VoIP测试环境
我想为一个学生项目构建一个 VoIP 测试环境。 应该使用SIP。
我的第一个想法是使用 Asterisk，OpenSIPS 或 FreeSWITCH 用于服务器，SIP Communicator，Minisip 或 Linphone 作为软件电话客户端。
- 有人有建立这样的环境的经验吗？
- 我应该使用哪种组合？
- 有没有关于设置此基础设施的教程？
I want to build a VoIP test environment for a student project. It should use SIP.
My first idea is to use Asterisk, OpenSIPS or FreeSWITCH for the server and SIP Communicator, Minisip or Linphone as softphone clients.
- Does anybody have experience with setting up such an environment?
- Which combination should I use?
- Are there any tutorials for setting up this infrastructure?
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我对
Asterisk和SJPhone有一些经验。如果您的网络不需要复杂的拨号计划（如热线等），并且您的
PC可以相互访问，那么您根本不需要服务器。SJphone只需使用IP地址或主机名即可建立点对点连接。如果您的 PC 无法相互访问，那么您应该设置
Asterisk并在其上启用客户端注册。在我的
Fedora 10上，Asterisk可以开箱即用。客户端注册就像在
ICQ或MSN上在线：服务器知道您在线并保持您的TCP流处于活动状态，以防有人想要打电话给你。它还可以用来显示人员的状态：谁在线、离线、忙碌等。
其中包含一个示例
sip.conf文件，它非常简单。 对于每个客户端，您创建一个如下所示的条目：有一些插件可以将这些数据保存在数据库中，但我还没有查看它们。
I had some experience with
AsteriskandSJPhone.If your network does not need complex dialing plans (like hotlines etc), and your
PC's are accessible from each other, then you don't need a server at all.SJphonecan establish a peer-to-peer connection just by using anIPaddress or a host name.If your PC's cannot access each other, then you should setup
Asteriskand enable client registration on it.On my
Fedora 10,Asteriskworked out of box.Client registration is like being online on
ICQorMSN: the server knows you're online and keeps yourTCPstream alive in case someone wants to call you.It also can be used to show status of people: who's online, offline, busy etc.
There is a sample
sip.conffile included, it's really simple. For each client, you create an entry like this:There are plugins that can keep this data in the database, but I didn't look at them yet.
请记住，OpenSIPS 与 Asterisk 或 FreeSWITCH 的目的非常不同。
OpenSIPS 是一个 SIP 代理，而不是任何类型的用户代理。 它不能充当呼叫端点，并且专门用于呼叫路由和低级 SIP 重整，以及一些其他杂项 UAS 类型功能（即注册器）。
Asterisk 和 FreeSWITCH 同时可以充当 PBX、IVR 服务器、媒体服务器等。
因此，这实际上取决于项目的目标是什么。 如果只是从一部手机向另一部手机拨打电话，则根本不需要中间进行任何操作。 如果要修补某些高级电话应用程序，请使用 Asterisk 或 FreeSWITCH。 如果是简单的路由和信令中介（没有媒体参与），请使用 OpenSIPS 等。
Keep in mind that the purpose of OpenSIPS vs. Asterisk or FreeSWITCH is very different.
OpenSIPS is a SIP proxy, not a user agent of any kind. It cannot function as a call endpoint and it is used exclusively for call routing and low-level SIP mangling, as well as some other miscellaneous UAS-type features (i.e. registrar).
Asterisk and FreeSWITCH meanwhile can function as PBXs, IVR servers, media servers, etc.
So, it really depends on what the objective of the project is. If it is simply to make calls from one phone to another, you do not need anything in the middle at all. If it is to tinker with certain high-level telephony applications, use Asterisk or FreeSWITCH. If it's simple routing and signaling intermediation (no media involvement), use OpenSIPS and whatnot.
请查看 Mobicents。 它是完整的 Java，由 Mobicents Sip Servlet 和 JAIN SLEE 组成，用于构建强大的 SIP、VoIP 或融合应用程序以及用于所有相关媒体内容（如会议 IVR 等）的媒体服务器，...
Please check out Mobicents. It's full Java, it's comprised of Mobicents Sip Servlets and JAIN SLEE to build powerful SIP, VoIP or converged applications and Media Server for all related media stuff such as conferencing IVR and so on,...
这取决于你想测试什么。 我建议：
It depends on what you want to test. I recommend:
我用 miniSipServer 和 Xlite 构建了一个 SIP 系统。 这两个软件都是基于 Windows 的。 这是非常容易使用。
并且有一个简单的分步文档来描述如何部署 VOIP 网络，请参阅：
http://www.myvoipapp.com/docs/faq/setup_ippbx_for_small_business_step_by_step/index.html
I build a SIP system with miniSipServer and Xlite. Both these software are based on Windows. It is very easy to use.
And there is a simple step by step document to describe how to deploy a VOIP network, please refer to:
http://www.myvoipapp.com/docs/faq/setup_ippbx_for_small_business_step_by_step/index.html
我想说，要使用的项目取决于您的主要关注点。 如果它是 SIP 并且您想了解该协议，我会使用 SIP Express Router 或派生的项目之一。 如果一般使用 voip，ASterisk 是正确的选择。 Asterisk 现在是一个很好的选择，这也取决于你想要做什么...我以前有时使用 Xlite，SJPhone、Linphone 和 KPhone...
对于客户来说，
I would say that the project to use depends on what is your primary focus. If it's SIP and you want to understand the protocol, I would use SIP Express Router or one of the project derived. If it's using voip in general, ASterisk is the right choice. Asterisk now is a good alternative
For the client, It depends on what you want to do too... I used to use Xlite sometimes, SJPhone, Linphone and KPhone...
我正在为自己构建这样一个测试服务器，您可以在这里找到一些想法：http://txlab。 wordpress.com/tag/xlab1/
I am building such a test server for myself, and you can find some ideas here: http://txlab.wordpress.com/tag/xlab1/