC服务器实施例的示意图; 9 is a schematic diagram of an embodiment of a WebRTC server according to an embodiment of the present invention;
图 10为本发明实施例电信网关实施例的示意图; FIG. 10 is a schematic diagram of an embodiment of a telecommunication gateway according to an embodiment of the present invention; FIG.
图 11为本发明实施例终端实施例的示意图; FIG. 11 is a schematic diagram of an embodiment of a terminal according to an embodiment of the present invention;
图 12为本发明实施例 WebRTC服务器实施例的硬件构成示意图; FIG. 12 is a schematic structural diagram of hardware of an embodiment of a WebRTC server according to an embodiment of the present invention; FIG.
图 13为本发明实施例电信网关实施例的硬件构成示意图; 13 is a schematic structural diagram of hardware of an embodiment of a telecommunication gateway according to an embodiment of the present invention;
图 14为本发明实施例终端实施例的硬件构成示意图。 具体实施方式 为使本发明的上述目的、 特征和优点能够更加明显易懂,下面结合附图和 具体实施方式对本发明实施例作进一步详细的说明。 FIG. 14 is a schematic structural diagram of hardware of an embodiment of a terminal according to an embodiment of the present invention. The embodiments of the present invention will be further described in detail below with reference to the drawings and specific embodiments.
本发明实施例的 WebRTC通信方法、 相关设备及系统可以用于 WebRTC通 信。 WebRTC是 HTML5标准中的一项新技术, WebRTC变革的核心在于媒体标 准化,信令去标准化。 即在 WebRTC标准中,详细定义了在两个客户端浏览器建 立连接后,传输的业务数据的格式,以及处理业务数据的方法。 但 WebRTC中没 有定义两个客户端浏览器建立起连接的信令格式。 The WebRTC communication method, related device and system of the embodiment of the present invention can be used for WebRTC communication. WebRTC is a new technology in the HTML5 standard. The core of the WebRTC revolution is media standardization and signaling standardization. That is, in the WebRTC standard, the format of the service data transmitted after the connection between the two client browsers and the method of processing the business data are defined in detail. However, WebRTC does not define a signaling format for two client browsers to establish a connection.
WebRTC关注客户端到客户端的音视频媒体流的传输,实时性使得 WebRTC 对信令的要求很高 , 而 WebRTC业务形态上和传统的电信业务高度重合。 WebRTC与电话信令结合,形成完整的业务,在技术上是一个很好的选择。因此, WebRTC和电信网络有融合的需求,而如何实现 WebRTC和电信网络互通,以保 证 WebRTC的实时通信,为此本发明实施例提供了如下的 WebRTC通信方法。 WebRTC pays attention to the transmission of audio and video media streams from client to client. The real-time nature makes WebRTC have high requirements for signaling, and the WebRTC service form is highly coincident with traditional telecommunication services. WebRTC combines with telephony signaling to form a complete service, which is a good choice in technology. Therefore, there is a need for convergence between the WebRTC and the telecommunication network, and how to implement the real-time communication between the WebRTC and the telecommunication network to ensure the real-time communication of the WebRTC. For this reason, the following embodiments of the present invention provide the following WebRTC communication method.
参见图 1所示,是本发明实施例中 WebRTC通信方法实施例 1的流程图,本 实施例可以由 WebRTC服务器实现该方法,可以包括以下步骤: 步骤 101 : WebRTC服务器接收主叫终端发送的呼叫请求,该呼叫请求为 Web 信令。 FIG. 1 is a flowchart of Embodiment 1 of a WebRTC communication method according to an embodiment of the present invention. This embodiment may be implemented by a WebRTC server, and may include the following steps: Step 101: The WebRTC server receives a call request sent by the calling terminal, where the call request is Web signaling.
主叫终端即发起主叫的 WebRTC客户端可以发起 WebRTC连接呼叫 ,可以通 过互联网将呼叫请求发送给 WebRTC服务器。 主叫终端可以是手机、 电脑或其他 安装有支持 WebRTC的浏览器的终端设备。 The calling terminal, that is, the WebRTC client that initiates the calling, can initiate a WebRTC connection call, and can send the call request to the WebRTC server via the Internet. The calling terminal can be a mobile phone, a computer or other terminal device equipped with a WebRTC-enabled browser.
WebRTC服务器接收主叫终端发送的呼叫请求,该呼叫请求为 web信令。 这 样,主叫终端可以与 WebRTC服务器建立一种长期保持的双向通信连接,例如 websocket连接,用于信令的传输。 同时, WebRTC服务器接收到的呼叫请求中可 以携带呼叫请求中携带的主叫终端信息、 主叫路由信息、 呼叫类型信息,主叫终 端信息如主叫终端的相关身份信息,例如 ,主叫终端的 WebRTC账号信息,主叫 路由信息如主叫终端的相关网络信息,呼叫类型信息例如本次呼叫代表 WebRTC 呼叫请求;呼叫请求中还可以包括被叫终端的相关身份信息,例如被叫终端的电 信账号信息或者被叫终端的 WebRTC账号信息。 The WebRTC server receives a call request sent by the calling terminal, and the call request is web signaling. In this way, the calling terminal can establish a long-term two-way communication connection with the WebRTC server, such as a websocket connection, for signaling transmission. At the same time, the call request received by the WebRTC server may carry the calling terminal information, the calling routing information, the call type information carried in the call request, and the calling terminal information, such as the related identity information of the calling terminal, for example, the calling terminal WebRTC account information, calling routing information such as related network information of the calling terminal, call type information such as the current call represents a WebRTC call request; the call request may also include related identity information of the called terminal, such as the telephone account of the called terminal Information or WebRTC account information of the called terminal.
主叫终端的 WebRTC账号信息、 被叫终端的 WebRTC账号信息均可以在 WebRTC服务器预先进行注册, 以使 WebRTC服务器可以获知需要进行 WebRTC 通信的双方的身份信息。 The WebRTC account information of the calling terminal and the WebRTC account information of the called terminal can be pre-registered in the WebRTC server, so that the WebRTC server can know the identity information of both parties that need to perform WebRTC communication.
步骤 102:根据呼叫请求获得被叫终端的电信账号信息,并根据呼叫请求中携 带的主叫终端信息、 主叫路由信息、 呼叫类型信息,在 WebRTC服务器上建立主 叫终端与被叫终端连接的会话资源。 Step 102: Obtain the telecommunication account information of the called terminal according to the call request, and establish a connection between the calling terminal and the called terminal on the WebRTC server according to the calling terminal information, the calling routing information, and the call type information carried in the call request. Session resources.
WebRTC服务器根据呼叫请求可以获得被叫终端的电信账号信息用于将被叫 终端的电信账号信息通知电信网关, 以使电信网关可以呼叫被叫终端。 The WebRTC server can obtain the telecommunication account information of the called terminal according to the call request for notifying the telecommunication gateway of the telecommunication account information of the called terminal, so that the telecommunication gateway can call the called terminal.
具体的,在本发明的一些实施例中 ,根据呼叫请求获得被叫终端的电信账号 信息的实现过程可以包括: Specifically, in some embodiments of the present invention, the process of obtaining the telecommunication account information of the called terminal according to the call request may include:
如果呼叫请求包含被叫终端的电信账号信息,直接获得被叫终端的电信账号 信息;或者,如果呼叫请求包含被叫终端的 WebRTC账号信息,査找被叫终端的 WebRTC账号信息与被叫终端的电信账号信息的映射关系 ,获得被叫终端的电信 账号信息。 If the call request includes the telecommunication account information of the called terminal, directly obtain the telecommunication account information of the called terminal; or, if the call request includes the WebRTC account information of the called terminal, look up the WebRTC account information of the called terminal and the telecommunication of the called terminal. The mapping relationship of the account information obtains the telecommunication account information of the called terminal.
即 WebRTC服务器根据呼叫请求判断其中是否直接包含了被叫终端的电信账 号信息,如果否,则需要通过预先保存 WebRTC账号信息与电信账号信息映射关 系 ,査找得到被叫终端的电信账号信息。 例如 , WebRTC服务器可以通过企业通 讯录或其他通信管理模块进行不同账号信息间的映射。 需要注意的是, WebRTC服务器中不同账号信息的映射模块是一个可选模块, 当 WebRTC服务器不支持査找被叫终端的 WebRTC账号信息与被叫终端的电信账 号信息的映射关系时,则呼叫请求中需要直接包含被叫终端的电信账号信息。 That is, the WebRTC server determines whether the telecom account information of the called terminal is directly included according to the call request. If not, the telecom account information of the called terminal needs to be searched by pre-storing the mapping relationship between the WebRTC account information and the telecom account information. For example, the WebRTC server can map different account information through a corporate address book or other communication management module. It should be noted that the mapping module of different account information in the WebRTC server is an optional module. When the WebRTC server does not support searching for the mapping relationship between the WebRTC account information of the called terminal and the telecom account information of the called terminal, the call request is It is necessary to directly include the telecommunication account information of the called terminal.
WebRTC服务器可以根据呼叫请求中携带的主叫终端信息、 主叫路由信息、 呼叫类型信息,在 WebRTC服务器上为主叫终端与被叫终端的连接分配会话资源, 即在 WebRTC服务器上建立一个用于主叫终端与被叫终端连接的会话资源。 该会 话资源可以包括会话资源的会话资源参数,例如该会话资源是该 WebRTC服务器 中的第 5个会话资源, 以使主叫终端与被叫终端能够连接到为其分配的会话资源 中。 The WebRTC server can allocate session resources to the connection between the calling terminal and the called terminal on the WebRTC server according to the calling terminal information, the calling routing information, and the call type information carried in the call request, that is, establish a session on the WebRTC server. The session resource that the calling terminal connects to the called terminal. The session resource may include a session resource parameter of the session resource, for example, the session resource is the fifth session resource in the WebRTC server, so that the calling terminal and the called terminal can connect to the session resource allocated to the called terminal.
步骤 103: 生成 WebRTC连接请求, WebRTC连接请求包括 WebRTC服务器 地址以及会话资源的会话资源参数。 Step 103: Generate a WebRTC connection request, where the WebRTC connection request includes a WebRTC server address and a session resource parameter of the session resource.
在本发明的一些实施例中 ,生成 WebRTC连接请求, WebRTC连接请求包括 WebRTC服务器地址以及会话资源的会话资源参数的实现过程可以包括: 生成包 括 URL地址的 WebRTC连接请求, URL地址代表 WebRTC服务器地址以及会话 资源的会话资源参数。 In some embodiments of the present invention, generating a WebRTC connection request, the WebRTC connection request including the WebRTC server address and the session resource parameter of the session resource may include: generating a WebRTC connection request including a URL address, the URL address representing the WebRTC server address, and Session resource parameters for session resources.
即 WebRTC服务器可以为代表主叫终端与被叫终端连接分配的会话资源指定 URL ( Uniform Resource Locator ,统一资源定位符)地址,其中包含了 WebRTC 服务器地址以及会话资源的会话资源参数,这样 WebRTC服务器可以将该 URL地 址放到 WebRTC连接请求中。 That is, the WebRTC server may specify a URL (Uniform Resource Locator) address for the session resource allocated on behalf of the calling terminal and the called terminal connection, and includes the WebRTC server address and the session resource parameter of the session resource, so that the WebRTC server can Place the URL address in the WebRTC connection request.
步骤 104:向电信网关发送 WebRTC连接请求以及被叫终端的电信账号信息, 以使电信网关向被叫终端转发 WebRTC连接请求。 Step 104: Send a WebRTC connection request to the telecommunication gateway and the telecommunication account information of the called terminal, so that the telecommunication gateway forwards the WebRTC connection request to the called terminal.
WebRTC服务器调用电信网关,将 WebRTC连接请求发送给被叫终端, 即 WebRTC服务器调用电信网关将 WebRTC服务器地址以及会话资源的会话资源参 数作为主叫方信息转发给被叫终端。 The WebRTC server invokes the telecommunication gateway to send the WebRTC connection request to the called terminal, that is, the WebRTC server invokes the telecommunication gateway to forward the WebRTC server address and the session resource parameter of the session resource to the called terminal as the calling party information.
需要注意的是, WebRTC服务器调用电信网关向被叫终端发送的 WebRTC连 接请求是电信信令,电信信令具有实时性强的特点,被叫终端无需实时与 WebRTC 服务器相连,也可以通过 WebRTC服务器调用电信网关实时收到 WebRTC连接请 求,从而实现与主叫终端的连接,保证了 WebRTC通信的实时性。 It should be noted that the WebRTC server invokes the WebRTC connection request sent by the telecommunication gateway to the called terminal to be telecommunication signaling. The telecommunication signaling has the characteristics of strong real-time performance. The called terminal does not need to be connected to the WebRTC server in real time, and can also be called through the WebRTC server. The telecom gateway receives the WebRTC connection request in real time, thereby realizing the connection with the calling terminal and ensuring the real-time performance of the WebRTC communication.
步骤 105:建立被叫终端到会话资源的连接,从而建立主叫终端与被叫终端的 连接,其中被叫终端到会话资源的连接是被叫终端根据 WebRTC连接请求发起的。 Step 105: Establish a connection between the called terminal and the session resource, thereby establishing a connection between the calling terminal and the called terminal, where the connection of the called terminal to the session resource is initiated by the called terminal according to the WebRTC connection request.
被叫终端如果同意建立通信,则可以根据 WebRTC连接请求获得 WebRTC服 务器地址以及会话资源的会话资源参数,可以通过打开本机浏览器的方式,连接 的 WebRTC服务器中相应的会话资源中 ,从而主叫终端与被叫终端可以建立连接。 If the called terminal agrees to establish communication, it can obtain the WebRTC service according to the WebRTC connection request. The server address and the session resource parameter of the session resource may be connected to the corresponding session resource in the WebRTC server by opening the local browser, so that the calling terminal and the called terminal can establish a connection.
本方法实施例通过 WebRTC与电信网络的融合通信,将 WebRTC服务器调用 电信网关将 WebRTC连接请求发送给被叫终端,使被叫终端不用一直保持与 WebRTC服务器的连接也能收到 WebRTC连接请求, 以建立主叫终端与被叫终端 的连接,从而保证了 WebRTC通信的实时性。 In the embodiment of the method, the WebRTC server invokes the telecommunication gateway to send the WebRTC connection request to the called terminal through the fusion communication between the WebRTC and the telecommunication network, so that the called terminal can receive the WebRTC connection request without maintaining the connection with the WebRTC server. The connection between the calling terminal and the called terminal is established, thereby ensuring the real-time performance of the WebRTC communication.
在本发明的一些实施例中 ,本发明实施例 WebRTC通信方法可以进一步包括: 接收主叫终端的 WebRTC初始化信息并发送给被叫终端,接收被叫终端的 In some embodiments of the present invention, the WebRTC communication method may further include: receiving the WebRTC initialization information of the calling terminal and transmitting the information to the called terminal, and receiving the called terminal.
WebRTC初始化信息并发送给主叫终端, 以使主叫终端与被叫终端根据 WebRTC 初始化信息完成 WebRTC通信。 The WebRTC initializes the information and sends it to the calling terminal, so that the calling terminal and the called terminal complete the WebRTC communication according to the WebRTC initialization information.
在本发明的一些实施例中 ,接收主叫终端的 WebRTC初始化信息并发送给被 叫终端,接收被叫终端的 WebRTC初始化信息并发送给主叫终端, 以使主叫终端 与被叫终端根据 WebRTC初始化信息完成 WebRTC通信可以具体为 :在被叫终端 选择通过电信网络建立连接时,接收主叫终端的 WebRTC初始化信息并通过电信 网关发送给被叫终端,通过电信网关接收被叫终端的 WebRTC初始化信息并发送 给主叫终端, 以使主叫终端与被叫终端根据 WebRTC初始化信息通过电信网关完 成 WebRTC通信。 In some embodiments of the present invention, the WebRTC initialization information of the calling terminal is received and sent to the called terminal, and the WebRTC initialization information of the called terminal is received and sent to the calling terminal, so that the calling terminal and the called terminal are according to WebRTC. The initialization information completes the WebRTC communication, which may be specifically: when the called terminal selects to establish a connection through the telecommunication network, receives the WebRTC initialization information of the calling terminal and sends the information to the called terminal through the telecommunication gateway, and receives the WebRTC initialization information of the called terminal through the telecommunication gateway. And sending to the calling terminal, so that the calling terminal and the called terminal complete the WebRTC communication through the telecommunication gateway according to the WebRTC initialization information.
即在被叫终端选择通过 WebRTC建立连接时, WebRTC服务器可以直接接收 及转发主叫终端与被叫终端的 WebRTC初始化信息;而在被叫终端选择通过电信 网络建立连接时, WebRTC服务器可以通过电信网关接收及转发主叫终端与被叫 终端的 WebRTC初始化信息。 That is, when the called terminal selects to establish a connection through WebRTC, the WebRTC server can directly receive and forward the WebRTC initialization information of the calling terminal and the called terminal; and when the called terminal selects to establish a connection through the telecommunication network, the WebRTC server can pass the telecommunication gateway. Receive and forward WebRTC initialization information of the calling terminal and the called terminal.
参见图 2所示,是本发明实施例中 WebRTC通信方法实施例 2的流程图,本 实施例可以由 WebRTC服务器实现该方法,可以包括以下步骤: Referring to FIG. 2, it is a flowchart of Embodiment 2 of the WebRTC communication method in the embodiment of the present invention. This embodiment may be implemented by a WebRTC server, and may include the following steps:
步骤 201 : WebRTC服务器接收主叫终端发送的呼叫请求,该呼叫请求为 Web 信令。 Step 201: The WebRTC server receives a call request sent by the calling terminal, where the call request is Web signaling.
步骤 202:判断呼叫请求中是否包含被叫终端的电信账号信息,如果是,进入 步骤 203 ,如果否 ,进入步骤 204。 Step 202: Determine whether the call request contains the telecom account information of the called terminal. If yes, go to step 203. If no, go to step 204.
步骤 203: 直接获得被叫终端的电信账号信息。 Step 203: directly obtain the telecommunication account information of the called terminal.
步骤 204:査找被叫终端的 WebRTC账号信息与被叫终端的电信账号信息的 映射关系 ,获得被叫终端的电信账号信息。 Step 204: Find a mapping relationship between the WebRTC account information of the called terminal and the telecommunication account information of the called terminal, and obtain the telecommunication account information of the called terminal.
步骤 205:根据呼叫请求中携带的主叫终端信息、 主叫路由信息、 呼叫类型信 息,在 WebRTC服务器上建立主叫终端与被叫终端连接的会话资源。 Step 205: According to the calling terminal information, the calling routing information, and the call type information carried in the call request The session resource of the calling terminal and the called terminal is established on the WebRTC server.
步骤 206: 生成 WebRTC连接请求, WebRTC连接请求包括 WebRTC服务器 地址以及会话资源的会话资源参数。 Step 206: Generate a WebRTC connection request, where the WebRTC connection request includes a WebRTC server address and a session resource parameter of the session resource.
步骤 207:向电信网关发送 WebRTC连接请求以及被叫终端的电信账号信息, 以使电信网关向被叫终端转发 WebRTC连接请求。 Step 207: Send a WebRTC connection request to the telecommunication gateway and the telecommunication account information of the called terminal, so that the telecommunication gateway forwards the WebRTC connection request to the called terminal.
步骤 208:建立被叫终端到会话资源的连接,从而建立主叫终端与被叫终端的 连接,其中被叫终端到会话资源的连接是被叫终端根据 WebRTC连接请求发起的。 Step 208: Establish a connection between the called terminal and the session resource, thereby establishing a connection between the calling terminal and the called terminal, where the connection of the called terminal to the session resource is initiated by the called terminal according to the WebRTC connection request.
步骤 209:在被叫终端选择通过 WebRTC建立连接时,接收主叫终端的 Step 209: Receive the calling terminal when the called terminal selects to establish a connection through the WebRTC.
WebRTC初始化信息并发送给被叫终端,接收被叫终端的 WebRTC初始化信息并 发送给主叫终端, 以使主叫终端与被叫终端根据 WebRTC初始化信息完成 The WebRTC initializes the information and sends it to the called terminal, receives the WebRTC initialization information of the called terminal and sends it to the calling terminal, so that the calling terminal and the called terminal complete according to the WebRTC initialization information.
WebRTC通信。 WebRTC communication.
主叫终端与被叫终端可以通过 WebRTC服务器完成 WebRTC连接所需的初始 化流程和信息交换,可以包括交换主叫终端与被叫终端的 SDP( Session Description Protocol ,会话描述协议)许可(或其他类似信令)、 主叫终端与被叫终端的 ip地 址、 参与通信的设备列表(如视频、 音频)、 媒体格式、 网络穿透协议(如 ice ) 和网络穿透服务器信息(如 google-ice )等信息。 主叫终端与被叫终端的通过交换 的 SDP信息,可以完成 WebRTC音视频或数据的通信。 The initial process and information exchange required for the calling terminal and the called terminal to complete the WebRTC connection through the WebRTC server may include exchanging SDP (Session Description Protocol) licenses of the calling terminal and the called terminal (or other similar letters). Order), the IP address of the calling terminal and the called terminal, the list of devices participating in the communication (such as video, audio), media format, network penetration protocol (such as ice) and network penetration server information (such as google-ice) information. The SDP information exchanged between the calling terminal and the called terminal can complete the communication of the WebRTC audio, video or data.
在通话期间 , WebRTC服务器还可以控制通话,如重新协定媒体信息,或挂 机等。 即在 WebRTC服务器在主叫终端与被叫终端在建立连接时, 以及在通话过 程中 ,可以协助通信双方传输信令信息,如双方的连接许可、 协定媒体信息、 挂 机等信令,这些信令均可以由 WebRTC服务器帮助传输。 During a call, the WebRTC server can also control calls, such as re-negotiating media information, or hanging up. That is, when the calling terminal establishes a connection with the called terminal and during the call, the WebRTC server can assist the communication party to transmit signaling information, such as connection permission, protocol media information, and on-hook signaling of the two parties. Both can be transmitted by the WebRTC server.
步骤 210:在被叫终端选择通过电信网络建立连接时,接收主叫终端的 Step 210: Receive the calling terminal when the called terminal selects to establish a connection through the telecommunication network.
WebRTC初始化信息并通过电信网关发送给被叫终端,通过电信网关接收被叫终 端的 WebRTC初始化信息并发送给主叫终端, 以使主叫终端与被叫终端根据 WebRTC初始化信息通过电信网关完成 WebRTC通信。 The WebRTC initialization information is sent to the called terminal through the telecommunication gateway, and the WebRTC initialization information of the called terminal is received by the telecommunication gateway and sent to the calling terminal, so that the calling terminal and the called terminal complete the WebRTC communication through the telecommunication gateway according to the WebRTC initialization information. .
当被叫终端不支持 WebRTC功能时,电信网关可以设置 WebRTC客户端代理 模块, WebRTC服务器依然完成的是 WebRTC连接所需的初始化流程和信息交换 功能,不同之处在于 WebRTC服务器不与被叫终端直接进行信息交换,而是经过 电信网关中的 WebRTC客户端代理模块转发与被叫终端直接进行信息交换。 从而 使主叫终端与被叫终端通过电信网关完成 WebRTC通信。 When the called terminal does not support the WebRTC function, the telecom gateway can set the WebRTC client proxy module. The WebRTC server still completes the initialization process and information exchange function required for the WebRTC connection. The difference is that the WebRTC server is not directly connected to the called terminal. The information exchange is performed, and the information exchange is directly performed with the called terminal through the WebRTC client proxy module in the telecommunication gateway. Thereby, the calling terminal and the called terminal complete the WebRTC communication through the telecommunication gateway.
本方法实施例与方法实施例 1相比,进一步包括了 WebRTC服务器进行 WebRTC初始化信息交换的过程, 以使主叫终端与被叫终端完成 WebRTC通信, 即完成主叫终端与被叫终端之间的音视频或数据的传输。 Compared with the method embodiment 1, the method embodiment further includes a WebRTC server. The WebRTC initializes the process of information exchange, so that the calling terminal and the called terminal complete the WebRTC communication, that is, complete the transmission of audio, video or data between the calling terminal and the called terminal.
参见图 3所示,是本发明实施例中 WebRTC通信方法实施例 3的流程图,本 实施例可以由电信网关实现该方法,可以包括以下步骤: Referring to FIG. 3, it is a flowchart of Embodiment 3 of the WebRTC communication method in the embodiment of the present invention. This embodiment may be implemented by a telecommunications gateway, and may include the following steps:
步骤 301 : 电信网关接收 WebRTC服务器发送的 WebRTC连接请求以及被叫 终端的电信账号信息,被叫终端的电信账号信息是 WebRTC服务器根据主叫终端 发送的呼叫请求获得的, WebRTC连接请求是 WebRTC服务器根据主叫终端发送 的 Web信令形式的呼叫请求生成的,包括 WebRTC服务器地址以及会话资源参数。 Step 301: The telecommunication gateway receives the WebRTC connection request sent by the WebRTC server and the telecommunication account information of the called terminal. The telecommunication account information of the called terminal is obtained by the WebRTC server according to the call request sent by the calling terminal, and the WebRTC connection request is based on the WebRTC server. The call request generated by the calling terminal in the form of web signaling includes the WebRTC server address and the session resource parameter.
WebRTC服务器通过电信网关的开放接口 (如 SIP接口)可以调用电信网关, 电信网关可以接收 WebRTC服务器发送的 WebRTC连接请求以及被叫终端的电信 账号信息。 The WebRTC server can invoke the telecommunication gateway through the open interface of the telecommunication gateway (such as the SIP interface), and the telecommunication gateway can receive the WebRTC connection request sent by the WebRTC server and the telecommunication account information of the called terminal.
在本发明的一些实施例中 , WebRTC连接请求包括代表 WebRTC服务器地址 以及会话资源的会话资源参数的统一资源定位符 URL地址。 WebRTC连接请求的 生成可以由 WebRTC服务器完成,可以参见本发明 WebRTC通信方法实施例 1的 相应部分,此处不再赘述。 In some embodiments of the invention, the WebRTC connection request includes a Uniform Resource Locator URL address representing the WebRTC server address and session resource parameters of the session resource. The generation of the WebRTC connection request can be performed by the WebRTC server. For details, refer to the corresponding part of the WebRTC communication method embodiment 1 of the present invention, and details are not described herein again.
步骤 302:向被叫终端发送 WebRTC连接请求,以使被叫终端连接到 WebRTC 服务器中的会话资源,从而建立主叫终端与被叫终端的连接,其中被叫终端到会 话资源的连接是被叫终端根据 WebRTC连接请求发起的,会话资源是 WebRTC服 务器根据呼叫请求中携带的主叫终端信息、 主叫路由信息、 呼叫类型信息,在 WebRTC服务器上为主叫终端与被叫终端连接分配的。 Step 302: Send a WebRTC connection request to the called terminal, so that the called terminal connects to the session resource in the WebRTC server, thereby establishing a connection between the calling terminal and the called terminal, where the connection of the called terminal to the session resource is called The terminal is initiated according to the WebRTC connection request, and the session resource is allocated by the WebRTC server according to the calling terminal information, the calling routing information, and the call type information carried in the call request, and is connected to the called terminal on the WebRTC server.
电信网关可以完成认证、 鉴权,通过电信网络中的信令管理通道将 WebRTC 连接请求推送给被叫终端,即 WebRTC连接请求为电信信令。 The telecommunication gateway can complete the authentication and authentication, and push the WebRTC connection request to the called terminal through the signaling management channel in the telecommunication network, that is, the WebRTC connection request is telecommunication signaling.
本方法实施例通过 WebRTC与电信网络的融合通信,利用电信网关将 The method embodiment of the method uses the communication communication between the WebRTC and the telecommunication network, and utilizes the telecommunication gateway.
WebRTC连接请求发送给被叫终端,使被叫终端不用一直保持与 WebRTC服务器 的连接也能收到 WebRTC连接请求, 以建立主叫终端与被叫终端的连接,从而保 证了 WebRTC通信的实时性。 The WebRTC connection request is sent to the called terminal, so that the called terminal can receive the WebRTC connection request without establishing the connection with the WebRTC server to establish the connection between the calling terminal and the called terminal, thereby ensuring the real-time performance of the WebRTC communication.
电信网关具有类似来电显示的功能,可以将包括 URL地址的 WebRTC连接请 求( WebRTC服务器的 WebRTC服务器地址以及会话资源参数)通过信令管理通 道发送给被叫终端。 来电显示功能泛指所有在信令通道中传输主叫方信息的方法, 如 BELL202 标准,允许传输 255个字符以内的主叫方信息。 The telecommunication gateway has a function similar to the caller ID, and the WebRTC connection request including the URL address (the WebRTC server address of the WebRTC server and the session resource parameter) can be sent to the called terminal through the signaling management channel. The Caller ID feature refers to all methods of transmitting caller information in the signaling channel, such as the BELL202 standard, which allows the transmission of caller information within 255 characters.
但是,在一些情况下电信网关的来电显示功能只能传输代表主叫方信息的电 话号码,而不能传输多个字符(如包括 URL地址的 WebRTC连接请求) , 因此, 电信网关可以使用编码的方式将电话号码与包括 URL地址的 WebRTC连接请求进 行映射。 However, in some cases, the caller ID function of the telecommunication gateway can only transmit the electricity representing the calling party information. The phone number cannot transmit multiple characters (such as a WebRTC connection request including a URL address), so the telecommunication gateway can use the encoding method to map the phone number with the WebRTC connection request including the URL address.
这样,在本发明的一些实施例中 ,本发明实施例 WebRTC通信方法可以进一 步包括:将 URL地址编码为电话号码;则向被叫终端发送 WebRTC连接请求,包 括: 向被叫终端发送包括电话号码的 WebRTC连接请求。 即电信网关可以向被叫 终端推送 WebRTC连接请求, WebRTC连接请求可以有两种不同形式,一种为包 括 URL地址的 WebRTC连接请求,另一种为包括电话号码的 WebRTC连接请求。 Thus, in some embodiments of the present invention, the WebRTC communication method of the embodiment of the present invention may further include: encoding the URL address into a phone number; and transmitting the WebRTC connection request to the called terminal, including: sending the included phone number to the called terminal WebRTC connection request. That is, the telecommunication gateway can push the WebRTC connection request to the called terminal. The WebRTC connection request can be in two different forms, one is a WebRTC connection request including a URL address, and the other is a WebRTC connection request including a phone number.
另外 , 由于被叫终端可能不支持 WebRTC功能,则可以在电信网关设置 WebRTC客户端代理模块。这样,在本发明的一些实施例中 本发明实施例 WebRTC 通信方法可以进一步包括:在被叫终端选择通过电信网络完成 WebRTC通信时, 建立与被叫终端的连接,并根据 WebRTC连接请求连接到 WebRTC服务器中的会 话资源,以使被叫终端连接到 WebRTC服务器中的会话资源;接收 WebRTC服务 器发送的主叫终端的 WebRTC初始化信息并发送给被叫终端,接收被叫终端的 WebRTC初始化信息并通过 WebRTC服务器发送给主叫终端;对主叫终端与被叫 终端之间发送的数据进行协议转换, 以使主叫终端与被叫终端根据 WebRTC初始 化信息完成 WebRTC通信。 In addition, since the called terminal may not support the WebRTC function, the WebRTC client proxy module may be set at the telecommunications gateway. Thus, in some embodiments of the present invention, the WebRTC communication method of the embodiment of the present invention may further include: establishing a connection with the called terminal when the called terminal selects to complete the WebRTC communication through the telecommunication network, and connecting to the WebRTC according to the WebRTC connection request. a session resource in the server, so that the called terminal is connected to the session resource in the WebRTC server; receives the WebRTC initialization information of the calling terminal sent by the WebRTC server and sends the information to the called terminal, receives the WebRTC initialization information of the called terminal, and passes the WebRTC The server sends the data to the calling terminal; and performs protocol conversion on the data sent between the calling terminal and the called terminal, so that the calling terminal and the called terminal complete the WebRTC communication according to the WebRTC initialization information.
即电信网关启动一个模拟 WebRTC客户端,完成与主叫终端的 WebRTC客户 端的连接,启动一个模拟电信客户端,完成与被叫终端的电信客户端的连接。 主 叫终端发送给被叫终端的数据由电信网关的模拟 WebRTC客户端接收,经过协议 转换后, 由电信网关的模拟电信网关发送给被叫终端。 同样的,被叫终端发送给 主叫终端的数据也通过电信网关进行转发 ,电信网关可以完成 WebRTC协议和电 信协议的转换。 That is, the telecommunication gateway starts an analog WebRTC client, completes the connection with the WebRTC client of the calling terminal, starts an analog telecommunication client, and completes the connection with the telecommunication client of the called terminal. The data sent by the calling terminal to the called terminal is received by the analog WebRTC client of the telecommunication gateway, and after being converted by the protocol, is sent by the analog telecommunication gateway of the telecommunication gateway to the called terminal. Similarly, the data sent by the called terminal to the calling terminal is also forwarded through the telecommunication gateway, and the telecommunication gateway can complete the conversion of the WebRTC protocol and the telecommunication protocol.
参见图 4所示,是本发明实施例中 WebRTC通信方法实施例 4的流程图,本 实施例可以由电信网关实现该方法,可以包括以下步骤: As shown in FIG. 4, it is a flowchart of Embodiment 4 of the WebRTC communication method in the embodiment of the present invention. This embodiment may be implemented by a telecommunication gateway, and may include the following steps:
步骤 401 : 电信网关接收 WebRTC服务器发送的 WebRTC连接请求以及被叫 终端的电信账号信息。 Step 401: The telecommunication gateway receives the WebRTC connection request sent by the WebRTC server and the telecommunication account information of the called terminal.
步骤 402: 向被叫终端发送 WebRTC连接请求。 Step 402: Send a WebRTC connection request to the called terminal.
步骤 403: 当被叫终端通过电信网络完成 WebRTC通信时,建立与被叫终端 的连接,并根据 WebRTC连接请求连接到 WebRTC服务器中的会话资源。 Step 403: When the called terminal completes the WebRTC communication through the telecommunication network, establish a connection with the called terminal, and connect to the session resource in the WebRTC server according to the WebRTC connection request.
步骤 404:接收 WebRTC服务器发送的主叫终端的 WebRTC初始化信息并发 送给被叫终端,接收被叫终端的 WebRTC初始化信息并通过 WebRTC服务器发送 给主叫终端。 Step 404: Receive WebRTC initialization information of the calling terminal sent by the WebRTC server. It is sent to the called terminal, receives the WebRTC initialization information of the called terminal and sends it to the calling terminal through the WebRTC server.
步骤 405:对主叫终端与被叫终端之间发送的数据进行协议转换,以使主叫终 端与被叫终端根据 WebRTC初始化信息完成 WebRTC通信。 Step 405: Perform protocol conversion on the data sent between the calling terminal and the called terminal, so that the calling terminal and the called terminal complete the WebRTC communication according to the WebRTC initialization information.
本方法实施例与方法实施例 3相比,进一步包括了 WebRTC初始化信息交换 的过程,特别是被叫终端选择通过电信网络完成 WebRTC通信时,需要电信网关 将被叫终端与 WebRTC服务器之间的数据、 信息进行转发 , 以使主叫终端与被叫 终端完成 WebRTC通信,从而完成主叫终端与被叫终端之间的音视频或数据的传 输。 Compared with the method embodiment 3, the method embodiment further includes a process of WebRTC initialization information exchange, in particular, when the called terminal selects to complete the WebRTC communication through the telecommunication network, the telecommunication gateway needs the data between the called terminal and the WebRTC server. The information is forwarded, so that the calling terminal and the called terminal complete the WebRTC communication, thereby completing the transmission of audio, video or data between the calling terminal and the called terminal.
参见图 5所示,是本发明实施例中 WebRTC通信方法实施例 5的流程图,本 实施例可以由被叫终端实现该方法,可以包括以下步骤: Referring to FIG. 5, which is a flowchart of Embodiment 5 of the WebRTC communication method in the embodiment of the present invention, the method may be implemented by the called terminal, and may include the following steps:
步骤 501 :被叫终端接收电信网关发送的 WebRTC连接请求; WebRTC连接 请求是 WebRTC服务器根据主叫终端发送的 Web信令形式的呼叫请求生成并发送 给电信网关的,包括 WebRTC服务器地址以及会话资源参数。 Step 501: The called terminal receives the WebRTC connection request sent by the telecommunication gateway. The WebRTC connection request is generated by the WebRTC server according to the call request in the form of Web signaling sent by the calling terminal, and is sent to the telecommunication gateway, including the WebRTC server address and the session resource parameter. .
在本发明的一些实施例中 ,WebRTC连接请求包括统一资源定位符 URL地址, In some embodiments of the invention, the WebRTC connection request includes a uniform resource locator URL address,
URL地址代表 WebRTC服务器地址以及会话资源的会话资源参数,或者,WebRTC 连接请求包括电话号码, 电话号码是电信网关对 URL地址编码获得的。 The URL address represents the WebRTC server address and the session resource parameter of the session resource, or the WebRTC connection request includes a phone number, which is obtained by the telecommunication gateway encoding the URL address.
当 WebRTC连接请求包括电话号码,代表电信网关对 WebRTC连接请求进行 了编码,需要对该电话号码进行解码,被叫终端中可以保存有电话号码与 URL地 址的对应关系,该对应关系可以由被叫终端的电信客户端自带,也可以由用户手 动更新。 When the WebRTC connection request includes a phone number, the telecommunications gateway encodes the WebRTC connection request, and the phone number needs to be decoded. The called terminal can store the correspondence between the phone number and the URL address, and the corresponding relationship can be called by the called party. The telecom client of the terminal comes with it, and can also be manually updated by the user.
这样,在本发明的一些实施例中 ,本发明实施例 WebRTC通信方法进一步可 以包括:当 WebRTC连接请求包括电话号码,对电话号码进行解码,获得包括 URL 地址的 WebRTC连接请求,URL地址代表 WebRTC服务器地址以及会话资源的会 话资源参数。 Thus, in some embodiments of the present invention, the WebRTC communication method of the embodiment of the present invention may further include: when the WebRTC connection request includes a phone number, decoding the phone number, obtaining a WebRTC connection request including a URL address, and the URL address represents a WebRTC server. The address and session resource parameters of the session resource.
步骤 502:根据 WebRTC连接请求连接到 WebRTC服务器中的会话资源,建 立与主叫终端的连接;会话资源是 WebRTC服务器根据主叫终端发送的呼叫请求 为主叫终端与被叫终端分配的。 Step 502: Connect to the session resource in the WebRTC server according to the WebRTC connection request, and establish a connection with the calling terminal; the session resource is allocated by the WebRTC server according to the call request sent by the calling terminal to the calling terminal and the called terminal.
本发明实施例通过 WebRTC与电信网络的融合通信,将 WebRTC连接请求通 过电信网关发送给被叫终端,利用电信网关实现信息推送,使用户终端不用一直 保持与 WebRTC服务器的连接也能收到 WebRTC连接请求,以建立 WebRTC通信, 从而保证了 WebRTC通信的实时性。 In the embodiment of the present invention, the WebRTC connection request is sent to the called terminal through the telecommunication gateway through the fusion communication between the WebRTC and the telecommunication network, and the information push is implemented by using the telecommunication gateway, so that the user terminal can receive the WebRTC connection without always maintaining the connection with the WebRTC server. Request to establish WebRTC communication, Thereby ensuring the real-time nature of WebRTC communication.
在本发明的一些实施例中 ,根据 WebRTC连接请求连接到 WebRTC服务器中 的会话资源,建立与主叫终端的连接的实现过程可以包括:选择通过 WebRTC连 接时,打开 URL地址,连接到 WebRTC服务器中的会话资源,建立与主叫终端的 连接。 In some embodiments of the present invention, according to the WebRTC connection request to connect to the session resource in the WebRTC server, the implementation process of establishing a connection with the calling terminal may include: when connecting through the WebRTC connection, opening the URL address and connecting to the WebRTC server The session resource establishes a connection with the calling terminal.
在本发明的一些实施例中 ,根据 WebRTC连接请求连接到 WebRTC服务器中 的会话资源,建立与主叫终端的连接的实现过程也可以包括:选择通过电信网络 建立连接时,建立与电信网关的连接, 以使电信网关根据 WebRTC连接请求连接 到 WebRTC服务器中的会话资源,连接到 WebRTC服务器中的会话资源;通过电 信网关,建立与主叫终端的连接。 In some embodiments of the present invention, the process of establishing a connection with the calling terminal according to the WebRTC connection request to connect to the session resource in the WebRTC server may also include: establishing a connection with the telecommunication gateway when establishing a connection through the telecommunication network. So that the telecommunication gateway connects to the session resource in the WebRTC server according to the WebRTC connection request, connects to the session resource in the WebRTC server, and establishes a connection with the calling terminal through the telecommunication gateway.
在本发明的一些实施例中 ,本发明实施例 WebRTC通信方法进一步可以包括: 选择通过 WebRTC建立连接时,向 WebRTC服务器发送 WebRTC初始化信息,接 收 WebRTC服务器发送的主叫终端的 WebRTC初始化信息,与主叫终端完成 WebRTC通信;或者,选择通过电信网络建立连接时, 向电信网关发送 WebRTC 初始化信息,接收电信网关发送的主叫终端的 WebRTC初始化信息,与主叫终端 完成 WebRTC通信。 In some embodiments of the present invention, the WebRTC communication method of the embodiment of the present invention may further include: when selecting to establish a connection through the WebRTC, sending the WebRTC initialization information to the WebRTC server, and receiving the WebRTC initialization information of the calling terminal sent by the WebRTC server, and the main The terminal is called to complete the WebRTC communication; or, when the connection is established through the telecommunication network, the WebRTC initialization information is sent to the telecommunication gateway, the WebRTC initialization information of the calling terminal sent by the telecommunication gateway is received, and the WebRTC communication is completed with the calling terminal.
即被叫终端中的通话管理软件在收到 WebRTC连接请求时,能够通过主叫方 信息判断出这是一个 WebRTC连接请求或者普通电话呼叫请求。 如果是 WebRTC 连接请求时,用户可以选择是否直接与主叫终端进行连接,如果是,则可以通过 打开浏览器,打开 URL地址,连接到 WebRTC服务器中的会话资源,建立与主叫 终端的连接,并可以直接通过 WebRTC服务器与主叫终端进行初始化信息的交换, 完成与主叫终端的 WebRTC通信;如果否,则可以与电信网关连接,通过电信网 关连接到 WebRTC服务器中的会话资源,通过电信网关,建立与主叫终端的连接, 通过电信网关与 WebRTC服务器进行主叫终端与被叫终端的初始化信息交换,通 过电信网关完成与主叫终端的 WebRTC通信。 That is, when the call management software in the called terminal receives the WebRTC connection request, it can judge by the calling party information that this is a WebRTC connection request or a normal telephone call request. If it is a WebRTC connection request, the user can choose whether to directly connect with the calling terminal. If yes, the user can open a URL, open a URL address, connect to the session resource in the WebRTC server, and establish a connection with the calling terminal. And the WebRTC server can directly exchange the initialization information with the calling terminal to complete the WebRTC communication with the calling terminal; if not, it can connect with the telecommunication gateway, connect to the session resource in the WebRTC server through the telecommunication gateway, and pass the telecommunication gateway. Establishing a connection with the calling terminal, and performing initialization information exchange between the calling terminal and the called terminal through the telecommunication gateway and the WebRTC server, and completing the WebRTC communication with the calling terminal through the telecommunication gateway.
也就是说,被叫终端选择直接与主叫终端进行连接时,在 WebRTC初始化信 息交换后,被叫终端选择与主叫终端直接通信,是真正的端到端的通信方式;而 被叫终端选择不直接与主叫终端进行连接时,在 WebRTC初始化信息交换后,被 叫终端选择与主叫终端通过电信网关的转发进行通信,不属于严格意义上的端到 端通信。 这样,本发明实施例在被叫终端没有安装有支持 WebRTC的浏览器时, 也可以与主叫终端完成实时通信。 参见图 6所示,是本发明实施例中 WebRTC通信方法实施例 6的流程图,本 实施例可以由被叫终端实现该方法,可以包括以下步骤: That is to say, when the called terminal chooses to directly connect with the calling terminal, after the WebRTC initialization information exchange, the called terminal selects direct communication with the calling terminal, which is a true end-to-end communication mode; and the called terminal selects no When directly connecting with the calling terminal, after the WebRTC initialization information exchange, the called terminal selects to communicate with the calling terminal through the forwarding of the telecommunication gateway, and does not belong to the end-to-end communication in a strict sense. Thus, in the embodiment of the present invention, when the called terminal does not have a browser supporting WebRTC, real-time communication with the calling terminal can also be completed. Referring to FIG. 6, which is a flowchart of Embodiment 6 of the WebRTC communication method in the embodiment of the present invention, the method may be implemented by the called terminal, and may include the following steps:
步骤 601 :被叫终端接收电信网关发送的 WebRTC连接请求。 Step 601: The called terminal receives a WebRTC connection request sent by the telecommunication gateway.
步骤 602:识别 WebRTC连接请求内容。 Step 602: Identify the content of the WebRTC connection request.
步骤 603: 当 WebRTC连接请求内容包括代表 WebRTC服务器地址以及会话 资源参数的 URL地址,获得该 URL地址。 Step 603: Obtain the URL address when the WebRTC connection request content includes a URL address representing a WebRTC server address and a session resource parameter.
步骤 604: 当 WebRTC连接请求包括电话号码,对电话号码进行解码,获得 包括代表 WebRTC服务器地址以及会话资源参数的 URL地址。 Step 604: When the WebRTC connection request includes a phone number, the phone number is decoded to obtain a URL address including a WebRTC server address and a session resource parameter.
步骤 605: 判断是否直接与主叫终端进行连接,如果是,进入步骤 606 , 如 果否,进入步骤 608。 Step 605: Determine whether to directly connect with the calling terminal. If yes, go to step 606. If no, go to step 608.
步骤 606:通过浏览器打开 URL地址,连接到 WebRTC服务器中的会话资源, 建立与主叫终端的连接。 Step 606: Open a URL address through a browser, connect to a session resource in the WebRTC server, and establish a connection with the calling terminal.
步骤 607:向 WebRTC服务器发送 WebRTC初始化信息,接收 WebRTC服务 器发送的主叫终端的 WebRTC初始化信息,与主叫终端完成 WebRTC通信。 Step 607: Send WebRTC initialization information to the WebRTC server, receive WebRTC initialization information of the calling terminal sent by the WebRTC server, and complete WebRTC communication with the calling terminal.
步骤 608:建立与电信网关的连接, 以使电信网关根据 WebRTC连接请求连 接到 WebRTC服务器中的会话资源;通过电信网关,建立与主叫终端的连接。 Step 608: Establish a connection with the telecommunication gateway, so that the telecommunication gateway connects to the session resource in the WebRTC server according to the WebRTC connection request; establish a connection with the calling terminal through the telecommunication gateway.
步骤 609: 向电信网关发送 WebRTC初始化信息,接收电信网关发送的主叫 终端的 WebRTC初始化信息,通过电信网关与主叫终端完成 WebRTC通信。 Step 609: Send WebRTC initialization information to the telecommunication gateway, receive the WebRTC initialization information of the calling terminal sent by the telecommunication gateway, and complete the WebRTC communication with the calling terminal through the telecommunication gateway.
本方法实施例与方法实施例 5相比,进一步包括了被叫终端直接或通过解码 的方式获得 WebRTC连接请求中包括代表 WebRTC服务器地址以及会话资源参数 的 URL地址的过程以及选择通过 WebRTC连接或选择通过电信网络建立连接并与 主叫终端交换 WebRTC初始化信息,完成 WebRTC通信的过程。 本方法实施例从 被叫终端的角度说明了被叫终端与主叫终端建立实时 WebRTC通信的过程。 Compared with the method embodiment 5, the method embodiment further includes the process that the called terminal obtains the URL address representing the WebRTC server address and the session resource parameter in the WebRTC connection request directly or by decoding, and selects to connect or select through the WebRTC. The process of establishing a connection through a telecommunication network and exchanging WebRTC initialization information with the calling terminal to complete the WebRTC communication process. The method embodiment illustrates the process of establishing real-time WebRTC communication between the called terminal and the calling terminal from the perspective of the called terminal.
与上述各个 WebRTC通信方法实施例相对应的,参见图 7所示,本发明实施 例还提供一种 WebRTC通信系统实施例,包括主叫终端 701、 WebRTC服务器 702、 电信网关 703以及被叫终端 704。 The embodiment of the present invention further provides an embodiment of a WebRTC communication system, including a calling terminal 701, a WebRTC server 702, a telecommunications gateway 703, and a called terminal 704, corresponding to the foregoing embodiments of the WebRTC communication method. .
主叫终端 701 ,用于向 WebRTC服务器发送呼叫请求,呼叫请求中可以包含 被叫终端的电信账号信息;连接到 WebRTC为主叫终端与被叫终端连接建立的会 话资源中 , 以建立与被叫终端的连接。 The calling terminal 701 is configured to send a call request to the WebRTC server, where the call request may include the telecommunication account information of the called terminal; and connect to the session resource established by the WebRTC to connect the calling terminal and the called terminal to establish and call Terminal connection.
在本发明的一些实施例中 ,主叫终端还用于: 向 WebRTC服务器发送主叫终 端的 WebRTC初始化信息,接收 WebRTC服务器发送的被叫终端的 WebRTC初始 化信息,完成与被叫终端的 WebRTC通信。 In some embodiments of the present invention, the calling terminal is further configured to: send the WebRTC initialization information of the calling terminal to the WebRTC server, and receive the WebRTC initial of the called terminal sent by the WebRTC server. The information is completed, and the WebRTC communication with the called terminal is completed.
WebRTC服务器 702 ,用于接收主叫终端发送的呼叫请求,呼叫请求为 Web 信令;根据呼叫请求获得被叫终端的电信账号信息,并根据呼叫请求中携带的主 叫终端信息、 主叫路由信息、 呼叫类型信息,在 WebRTC服务器上建立主叫终端 与被叫终端连接的会话资源;生成 WebRTC连接请求, WebRTC连接请求包括 WebRTC服务器地址以及会话资源的会话资源参数; 向电信网关发送 WebRTC连 接请求以及被叫终端的电信账号信息, 以使电信网关向被叫终端转发 WebRTC连 接请求;建立被叫终端到会话资源的连接,从而建立主叫终端与被叫终端的连接, 其中被叫终端到会话资源的连接是被叫终端根据 WebRTC连接请求发起的。 The WebRTC server 702 is configured to receive a call request sent by the calling terminal, and the call request is a Web signaling; obtain the telecommunication account information of the called terminal according to the call request, and according to the calling terminal information and the calling routing information carried in the call request And calling type information, establishing a session resource connected between the calling terminal and the called terminal on the WebRTC server; generating a WebRTC connection request, the WebRTC connection request includes a WebRTC server address and a session resource parameter of the session resource; and sending a WebRTC connection request to the telecommunication gateway and The telecom account information of the called terminal, so that the telecommunication gateway forwards the WebRTC connection request to the called terminal; establishes a connection of the called terminal to the session resource, thereby establishing a connection between the calling terminal and the called terminal, wherein the called terminal reaches the session resource The connection is initiated by the called terminal according to the WebRTC connection request.
在本发明的一些实施例中 ,根据呼叫请求获得被叫终端的电信账号信息,包 括:如果呼叫请求包含被叫终端的电信账号信息,直接获得被叫终端的电信账号 信息;或者,如果呼叫请求包含被叫终端的 WebRTC账号信息,査找被叫终端的 WebRTC账号信息与被叫终端的电信账号信息的映射关系 ,获得被叫终端的电信 账号信息。 In some embodiments of the present invention, obtaining the telecommunication account information of the called terminal according to the call request includes: if the call request includes the telecommunication account information of the called terminal, directly obtaining the telecommunication account information of the called terminal; or, if the call request The WebRTC account information of the called terminal is searched for, and the mapping relationship between the WebRTC account information of the called terminal and the telecom account information of the called terminal is obtained, and the telecom account information of the called terminal is obtained.
在本发明的一些实施例中 ,生成 WebRTC连接请求, WebRTC连接请求包括 In some embodiments of the invention, a WebRTC connection request is generated, and the WebRTC connection request includes
WebRTC服务器地址以及会话资源的会话资源参数,包括: 生成包括统一资源定 位符 URL地址的 WebRTC连接请求, URL地址代表 WebRTC服务器地址以及会 话资源的会话资源参数。 The WebRTC server address and the session resource parameters of the session resource include: generating a WebRTC connection request including a uniform resource locator URL address, the URL address representing a WebRTC server address, and a session resource parameter of the session resource.
在本发明的一些实施例中 ,WebRTC服务器还用于:接收主叫终端的 WebRTC 初始化信息并发送给被叫终端,接收被叫终端的 WebRTC初始化信息并发送给主 叫终端, 以使主叫终端与被叫终端根据 WebRTC初始化信息完成 WebRTC通信, 在本发明的一些实施例中 ,在被叫终端选择通过电信网络建立连接时,接收 主叫终端的 WebRTC初始化信息并发送给被叫终端,接收被叫终端的 WebRTC初 始化信息并发送给主叫终端, 以使主叫终端与被叫终端根据 WebRTC初始化信息 完成 WebRTC通信具体为 :接收主叫终端的 WebRTC初始化信息并通过电信网关 发送给被叫终端,通过电信网关接收被叫终端的 WebRTC初始化信息并发送给主 叫终端, 以使主叫终端与被叫终端根据 WebRTC初始化信息通过电信网关完成 WebRTC通信。 In some embodiments of the present invention, the WebRTC server is further configured to: receive the WebRTC initialization information of the calling terminal and send the information to the called terminal, receive the WebRTC initialization information of the called terminal, and send the information to the calling terminal, so that the calling terminal And the called terminal completes the WebRTC communication according to the WebRTC initialization information. In some embodiments of the present invention, when the called terminal selects to establish a connection through the telecommunication network, the WebRTC initialization information of the calling terminal is received and sent to the called terminal, and the receiving is received. The WebRTC initialization information of the terminal is sent to the calling terminal, so that the calling terminal and the called terminal complete the WebRTC communication according to the WebRTC initialization information, specifically: receiving the WebRTC initialization information of the calling terminal and transmitting the information to the called terminal through the telecommunication gateway. The WebRTC initialization information of the called terminal is received by the telecommunication gateway and sent to the calling terminal, so that the calling terminal and the called terminal complete the WebRTC communication through the telecommunication gateway according to the WebRTC initialization information.
电信网关 703 ,用于接收 WebRTC服务器发送的 WebRTC连接请求以及被叫 终端的电信账号信息,被叫终端的电信账号信息是 WebRTC服务器根据主叫终端 发送的呼叫请求获得的, WebRTC连接请求是 WebRTC服务器根据主叫终端发送 的 Web信令形式的呼叫请求生成的,包括 WebRTC服务器地址以及会话资源参数; 向被叫终端发送 WebRTC连接请求,以使被叫终端连接到 WebRTC服务器中的会 话资源,从而建立主叫终端与被叫终端的连接,其中被叫终端到会话资源的连接 是被叫终端根据 WebRTC连接请求发起的,会话资源是 WebRTC服务器根据呼叫 请求中携带的主叫终端信息、 主叫路由信息、 呼叫类型信息,在 WebRTC服务器 上为主叫终端与被叫终端连接分配的。 The telecommunication gateway 703 is configured to receive the WebRTC connection request sent by the WebRTC server and the telecommunication account information of the called terminal. The telecommunication account information of the called terminal is obtained by the WebRTC server according to the call request sent by the calling terminal, and the WebRTC connection request is the WebRTC server. Send according to the calling terminal Generated by the call request in the form of web signaling, including the WebRTC server address and the session resource parameter; sending a WebRTC connection request to the called terminal, so that the called terminal connects to the session resource in the WebRTC server, thereby establishing the calling terminal and the called terminal The connection of the terminal is called, wherein the connection of the called terminal to the session resource is initiated by the called terminal according to the WebRTC connection request, and the session resource is the calling terminal information, the calling routing information, and the call type information carried by the WebRTC server according to the call request. The call is connected to the called terminal on the WebRTC server.
在本发明的一些实时例中 , WebRTC连接请求包括代表 WebRTC服务器地址 以及会话资源的会话资源参数的统一资源定位符 URL地址。 In some real-time examples of the present invention, the WebRTC connection request includes a Uniform Resource Locator URL address representing the WebRTC server address and session resource parameters of the session resource.
在本发明的一些实施例中 , 电信网关还用于:将 URL地址编码为电话号码; 向被叫终端发送 WebRTC连接请求,包括: 向被叫终端发送包括电话号码的 WebRTC连接请求。 In some embodiments of the present invention, the telecommunication gateway is further configured to: encode the URL address into a phone number; and send the WebRTC connection request to the called terminal, including: sending a WebRTC connection request including the phone number to the called terminal.
在本发明的一些实施例中 , 电信网关还用于:在被叫终端选择通过电信网络 完成 WebRTC通信时,建立与被叫终端的连接,并根据 WebRTC连接请求连接到 WebRTC服务器中的会话资源, 以使被叫终端连接到 WebRTC服务器中的会话资 源;接收 WebRTC服务器发送的主叫终端的 WebRTC初始化信息并发送给被叫终 端,接收被叫终端的 WebRTC初始化信息并通过 WebRTC服务器发送给主叫终端; 对主叫终端与被叫终端之间发送的数据进行协议转换, 以使主叫终端与被叫终端 根据 WebRTC初始化信息完成 WebRTC通信。 In some embodiments of the present invention, the telecommunication gateway is further configured to: when the called terminal selects to complete the WebRTC communication through the telecommunication network, establish a connection with the called terminal, and connect to the session resource in the WebRTC server according to the WebRTC connection request, To enable the called terminal to connect to the session resource in the WebRTC server; receive the WebRTC initialization information of the calling terminal sent by the WebRTC server and send it to the called terminal, receive the WebRTC initialization information of the called terminal, and send it to the calling terminal through the WebRTC server. Translating data transmitted between the calling terminal and the called terminal to enable the calling terminal and the called terminal to complete WebRTC communication according to the WebRTC initialization information.
被叫终端 704 ,用于接收电信网关发送的 WebRTC连接请求; WebRTC连接 请求是 WebRTC服务器根据主叫终端发送的 Web信令形式的呼叫请求生成并发送 给电信网关的,包括 WebRTC服务器地址以及会话资源参数;根据 WebRTC连接 请求连接到 WebRTC服务器中的会话资源,建立与主叫终端的连接;会话资源是 WebRTC服务器根据主叫终端发送的呼叫请求为主叫终端与被叫终端分配的。 The called terminal 704 is configured to receive a WebRTC connection request sent by the telecommunication gateway; the WebRTC connection request is generated by the WebRTC server according to the call request in the form of Web signaling sent by the calling terminal, and sent to the telecommunication gateway, including the WebRTC server address and the session resource. The parameter is connected to the session resource in the WebRTC server according to the WebRTC connection request, and establishes a connection with the calling terminal; the session resource is allocated by the WebRTC server to the calling terminal and the called terminal according to the call request sent by the calling terminal.
在本发明的一些实施例中 ,WebRTC连接请求包括统一资源定位符 URL地址, URL地址代表 WebRTC服务器地址以及会话资源的会话资源参数,或者,WebRTC 连接请求包括电话号码, 电话号码是电信网关对 URL地址编码获得的。 In some embodiments of the present invention, the WebRTC connection request includes a uniform resource locator URL address, the URL address represents a WebRTC server address, and a session resource parameter of the session resource, or the WebRTC connection request includes a phone number, and the phone number is a telecommunication gateway to the URL. Address code obtained.
在本发明的一些实施例中 ,被叫终端还用于: 当 WebRTC连接请求包括电话 号码,对电话号码进行解码,获得包括 URL地址的 WebRTC连接请求, URL地 址代表 WebRTC服务器地址以及会话资源的会话资源参数。 In some embodiments of the present invention, the called terminal is further configured to: when the WebRTC connection request includes a phone number, decode the phone number, obtain a WebRTC connection request including a URL address, the URL address represents a WebRTC server address, and a session resource session Resource parameters.
在本发明的一些实施例中 ,根据 WebRTC连接请求连接到 WebRTC服务器中 的会话资源,建立与主叫终端的连接,包括:选择通过 WebRTC连接时,打开 URL 地址,连接到 WebRTC服务器中的会话资源,建立与主叫终端的连接。 在本发明的一些实施例中 ,根据 WebRTC连接请求连接到 WebRTC服务器中 的会话资源,建立与主叫终端的连接,包括:选择通过电信网络建立连接时,建 立与电信网关的连接,以使电信网关根据 WebRTC连接请求连接到 WebRTC服务 器中的会话资源;通过电信网关,建立与主叫终端的连接。 In some embodiments of the present invention, a connection to a calling terminal is established according to a WebRTC connection request to connect to a session resource in the WebRTC server, including: selecting a URL to open when connecting via WebRTC The address, connected to the session resource in the WebRTC server, establishes a connection with the calling terminal. In some embodiments of the present invention, the connection to the calling terminal is established according to the WebRTC connection request to connect to the session resource in the WebRTC server, including: when establishing a connection through the telecommunication network, establishing a connection with the telecommunication gateway to enable telecommunication The gateway connects to the session resource in the WebRTC server according to the WebRTC connection request; establishes a connection with the calling terminal through the telecommunication gateway.
在本发明的一些实施例中 ,被叫终端还用于:选择通过 WebRTC建立连接时, 向 WebRTC服务器发送 WebRTC初始化信息,接收 WebRTC服务器发送的主叫终 端的 WebRTC初始化信息,与主叫终端完成 WebRTC通信;或者,选择通过电信 网络建立连接时, 向电信网关发送 WebRTC初始化信息,接收电信网关发送的主 叫终端的 WebRTC初始化信息,与主叫终端完成 WebRTC通信。 In some embodiments of the present invention, the called terminal is further configured to: when the connection is established through the WebRTC, send the WebRTC initialization information to the WebRTC server, receive the WebRTC initialization information of the calling terminal sent by the WebRTC server, and complete the WebRTC with the calling terminal. Or; when selecting a connection through the telecommunication network, sending WebRTC initialization information to the telecommunication gateway, receiving WebRTC initialization information of the calling terminal sent by the telecommunication gateway, and completing WebRTC communication with the calling terminal.
结合图 8所示的信令交互示意图,对上述各个部分所起作用以及各部分间的 信息交互过程进行简单介绍。 Referring to the signaling interaction diagram shown in FIG. 8, the function of each part mentioned above and the information interaction process between the parts are briefly introduced.
步骤 801 :主叫终端向 WebRTC服务器发送呼叫请求,该呼叫请求为 Web信 令 步骤 802: WebRTC服务器根据呼叫请求获得被叫终端的电信账号信息。 Step 801: The calling terminal sends a call request to the WebRTC server, where the call request is a Web signaling. Step 802: The WebRTC server obtains the telecom account information of the called terminal according to the call request.
步骤 803: WebRTC服务器根据呼叫请求中携带的主叫终端信息、 主叫路由信 息、 呼叫类型信息,在 WebRTC服务器上建立主叫终端与被叫终端连接的会话资 源。 Step 803: The WebRTC server establishes session resources connected between the calling terminal and the called terminal on the WebRTC server according to the calling terminal information, the calling routing information, and the call type information carried in the call request.
步骤 804: WebRTC服务器生成 WebRTC连接请求,调用电信网关, 向电信 网关发送 WebRTC连接请求以及被叫终端的电信账号信息。 Step 804: The WebRTC server generates a WebRTC connection request, invokes a telecommunication gateway, and sends a WebRTC connection request to the telecommunication gateway and the telecommunication account information of the called terminal.
步骤 805: 电信网关通过认证、 鉴权,准备呼叫被叫终端,通过信令通道将 WebRTC连接请求推送给被叫终端。 Step 805: The telecommunication gateway prepares to call the called terminal through authentication and authentication, and pushes the WebRTC connection request to the called terminal through the signaling channel.
步骤 806:被叫终端接收 WebRTC连接请求,根据 WebRTC连接请求,判断 是否直接与主叫终端进行连接,如果是, 即选择通过 WebRTC建立连接,进入步 骤 807 ,如果否,即选择通过电信网络建立连接,进入步骤 809。 Step 806: The called terminal receives the WebRTC connection request, and determines whether to directly connect with the calling terminal according to the WebRTC connection request. If yes, the connection is established through WebRTC, and the process proceeds to step 807. If not, the connection is established through the telecommunication network. Go to step 809.
步骤 807:被叫终端通过浏览器打开 URL地址,连接到 WebRTC服务器中的 会话资源,建立与主叫终端的连接。 Step 807: The called terminal opens the URL address through the browser, connects to the session resource in the WebRTC server, and establishes a connection with the calling terminal.
步骤 808: WebRTC服务器交换主叫终端与被叫终端的 WebRTC初始化信息, 完成主叫终端与被叫终端的 WebRTC通信。 Step 808: The WebRTC server exchanges the WebRTC initialization information of the calling terminal and the called terminal, and completes the WebRTC communication between the calling terminal and the called terminal.
步骤 809:被叫终端建立与电信网关的连接, 电信网关根据 WebRTC连接请 求连接到 WebRTC服务器中的会话资源;通过电信网关建立被叫终端与主叫终端 的连接。 Step 809: The called terminal establishes a connection with the telecommunication gateway, and the telecommunication gateway connects to the session resource in the WebRTC server according to the WebRTC connection request; establishes the called terminal and the calling terminal through the telecommunication gateway. Connection.
步骤 810: WebRTC服务器通过电信网关交换主叫终端与被叫终端的 WebRTC 初始化信息,通过电信网关完成主叫终端与被叫终端的 WebRTC通信。 Step 810: The WebRTC server exchanges the WebRTC initialization information of the calling terminal and the called terminal through the telecommunication gateway, and completes the WebRTC communication between the calling terminal and the called terminal through the telecommunication gateway.
参见图 9所示,是本发明实施例中 WebRTC服务器实施例的示意图 ,可以包 括: FIG. 9 is a schematic diagram of an embodiment of a WebRTC server according to an embodiment of the present invention, which may include:
接收单元 901 ,用于 WebRTC服务器接收主叫终端发送的呼叫请求,呼叫请 求为 Web信令; The receiving unit 901 is configured to receive, by the WebRTC server, a call request sent by the calling terminal, where the call request is a Web signaling;
建立单元 902 ,用于根据呼叫请求获得被叫终端的电信账号信息,并根据呼叫 请求中携带的主叫终端信息、 主叫路由信息、 呼叫类型信息,在 WebRTC服务器 上建立主叫终端与被叫终端连接的会话资源; The establishing unit 902 is configured to obtain the telecommunication account information of the called terminal according to the call request, and establish the calling terminal and the called party on the WebRTC server according to the calling terminal information, the calling routing information, and the call type information carried in the call request. Session resources connected by the terminal;
生成单元 903 ,用于生成 WebRTC连接请求,WebRTC连接请求包括 WebRTC 服务器地址以及会话资源的会话资源参数; a generating unit 903, configured to generate a WebRTC connection request, where the WebRTC connection request includes a WebRTC server address and a session resource parameter of the session resource;
发送单元 904 ,用于向电信网关发送生成单元生成的 WebRTC连接请求以及 建立单元建立的被叫终端的电信账号信息,以使电信网关向被叫终端转发 WebRTC 连接请求; The sending unit 904 is configured to send, to the telecommunication gateway, the WebRTC connection request generated by the generating unit and the telecommunication account information of the called terminal established by the establishing unit, so that the telecommunication gateway forwards the WebRTC connection request to the called terminal;
连接单元 905 ,建立被叫终端到会话资源的连接,从而建立主叫终端与被叫终 端的连接,其中被叫终端到会话资源的连接是被叫终端根据发送单元发送的 WebRTC连接请求发起的。 The connecting unit 905 establishes a connection between the called terminal and the session resource, thereby establishing a connection between the calling terminal and the called terminal, wherein the connection of the called terminal to the session resource is initiated by the called terminal according to the WebRTC connection request sent by the sending unit.
在本发明的一些实施例中 ,建立单元可以具体用于: In some embodiments of the invention, the establishing unit may be specifically configured to:
如果接收单元接收的呼叫请求包含被叫终端的电信账号信息,直接获得被叫 终端的电信账号信息,建立主叫终端与被叫终端连接的会话资源; If the call request received by the receiving unit includes the telecommunication account information of the called terminal, directly obtain the telecommunication account information of the called terminal, and establish a session resource that the calling terminal connects with the called terminal;
或者,如果接收单元接收的呼叫请求包含被叫终端的 WebRTC账号信息,査 找被叫终端的 WebRTC账号信息与被叫终端的电信账号信息的映射关系 ,获得被 叫终端的电信账号信息,建立主叫终端与被叫终端连接的会话资源。 Alternatively, if the call request received by the receiving unit includes the WebRTC account information of the called terminal, the mapping relationship between the WebRTC account information of the called terminal and the telecommunication account information of the called terminal is obtained, the telecommunication account information of the called terminal is obtained, and the calling party is established. The session resource that the terminal connects to the called terminal.
在本发明的一些实施例中 ,生成单元可以具体用于: In some embodiments of the present invention, the generating unit may be specifically configured to:
生成包括统一资源定位符 URL地址的 WebRTC连接请求, URL地址代表 WebRTC服务器地址以及建立单元建立的会话资源的会话资源参数。 A WebRTC connection request including a Uniform Resource Locator URL address is generated, and the URL address represents a WebRTC server address and a session resource parameter of the session resource established by the establishment unit.
在本发明的一些实施例中 ,本发明实施例 WebRTC服务器还可以包括: 初始化单元,接收主叫终端的 WebRTC初始化信息并发送给被叫终端,接收 被叫终端的 WebRTC初始化信息并发送给主叫终端, 以使主叫终端与被叫终端根 据 WebRTC初始化信息完成 WebRTC通信。 在本发明的一些实施例中 ,在被叫终端选择通过电信网络建立连接时,初始 化单元具体用于:在被叫终端选择通过电信网络建立连接时,接收主叫终端的In some embodiments of the present invention, the WebRTC server of the embodiment of the present invention may further include: an initialization unit that receives the WebRTC initialization information of the calling terminal and sends the information to the called terminal, receives the WebRTC initialization information of the called terminal, and sends the information to the calling party. The terminal, so that the calling terminal and the called terminal complete the WebRTC communication according to the WebRTC initialization information. In some embodiments of the present invention, when the called terminal selects to establish a connection through the telecommunication network, the initializing unit is specifically configured to: when the called terminal selects to establish a connection through the telecommunication network, receives the calling terminal.
WebRTC初始化信息并通过电信网关发送给被叫终端,通过电信网关接收被叫终 端的 WebRTC初始化信息并发送给主叫终端, 以使主叫终端与被叫终端根据 WebRTC初始化信息通过电信网关完成 WebRTC通信。 The WebRTC initialization information is sent to the called terminal through the telecommunication gateway, and the WebRTC initialization information of the called terminal is received by the telecommunication gateway and sent to the calling terminal, so that the calling terminal and the called terminal complete the WebRTC communication through the telecommunication gateway according to the WebRTC initialization information. .
参见图 10所示,是本发明实施例中电信网关实施例的示意图 ,可以包括: 接收单元 1001 ,用于接收 WebRTC服务器发送的 WebRTC连接请求以及被叫 终端的电信账号信息,被叫终端的电信账号信息是 WebRTC服务器根据主叫终端 发送的呼叫请求获得的, WebRTC连接请求是 WebRTC服务器根据主叫终端发送 的 Web信令形式的呼叫请求生成的,包括 WebRTC服务器地址以及会话资源参数; 发送单元 1002 ,用于向被叫终端发送接收单元接收的 WebRTC连接请求,以 使被叫终端连接到 WebRTC服务器中的会话资源,从而建立主叫终端与被叫终端 的连接,其中被叫终端到会话资源的连接是被叫终端根据 WebRTC连接请求发起 的,会话资源是 WebRTC服务器根据呼叫请求中携带的主叫终端信息、 主叫路由 信息、 呼叫类型信息,在 WebRTC服务器上为主叫终端与被叫终端连接分配的。 As shown in FIG. 10, it is a schematic diagram of an embodiment of a telecommunication gateway in an embodiment of the present invention, which may include: a receiving unit 1001, configured to receive a WebRTC connection request sent by a WebRTC server, and a telecommunication account information of the called terminal, and a telecommunication of the called terminal. The account information is obtained by the WebRTC server according to the call request sent by the calling terminal, and the WebRTC connection request is generated by the WebRTC server according to the call request in the form of Web signaling sent by the calling terminal, including the WebRTC server address and the session resource parameter; the sending unit 1002 And sending, by the called terminal, a WebRTC connection request received by the receiving unit, so that the called terminal connects to the session resource in the WebRTC server, thereby establishing a connection between the calling terminal and the called terminal, where the called terminal reaches the session resource. The connection is initiated by the called terminal according to the WebRTC connection request, and the session resource is the calling terminal information, the calling routing information, and the call type information carried by the WebRTC server according to the call request, and the calling terminal is connected to the called terminal on the WebRTC server. distributed.
在本发明的一些实施例中 , WebRTC连接请求可以包括代表 WebRTC服务器 地址以及会话资源参数的统一资源定位符 URL地址。 In some embodiments of the invention, the WebRTC connection request may include a Uniform Resource Locator URL address representing the WebRTC server address and session resource parameters.
在本发明的一些实施例中 ,本发明实施例中电信网关还可以包括: In some embodiments of the present invention, the telecommunication gateway in the embodiment of the present invention may further include:
编码单元,用于将 URL地址编码为电话号码; a coding unit for encoding a URL address as a phone number;
发送单元可以具体用于: 向被叫终端发送编码单元编码的包括电话号码的 The sending unit may be specifically configured to: send, to the called terminal, the coding unit code including the phone number
WebRTC连接请求。 WebRTC connection request.
在本发明的一些实施例中 ,本发明实施例中电信网关还可以包括: In some embodiments of the present invention, the telecommunication gateway in the embodiment of the present invention may further include:
代理单元,用于在被叫终端选择通过电信网络完成 WebRTC通信时,建立与 被叫终端的连接,并根据 WebRTC连接请求连接到 WebRTC服务器中的会话资源, 以使被叫终端连接到 WebRTC服务器中的会话资源; The proxy unit is configured to establish a connection with the called terminal when the called terminal selects to complete the WebRTC communication through the telecommunication network, and connect to the session resource in the WebRTC server according to the WebRTC connection request, so that the called terminal is connected to the WebRTC server. Conversational resources;
初始化单元,用于接收 WebRTC服务器发送的主叫终端的 WebRTC初始化信 息并发送给被叫终端,接收被叫终端的 WebRTC初始化信息并通过 WebRTC服务 器发送给主叫终端; The initialization unit is configured to receive the WebRTC initialization information of the calling terminal sent by the WebRTC server, and send the information to the called terminal, receive the WebRTC initialization information of the called terminal, and send the information to the calling terminal through the WebRTC server;
代理单元,还可以用于对主叫终端与被叫终端之间发送的数据进行协议转换, 以使主叫终端与被叫终端根据 WebRTC初始化信息完成 WebRTC通信。 The proxy unit may also be configured to perform protocol conversion on the data sent between the calling terminal and the called terminal, so that the calling terminal and the called terminal complete the WebRTC communication according to the WebRTC initialization information.
参见图 11所示,是本发明实施例中终端实施例的示意图 ,该终端可以为被叫 终端,可以包括: FIG. 11 is a schematic diagram of an embodiment of a terminal in the embodiment of the present invention, where the terminal may be called The terminal can include:
接收单元 1101 ,用于接收电信网关发送的 WebRTC连接请求; WebRTC连接 请求是 WebRTC服务器根据主叫终端发送的 Web信令形式的呼叫请求生成并发送 给电信网关的,包括 WebRTC服务器地址以及会话资源参数; The receiving unit 1101 is configured to receive a WebRTC connection request sent by the telecommunication gateway; the WebRTC connection request is generated by the WebRTC server according to the call request in the form of Web signaling sent by the calling terminal, and sent to the telecommunication gateway, including the WebRTC server address and the session resource parameter. ;
连接单元 1102 ,用于根据接收单元接收的 WebRTC连接请求连接到 WebRTC 服务器中的会话资源,建立与主叫终端的连接;会话资源是 WebRTC服务器根据 主叫终端发送的呼叫请求为主叫终端与被叫终端分配的。 The connecting unit 1102 is configured to establish a connection with the calling terminal according to the WebRTC connection request received by the receiving unit and connect to the session resource in the WebRTC server; the session resource is the WebRTC server, the calling terminal and the called terminal according to the call request sent by the calling terminal. Called by the terminal.
在本发明的一些实施例中 , WebRTC连接请求可以包括统一资源定位符 URL 地址, URL地址代表 WebRTC服务器地址以及会话资源的会话资源参数,或者, WebRTC连接请求包括电话号码, 电话号码是电信网关对 URL地址编码获得的。 In some embodiments of the present invention, the WebRTC connection request may include a uniform resource locator URL address, the URL address represents a WebRTC server address and a session resource parameter of the session resource, or the WebRTC connection request includes a phone number, and the phone number is a telecommunication gateway pair. URL address encoding obtained.
在本发明的一些实施例中 ,本发明实施例中终端还可以包括: In some embodiments of the present invention, the terminal in the embodiment of the present invention may further include:
解码单元,用于当 WebRTC连接请求包括电话号码,对电话号码进行解码, 获得包括 URL地址的 WebRTC连接请求, URL地址代表 WebRTC服务器地址以 及会话资源的会话资源参数。 The decoding unit is configured to: when the WebRTC connection request includes a phone number, decode the phone number, obtain a WebRTC connection request including a URL address, the URL address represents a WebRTC server address, and a session resource parameter of the session resource.
在本发明的一些实施例中 ,连接单元可以具体用于: In some embodiments of the invention, the connecting unit may be specifically configured to:
选择通过 WebRTC连接时,打开接收单元接收的或解码单元解码的 URL地址, 连接到 WebRTC服务器中的会话资源,建立与主叫终端的连接。 When the connection through WebRTC is selected, the URL address received by the receiving unit or decoded by the decoding unit is opened, and the session resource in the WebRTC server is connected to establish a connection with the calling terminal.
在本发明的一些实施例中 ,连接单元可以具体用于: In some embodiments of the invention, the connecting unit may be specifically configured to:
选择通过电信网络建立连接时,建立与电信网关的连接, 以使电信网关根据 WebRTC连接请求连接到 WebRTC服务器中的会话资源;通过电信网关,建立与 主叫终端的连接。 When the connection is established through the telecommunication network, a connection with the telecommunication gateway is established, so that the telecommunication gateway connects to the session resource in the WebRTC server according to the WebRTC connection request; and establishes a connection with the calling terminal through the telecommunication gateway.
在本发明的一些实施例中 ,本发明实施例中终端还可以包括: In some embodiments of the present invention, the terminal in the embodiment of the present invention may further include:
初始化单元,用于选择通过 WebRTC建立连接时, 向 WebRTC服务器发送 WebRTC初始化信息,接收 WebRTC服务器发送的主叫终端的 WebRTC初始化信 息,与主叫终端完成 WebRTC通信;或者,选择通过电信网络建立连接时, 向电 信网关发送 WebRTC初始化信息,接收电信网关发送的主叫终端的 WebRTC初始 化信息,与主叫终端完成 WebRTC通信。 The initializing unit is configured to: when the connection is established through the WebRTC, send the WebRTC initialization information to the WebRTC server, receive the WebRTC initialization information of the calling terminal sent by the WebRTC server, complete the WebRTC communication with the calling terminal; or select the connection established through the telecommunication network. Sending WebRTC initialization information to the telecommunication gateway, receiving the WebRTC initialization information of the calling terminal sent by the telecommunication gateway, and completing the WebRTC communication with the calling terminal.
进一步地,本发明实施例还分别提供了 WebRTC服务器、 电信网关和终端的 硬件构成。 可包括至少一个处理器(例如 CPU ) ,至少一个网络接口或者其他通 信接口 ,存储器,和至少一个通信总线,用于实现这些装置之间的连接通信。 处 理器用于执行存储器中存储的可执行模块,例如计算机程序。 存储器可能包含高 速随机存取存储器(RAM: Random Access Memory ) ,也可能还包括非不稳定的 存储器( non-volatile memory ) ,例如至少一个磁盘存储器。 通过至少一个网络接 口 (可以是有线或者无线)实现该系统网关与至少一个其他网元之间的通信连接, 可以使用互联网 ,广域网 ,本地网 ,城域网等。 Further, the embodiments of the present invention further provide hardware configurations of the WebRTC server, the telecommunication gateway, and the terminal, respectively. At least one processor (e.g., a CPU), at least one network interface or other communication interface, memory, and at least one communication bus may be included for enabling connection communication between the devices. The processor is for executing an executable module, such as a computer program, stored in the memory. Memory may contain high A random access memory (RAM) may also include a non-volatile memory such as at least one disk storage. The communication connection between the system gateway and at least one other network element may be implemented through at least one network interface (which may be wired or wireless), and may use an Internet, a wide area network, a local network, a metropolitan area network, or the like.
对于 WebRTC服务器来说,参见图 12所示,在一些实施方式中 ,存储器中存 储了程序指令,程序指令可以被处理器执行,其中 ,程序指令可包括接收单元 901、 建立单元 902、 生成单元 903、 发送单元 904、 连接单元 905 ,或者程序指令还可 以包括初始化单元。 各单元的具体实现可参见图 9所揭示的相应单元,这里不再 赘述。 For the WebRTC server, as shown in FIG. 12, in some embodiments, program instructions are stored in the memory, and the program instructions may be executed by the processor, wherein the program instructions may include a receiving unit 901, an establishing unit 902, and a generating unit 903. The sending unit 904, the connecting unit 905, or the program instructions may further include an initializing unit. For the specific implementation of each unit, refer to the corresponding unit disclosed in FIG. 9, and details are not described herein again.
对于电信网关来说,参见图 13所示,在一些实施方式中 ,存储器中存储了程 序指令,程序指令可以被处理器执行,其中 ,程序指令可包括接收单元 1001、 发 送单元 1002 ,或者程序指令还可以包括编码单元、 代理单元、 初始化单元。 各单 元的具体实现可参见图 10所揭示的相应单元,这里不再赘述。 For the telecommunications gateway, as shown in FIG. 13, in some embodiments, program instructions are stored in the memory, and the program instructions may be executed by the processor, wherein the program instructions may include the receiving unit 1001, the sending unit 1002, or the program instructions. It may also include a coding unit, a proxy unit, and an initialization unit. For the specific implementation of each unit, refer to the corresponding unit disclosed in FIG. 10, and details are not described herein again.
对于终端来说,参见图 14所示,在一些实施方式中 ,存储器中存储了程序指 令,程序指令可以被处理器执行,其中 ,程序指令可包括接收单元 1 101、 连接单 元 1 102 ,或者程序指令还可以包括解码单元、 初始化单元。 各单元的具体实现可 参见图 11所揭示的相应单元,这里不再赘述。 For the terminal, as shown in FIG. 14, in some embodiments, program instructions are stored in the memory, and the program instructions may be executed by the processor, wherein the program instructions may include the receiving unit 1 101, the connecting unit 1 102, or the program The instructions may also include a decoding unit, an initialization unit. For the specific implementation of each unit, reference may be made to the corresponding unit disclosed in FIG. 11 , and details are not described herein again.
需要说明的是,本说明书中各个实施例采用递进的方式描述,每个实施例重 点说明的都是与其他实施例的不同之处,各个实施例之间相同相似部分互相参见 即可。 对于实施例公开的系统或装置而言, 由于其与实施例公开的方法相对应, 所以描述的比较简单,相关之处参见方法部分说明即可。 It should be noted that the various embodiments in the present specification are described in a progressive manner, and each embodiment focuses on differences from other embodiments, and the same similar parts between the embodiments can be referred to each other. For the system or device disclosed in the embodiment, since it corresponds to the method disclosed in the embodiment, the description is relatively simple, and the relevant parts can be referred to the method part for description.
还需要说明的是,在本文中 ,诸如第一和第二等之类的关系术语仅仅用来将 一个实体或者操作与另一个实体或操作区分开来,而不一定要求或者暗示这些实 体或操作之间存在任何这种实际的关系或者顺序。 而且,术语"包括"、 "包含 "或者 其任何其他变体意在涵盖非排他性的包含,从而使得包括一系列要素的过程、 方 法、 物品或者设备不仅包括那些要素,而且还包括没有明确列出的其他要素,或 者是还包括为这种过程、 方法、 物品或者设备所固有的要素。 在没有更多限制的 情况下,由语句 "包括一个 ...... "限定的要素,并不排除在包括所述要素的过程、 方 法、 物品或者设备中还存在另外的相同要素。 It should also be noted that, in this context, relational terms such as first and second, etc. are used merely to distinguish one entity or operation from another entity or operation, without necessarily requiring or implying such entities or operations. There is any such actual relationship or order between them. Furthermore, the terms "including", "comprising" or "comprising" or "comprising" are intended to encompass a non-exclusive inclusion, such that a process, method, article, or device that includes a plurality of elements includes not only those elements but also Other elements, or elements that are inherent to such a process, method, item, or device. In the absence of further restrictions, the elements defined by the phrase "comprising a ..." do not exclude the presence of additional elements in the process, method, article, or device that comprises the element.
结合本文中所公开的实施例描述的方法或算法的步骤可以直接用硬件、 处理 器执行的软件模块,或者二者的结合来实施。 软件模块可以置于随机存储器 ( RAM )、 内存、 只读存储器(ROM )、 电可编程 ROM、 电可擦除可编程 ROM、 寄存器、 硬盘、 可移动磁盘、 CD-ROM、 或技术领域内所公知的任意其它形式的 存储介质中。 The steps of a method or algorithm described in connection with the embodiments disclosed herein can be implemented directly in hardware, a software module executed by a processor, or a combination of both. Software modules can be placed in random access memory (RAM), memory, read only memory (ROM), electrically programmable ROM, electrically erasable programmable ROM, registers, hard disk, removable disk, CD-ROM, or any other form of storage known in the art. In the medium.
对所公开的实施例的上述说明,使本领域专业技术人员能够实现或使用本发 明。 对这些实施例的多种修改对本领域的专业技术人员来说将是显而易见的,本 文中所定义的一般原理可以在不脱离本发明的精神或范围的情况下,在其它实施 例中实现。 因此,本发明将不会被限制于本文所示的这些实施例,而是要符合与 本文所公开的原理和新颖特点相一致的最宽的范围。 The above description of the disclosed embodiments enables those skilled in the art to make or use the invention. Various modifications to these embodiments are obvious to those skilled in the art, and the general principles defined herein may be implemented in other embodiments without departing from the spirit or scope of the invention. Therefore, the present invention is not intended to be limited to the embodiments shown herein, but the scope of the inventions
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| CN114285829B (en) * | 2021-12-14 | 2023-04-18 | 上海哔哩哔哩科技有限公司 | WebRTC (Web real-time communication) connection method and system |
| CN115622981B (en) * | 2022-10-21 | 2023-11-10 | 南京北路智控科技股份有限公司 | WebRTC communication method, device, equipment and storage medium |
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| US20230275778A1 (en) * | 2017-08-13 | 2023-08-31 | Carbyne Ltd. | System, method, and computer-readable medium for streaming real-time data from a user device |
| US12309207B2 (en) * | 2017-08-13 | 2025-05-20 | Carbyne Ltd | System, method, and computer-readable medium for streaming real-time data from a user device |
Also Published As
| Publication number | Publication date |
|---|---|
| CN104283760A (en) | 2015-01-14 |
| CN104283760B (en) | 2018-05-04 |
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