WebRTC 如何处理多对多连接? [英] How does WebRTC handle many-to-many connections?

问题描述

如果我和其他 7 个用户在一个房间里，我想知道 WebRTC 是否会强制每个用户与其他每个参与者建立连接.

if I am in a room with other 7 users, I am wondering if WebRTC force every user to establish a connection to each one of other participants.

显然它会消耗 7kb/s*7 的下载甚至上传，如果它们的连接已经很忙，许多连接无法处理这个.相反，使用某种媒体中继，带宽使用率仅为 7kb/s，但您将失去对等点之间的带宽自适应.

Obviously it would consume something like 7kb/s*7 download and even upload, and many connection cannot handle this if their connection is already busy. Instead with some kind of media relay the bandwidth usage would be only 7kb/s but you would lose bandwidth adaptation between peers.

您知道任何媒体转播或解决此问题的方法吗?是 TURN 服务器(例如 https://code.google.com/p/rfc5766-turn-server/a> ) 适合这种工作(包括多播)?

Do you know any media relay, or way to solve this problem? is TURN server ( like https://code.google.com/p/rfc5766-turn-server/ ) suitable for this kind of job ( multicast included )?

推荐答案

TURN 服务器用作回退中继服务器，以便在由于防火墙或其他网络问题而无法进行直接对等连接时启用连接.(更多信息此处:按 P 获取演讲者备注.)TURN 服务器不是为媒体分发而设计的.

A TURN server works as a fallback relay server in order to enable connectivity when direct peer-to-peer connectivity is impossible because of firewalls or other network issues. (More information here: press P for speaker notes.) TURN servers are not designed for media distribution.

多点控制单元<​​/a>可以解决您提到的问题:此此处.如该幻灯片的注释所述:

A Multipoint Control Unit could solve the problem you refer to: there's an example topology for this here. As stated in the notes for that slide:

这是一个专门用于分发媒体的服务器，并且可以处理大量参与者；它也可以做聪明诸如选择性流转发、音频或视频混合之类的事情，或录音.

This is a server that's made specifically to do distribution of media, and can handle large numbers of participants; it can also do smart things like selective stream forwarding, mixing of the audio or video, or recording.

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