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Editorial Team
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Asked: 2026-05-23T18:17:43+00:00

I’m developing a flash-sip bridge application that connects the two. I have my own

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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:

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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  1. Editorial Team

    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.)

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