US 12,231,471 B2
Method for realizing video conference, and terminal and SIP gateway
Long Shu, Beijing (CN); Jingyu Zhang, Beijing (CN); and Xiaoqin Guo, Beijing (CN)
Assigned to BOE Technology Group Co., Ltd., Beijing (CN)
Appl. No. 17/778,198
Filed by BOE Technology Group Co., Ltd., Beijing (CN)
PCT Filed Jun. 17, 2021, PCT No. PCT/CN2021/100550
§ 371(c)(1), (2) Date May 19, 2022,
PCT Pub. No. WO2021/259124, PCT Pub. Date Dec. 30, 2021.
Claims priority of application No. 202010584051.9 (CN), filed on Jun. 23, 2020.
Prior Publication US 2022/0417294 A1, Dec. 29, 2022
Int. Cl. H04L 65/1069 (2022.01); H04L 65/1104 (2022.01); H04L 65/1108 (2022.01); H04L 65/403 (2022.01); H04L 65/65 (2022.01)
CPC H04L 65/1069 (2013.01) [H04L 65/1104 (2022.05); H04L 65/1108 (2022.05); H04L 65/403 (2013.01); H04L 65/65 (2022.05)] 11 Claims
OG exemplary drawing
 
1. A method of implementing a video conference, applied to a web real-time communications (WebRTC) terminal, comprising:
sending, by the WebRTC terminal to a session initialization protocol (SIP) gateway, an HTTP GET signaling carrying request for registering an SIP account;
receiving, by the WebRTC terminal from the SIP gateway, a switching protocols signaling instructing the WebRTC terminal to use a WebSocket protocol to complete the request for registering the SIP account;
receiving, by the WebRTC terminal from the SIP gateway, a register signaling sent through the WebSocket protocol, the register signaling carrying the SIP account allocated by the SIP gateway for the WebRTC terminal;
performing an interaction of SIP signalings with the SIP gateway through the SIP account, to establish a video conference connection with an SIP terminal; wherein the WebRTC terminal is capable of parsing the received SIP signalings which is transmitted through the WebSocket protocol; and
sending a video stream that is obtained locally, and/or receiving a video stream of the SIP terminal and playing the video stream of the SIP terminal with a browser;
wherein in accordance with a determination that the video conference is created by the WebRTC terminal, and the WebRTC terminal invites the SIP terminal to join the video conference, the performing an interaction of SIP signalings with the SIP gateway through the SIP account comprises:
sending, by the WebRTC terminal to the SIP gateway, a first SIP signaling of INVITE through the registered SIP account, wherein the first SIP signaling carries a first invitation inviting the SIP terminal to join the video conference;
receiving, by the WebRTC terminal from the SIP gateway, a temporary response of Trying;
receiving, by the WebRTC terminal, a second SIP signaling of 200 OK forwarded by the SIP gateway, wherein the second SIP signaling carries an acknowledgement of the first invitation by the SIP terminal; and
sending, by the WebRTC terminal to the SIP gateway, an ACK signaling for acknowledging the receipt of the second SIP signaling;
wherein each of the first SIP signaling of INVITE, the temporary response of Trying, the second SIP signaling of 200 OK and the ACK signaling is an SIP signaling transmitted through the WebSocket protocol.