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Explanation on webrtc's stun/turn service in tsingsee green rhino video

2022-06-23 08:11:00 Tsingsee green rhino video

EasyGBS、EasyCVR、EasyNVR、EasyDSS And other video platforms currently provide WebRTC function . In the process of use, an additional stun/turn service . We have also received many inquiries from users recently , About what the service does ?

In this article, we will share with you the description of this service .

stay WebRTC In communication , Generally, two users are in two LANs respectively , Not able to communicate directly . Both can be connected to the public network , But there is no way to communicate directly , Because I don't know the other party's Internet address . therefore , That's what happened stun service .

Before two users communicate , First, we will report to the public network stun The service sends the request , Get your own public address , Then the server forwards their public network addresses to each other , In this way, both parties will know each other's public network address . According to this public network address, you can communicate directly .

Google Provides a test stun/turn Website of the service , You can test the corresponding stun service .

https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/

In the red box above is stun The public network address returned by the service 36.5.146.87, This can prove that the network is unobstructed .

stun Available in most cases , But in symmetry NAT In mode, you cannot connect , Therefore, the network communication must be carried out by means of forwarding and relaying , You need to use turn service . In our video platform , We also built a turn service .

Because the forwarding mode is adopted , So the final return is turn The address and port of the service . As can be seen from the above figure , The dynamically allocated public network address is 212.64.34.165, Port is 5122.

Return at the end of the page 701 error , This is a Google Page error for , You can ignore .

WebRTC Technology provides high concurrency for the industry 、 Low latency 、 HD fluency 、 Safe and reliable audio and video services , For online education 、 Cloud Conference 、 Social entertainment and other industries and scenes . Our video platforms currently support WebRTC Play , If you are interested , You can go to the official website to experience and test .

meanwhile , We will also update the platform function development and Optimization in the article from time to time 、 Configuration tutorial 、 Problem solving 、 Industry solutions, etc , Welcome to our update , Or leave a message to interact with us .

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