RE: sip / rtp to rtmp Conference bridge

( cyber translator went crazy)

Correct, SIP is basically a two way protocol as commonly used for voice
over IP and is similar to the original H323 protocols used by Cisco.=20

SIP Is the protocol used by Vonage and most other voice over IP
carriers.=20

A typical SIP address would look very similar to an e-mail address
prefixed with a SIP: i.e. SIP:sipUserName-lHDq7Nw0oblISpjhRZutS22zBviIOh9J@p
ublic.gmane.org where
the SIP username is normally the imitation of some sort of phone number
or the DID ( direct inward dial) and the SIP server and domain would be
gateway address used to negotiate the handshake.

In normal voice over IP you would create a conference bridge allowing
SIP and pstn devices to hear each other by simply muxing analog channels
together similar to 3way calling.=20

However by adding rtmp to the conference bridge somehow it would allow
both pstn and voice over IP devices to interact with and rtmp
application.

It is always been a dream of mine to be able to dial into a FMS
conference from a phone.=20

The conference server would simply have to open up the microphone device
into the rtmp application and stream its analog in and out of FMS
applications.  Anyone dialing into the conference bridge would
immediately be part of the art RTMP application.

Now that I've explained it all we need is someone to do it.

Scott

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