I'm developing a flash-sip bridge application that connects the two. I have my own server side RTMP implementation, which means I can do whatever I want to do with the streaming data. I also have a phone conference service provider, to integrate with their service first I call their web API to create a conference room, then make a SIP call to their IP address to receive audio received from phone attendees, and send PC attendees' audio to them.
So that's my requirement. I don't have much experience on the world of SIP/Voip, so I searched for open source projects that does the similar to learn. I found peers <http://sf.net/projects/peers>, and successfully called other 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 it's incomplete a solution. Then I tried red5phone <http://code.google.com/p/red5phone>, as its name states it's a project that does the audio bridging between flash audio and SIP audio. 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: *******@sip2sip.info* * Password: ******* - Username: ****** - 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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