如何在服务器端实现VOIP服务?

How to implement a VOIP service on the server?

我知道我的问题很宽泛,但我不知道从哪里开始研究。

如何在服务器上实现 VoIP?我相当确定它不使用 http/https 协议。如果是这样,可以使用哪些 standard/famous 协议?有没有开源的?有哪些好的参考资料可以着手开展这项工作?

首先查看 SIP、RTP 和 RTCP 协议。我相信它们形成了一组极简的 VoiP 所需协议

一些相关的开源项目:

http://www.fsf.org/campaigns/priority-projects/priority-projects/highpriorityprojects#Replaceskype

"There are a number of such programs, such as Ekiga, Twinkle, Coccinella, QuteCom, and Jitsi. Unfortunately, these programs only replace some of Skype's functionality, and only in some situations. WebRTC has a mission to enable rich, high quality, Real-Time Communications (RTC) applications to be developed in the browser via simple Javascript APIs and HTML5. Developers should consider helping free software VoIP and video, chat, and multimedia communications projects."