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I have a POTS conference call taking place and I want to live stream the audio over the internet using Wowza. My plan is to dial into the conference call essentially bridging the audio onto the Wowza server. I've got Wowza working very well on ec2, however I need to somehow turn the audio of a SIP call into a format Wowza can accept and turn into a stream.

Does anyone know if there a SIP phone that runs on Fedora which will output an RTMP stream or if there is some other workable solution?

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This is an extremely complicated question. First off, how are you incorporating POTS into a SIP call? are you hosting the gateway? if that is the case, look here for a conversion between sip and rtmp at the gateway.

For client side translation, you could try something like Xuggle. Basically, RTP is the what carries the audio streams in SIP, so you need something to translate the RTP session.

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Thanks for your answer. The SIPRTMP library seems to act as an intermediary between a browser based Flash phone and a real SIP system. I think what I am trying to do is simpler as I just want a one-way broadcast of the audio from a telephone call into an RTMP stream. Do you think maybe the SIPRTMP that you linked may also be able to do this? – Richard May 27 '11 at 23:27
@Richard, updated answer to reflect a possible client side method. – MaQleod May 28 '11 at 3:56

Take a look RTMP SIP Gateway based on Wowza Media Server I think it is exactly what you need.

Flashphoner is Wowza Media Server plugin/module that enables to make SIP call using client-side or server-side API and share incoming (SIP to Wowza) audio stream between unlimited number of Flash clients connected to Wowza Media Server.

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