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用于Web会议和文件共享的SIP / XMPP有什么区别？

[英]What's the difference between SIP/XMPP for web conferencing and file-sharing?

原文 2014-05-07 13:51:06 3 1 xmpp/ sip

I want to setup a personal videoconferencing service for my family, friends and myself. 我想为我的家人，朋友和我自己设置个人视频会议服务。 The main problem I have with current options is that they are either closed-source and centralized (GG hangouts, skype) or open-source but not working in corporate environment or in hotels (due to strict firewalling rules and the "Skype is going through, if you want VOIP use that" kind of netadmin reaction). 我目前使用的选项的主要问题是它们是封闭源代码和集中式（GG环聊，skype）或开放源代码，但不能在公司环境或酒店中工作（由于严格的防火墙规则，“ Skype正在通过，如果您想VOIP使用这种“ netadmin反应”。

I have two solutions then. 那我有两个解决方案。 Either setup a STUN/TURN relay server and use XMPP and SIP as I used to, but that would require my friends to setup that too. 要么设置一个STUN / TURN中继服务器，然后像以前一样使用XMPP和SIP，但是那也需要我的朋友进行设置。 Or setup a whole VOIP server. 或设置整个VOIP服务器。 2 solutions come to mind: SIP and XMPP. 我想到了两个解决方案：SIP和XMPP。 Though to my knowledge, each of them ultimately uses the (S)RTP/RTCP protocol. 据我所知，它们最终都使用（S）RTP / RTCP协议。

And that's the problem. 这就是问题所在。 Out of the specific signaling part used by the two of them, I really can't figure out the difference between them, their typical use case. 在他们两个使用的特定信号部分之外，我真的无法弄清它们之间的区别，即它们的典型用例。

1 个解决方案

I think you're right in that as far as setting up a video conferencing system XMPP and SIP are equivalent. 我认为就视频会议系统XMPP和SIP的等效而言，您是正确的。 They both are signalling only protocols and the media sessions they set up typically use RTP (although they can both be used to set up any kind of session you want but RTP is the norm). 它们都只是信令协议，它们建立的媒体会话通常使用RTP（尽管它们都可以用来建立您想要的任何类型的会话，但RTP是正常的）。

The biggest problem is also going to be the one you mention about getting video streams out of a corporate firewall. 您提到的最大问题也将是您提到的将视频流从公司防火墙中删除的问题。 Skype overcomes this obstacle by sending it's media over an SSL connection and is thus able to get through firewalls. Skype通过通过SSL连接发送媒体来克服了这一障碍，因此能够通过防火墙。 Theoretically you could do the same with RTP and in the past I once used openvpn connections with a SIP client to test some audio calls. 从理论上讲，您可以使用RTP进行相同的操作，过去我曾经与SIP客户端一起使用openvpn连接来测试一些音频呼叫。 My experience wasn't great as the audio was very choppy, assumedly as a result of all the extra packaging that is required to get the high volume of small audio packets from one end to the other. 我的经验不是很好，因为音频非常不稳定，这可能是由于要从一端到另一端获取大量小音频数据包所需的所有额外包装的结果。 That was nearly a decade ago though so perhaps with the better CPU and bandwidth resources available now it would work better. 那是将近10年前，因此，也许现在有了更好的CPU和带宽资源，它会更好地工作。

Personally I think I'd stick with Skype as it's going to be a big hassle to set up your own system. 我个人认为我会坚持使用Skype，因为建立您自己的系统会很麻烦。 If you were to go ahead with your own the first option I would try would be Asterisk combined with openvpn so that if the clients were behind a firewall or had NAT issues they could connect over it. 如果您要自己动手，我会尝试将Asterisk与openvpn结合使用，这样，如果客户端位于防火墙后面或出现NAT问题，他们可以通过它进行连接。