Twilio WebRTC 与 DIY WebRTC [英] Twilio WebRTC vs DIY WebRTC
问题描述
WebRTC 是免费的，您可以自行设置整个过程，而无需花一分钱.这有点困难，减轻故障转移确实很困难，但您当然可以免费做到.
Tokbox 或 Twilio 收费是因为这些工具抽象了在 WebRTC 应用程序中设置、运行和管理故障转移的一些非常严格的复杂性.
在 TokBox 的案例中:
无需设置STUN、TURN服务器，不用担心关于与 android 或 IOS 客户端的集成，他们提供了一个插件对于 IE 也是如此，所以开箱即用，您可以获得一切，而您只需要专注于您的应用程序逻辑而不是 WebRTC 的细微差别.这是一个很大的优势.
RELAY 和 ROUTED 方案都在框中，因此您可以编写
RELAY 通信失败时的故障转移方案.虽然有一些很好的基于 JavaScript 的框架，它们以更清晰的方式做到这一点方式.- 它添加了大量其他有助于构建 android 和 IOS 的好东西客户没有任何痛苦.
STUN 或 TURN 服务器仅用于信令目的，并且此信令发生在任何实际传输之前.这个信号非常小并且携带两个浏览器(运行浏览器的机器)的IP地址.对于传输，通信是在浏览器(点对点)之间完成的，因此不涉及服务器.
<块引用>您的中继不是从中央服务器发生的，因此您没有支付传出带宽成本.
要设置 Turn Server，
使用此服务器，构建它并将其放入 Rackspace/Amazon Web服务实例，您对 TURN 感到满意服务器.就是这样，设置您的应用程序并享受 WebRTC 带来的乐趣免费.
rfc5766-turn-server
如果你想使用一些更自由的框架来让自己更轻松，请查看:EasyRTC 和 PeerJS
享受使用 WebRTC 开发....
Is WebRTC going to be free for web developers to set up video calls on web pages? why does Twilio having pricing 25c per mins for video calls, is it going to be too expensive for the small guy to mange video calls on web hosting servers?
any advice from anyone deep into WebRTC already?
Some of the comments above are not well informed.
Someone wrote, since the bandwidth needed in case of media relay is higher as well. This is not entirely true, transmission happens between Peers(Browsers), servers are used just for signalling(relaying IP addresses of connecting peers and some more info), you can ROUTE your transmission from central server(for fail overs), but can surely do without it for free.
WebRTC is Free and you can setup the whole thing on your own without having to shell out even a penny. It is a bit hard and mitigating fail-overs is really difficult, but you can certainly do it for free.
Tokbox or Twilio charge money because these tools abstract some very rigid complexities of setting up, running and managing fail-overs in a WebRTC application.
In TokBox's Case:
You don't need to setup STUN, TURN servers, you don't have to worry about integration with android or IOS clients, they provide a plugin for IE too, so out of box you get everything and you just have to concentrate on your application logic rather than WebRTC nuances. This is a big plus.
Both RELAY and ROUTED schemes came in the box hence you can write
fail-over scenarios if RELAY communication fails. Although there are some good JavaScript based frameworks that do this in a much cleaner manner.- It adds slew of other goodies which help in building android and IOS clients without any pain.
STUN or TURN Servers are used only for Signalling Purpose, and this signalling happens before any actual transmission. This signal is very small and carries the IP address of both the browsers(machines running browsers). For Transmission the communication is done between Browsers(Peer to Peer) themselves, so no server is involved.
Your relay is not happening from a central server so you don't have to pay for the outgoing bandwidth cost.
To Setup Turn Server,
Use this server, build it and put it into a Rackspace/Amazon Web Services instance and you are Good with your TURN Server. That is It, setup your application and have fun with WebRTC for FREE.
rfc5766-turn-server
If you wish to Use some more free framework to ease yourself more, check out: EasyRTC and PeerJS
Enjoy Developing with WebRTC....
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