red5phone 有什么用？为什么需要星号？我该如何测试它？
我正在开发一个连接两者的 flash-sip 桥接应用程序。我有自己的服务器端 RTMP 实现，因此我可以对流数据做任何我想做的事情。我还有一个电话会议服务提供商，为了使用他们的服务，我调用他们的 Web API 创建会议室，然后我对他们的 IP 地址进行 SIP 呼叫以接收从电话与会者收到的音频，并将 PC 与会者的音频发送回他们。
这就是我所需要的。我在 SIP/Voip 领域没有太多经验，所以我搜索了具有类似功能的开源项目，我发现 peers，我用它成功呼叫了一些SIP地址。我认为它应该是解决方案的一部分，因为有了它我可以调用我的服务提供商地址来交换音频流。然后是编解码器问题。从 SIP 连接接收的音频采用 G711 编码，但 Flash 音频通常采用 Nellymouse/AAC 编码。因此，只有 peers 我无法做我需要的事情。
然后我尝试了 red5phone，正如其名称所示，它是一个在 Flash 音频和SIP 音频。所以它应该完全符合我的需求。我尝试浏览演示项目，发现有一些我的 SIP 帐户提供商没有提供的信息。
我有一个来自 Sip2sip.info 的免费 SIP 帐户，详细信息如下：
- SIP 地址：[电子邮件受保护]
- 密码：password
- 用户名：password
- 域/领域：sip2sip.info
- 出站代理：proxy.sipthor.net
- XCAP 根：https://xcap.sipthor.net/xcap-root
red5phone在登录界面要求的信息：
- 电话# ____
- 用户名 ____ >
- 密码____
- 会议____
- SIP领域___
- _ SIP 服务器____ >
- OB 代理____
- Red5 URL ____
如您所见，我的 SIP 帐户提供商没有给我电话#、会议和SIP服务器。所以我的问题是，如何使用我的SIP帐户来使用red5phone？或者我是否需要设置另一个服务（本地或来自其他服务提供商）才能使用它？
I'm developing a flash-sip bridge application that connects the two. I have my own server side RTMP implementation, so I can do whatever I want to with the streaming data. I also have a phone conference service provider, to use their service I call their web API to create a conference room, then I make a SIP call to their IP address to receive audio received from phone attendees, and send PC attendees' audio back to them.
So that's what I need. I don't have much experience in the world of SIP/Voip, so I searched for open source project that does the similar, and I found peers, with which I successfully called some SIP addresses. I think it should be part of the solution because with it I can call my service providers address to exchange audio stream. And then it came the codec problem. Audio received from SIP connection are encoded in G711, but flash audio is usually in Nellymouse/AAC. So with only peers I can't do what I need.
Then I tried red5phone, as its name states it's a project that does the audio bridging between flash audio and SIP audio. So it should fit my needs, completely. I tried to go through the demo project and find there are some information my SIP account provider didn't give.
I have a free SIP account from Sip2sip.info, and here's the details:
- SIP address: [email protected]
- Password: password
- Username: password
- Domain/Realm: sip2sip.info
- Outbound proxy: proxy.sipthor.net
- XCAP root: https://xcap.sipthor.net/xcap-root
Information asked for by red5phone in the login interface:
- Phone# ____
- Username ____
- Password ____
- Conference ____
- SIP Realm ____
- SIP Server ____
- OB Proxy ____
- Red5 URL ____
As you can see my SIP account provider didn't give me a phone#, conference and SIP Server.So my question is, how do I use my SIP account to use red5phone? Or do I need to setup another service(either locally or from other service providers) to use it?
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几年前我玩过 red5phone：它很棒，并且完全满足您的需要。
它是在服务器上运行的 SIP 用户代理，使用 RTMP 将语音传输到客户端的麦克风和扬声器。
这是理想的配置，因为它使专有协议尽可能靠近客户端，并从此使用开放的标准 SIP。
对于涉及资金的实际部署，您肯定希望通过出站代理（例如 Asterisk）传递 SIP 流量以进行记帐和授权，可能还包括媒体转码。但这部分是干净的、标准的 SIP。为您执行此操作的人员根本不需要 flash 或 red5 经验，只需要 SIP。
服务器名称应为 sip2sip.info，再次尝试输入电话号码# 的用户名。
如果它不起作用，请在每个字段中放入不同的字符串，在服务器上启动wireshark或tcpdump捕获（服务器和客户端之间的flash通信对我们来说根本没有信息），然后再试一次。 SIP 是一个很好的纯文本协议，您肯定能够理解它。 （或者只是将其发布在这里，我们会提供帮助。）
I played with red5phone a few years ago: It's great, and does exactly what you need.
It's a SIP user agent running on the server, using RTMP to channel the voice to and from the microphone and speaker of the client.
This is a desirable configuration, because it keeps the proprietary protocols as close to the client as possible, and uses open, standard SIP from there on.
For a real deployment with money involved, you will definitely want to pass the SIP traffic through an outbound proxy (Asterisk for example) for accounting and authorization, possibly media transcoding. But this part is clean, standard SIP. The person doing this for you will need no flash or red5 experience at all, only SIP.
The server name should be sip2sip.info, try the username again for phone#.
If it's not working, put distinctive strings in each field, start a wireshark or tcpdump capture ON THE SERVER (the flash communication between the server and the client is not informative to us at all), and give it another go. SIP is a nice plain text protocol, you'll be able to figure it out for sure. (or just post it here and we'll help.)