我正在研究用于音频/图片通信的WebRTC解决方案，并且当局域网中的两个同事进行通信时，我有点担心缺少带宽控制。
基本上我希望能够为我的交换机分配带宽优先级并预先分配WebRTC呼叫。 但是我看不到在P2P呼叫中过滤数据包的正确方法。
另外，由于此操作可能会造成延迟，因此我不想对数据包进行解码。
我希望你们能以正确的方式告诉我，或者告诉我，我的解决方案能否奏效。
我不是100％肯定我计划测试的想法，因为我不知道TURN服务器如何在内部工作。
但这是一个想法：
而我不知道的是： 是否有可能让2台服务器相互认识？ 他们会像来电者之间的2层代理一样工作吗？ 如果是的话，你能告诉我为了使它工作我需要做些什么吗？
I'm working on a WebRTC solution for audio/picture comunication and I'm a bit concerned about the lack of bandwidth control when two Peers in LAN are communicating.
Basically I want to be able to prioritize and pre-allocate bandwidth on my switch for WebRTC calls. But I couldn't see a proper way of filtering the packets when they are in a P2P call.
Also, I don't want to decode the packet to do that, because of the possible delay caused by this operation.
I hope you guys can show me a proper way or just tell me if my teorical solution could work.
I'm not 100% sure about the idea I'm planning to test, because I don't know how TURN server works internally.
But here is the idea:
And what I dont know is: Is it possible to make 2 turn servers know each other? Would they work like a 2 layer proxy between callers? If yes, could you please show me what I have to do to make it work?
最满意答案
只需将内部代理服务器安装到外部转向服务器，并在您的局域网上优先使用代理服务器。
（在意识到解决方案比我想象的容易之后回答我自己的问题）
Just install a internal proxy server to the external turn server and prioritize the proxy on your lan.
(answering my own question after realizing the solution was easier than I thought)
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