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?