| Scott E. Richter 2006-06-13, 1:11 pm |
| Correct, said is basically a two way protocol as commonly used for voice
over IP and is similar to the original H323 protocols used by Cisco. SIP
Is the protocol used by Vonage and most other voice over IP carriers. A
typical SIP address would look very similar to an e-mail address
prefixed with a SIP: i.e. SIP:sipUserName-lHDq7Nw0oblISpjhRZutS22zBviIOh9J@public.gmane.org where
the SIP username is normally the imitation of some sort of phone number
for the DID and the SIP server would simply be the server or a way used
to negotiate the handshake.
In normal voice over IP you would create a conference bridge were sent
them pstn devices would be able to hear each other by simply log saying
the analog channels together IE 3way calling. 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 an two and SMS
conference for myself of the war might desktops and phone. The
conference server would simply have to open up the microphone device
into the rtmp application and stream its analog in and out of San
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
-----Original Message-----
From: flashcomm-bounces-1Ss2GqJETD3yZ38Mhd3e/9ZfFG6BLHNm@public.gmane.org
[mailto:flashcomm-bounces-1Ss2GqJETD3yZ38Mhd3e/9ZfFG6BLHNm@public.gmane.org] On Behalf Of Owen van
Dijk
Sent: Tuesday, June 13, 2006 11:29 AM
To: wdb-WqTeb0HIzk1M0PWSD34TJQ@public.gmane.org; FlashComm Mailing List
Subject: Re: [FlashComm] sip / rtp to rtmp Conference bridge
On 6/13/06, Weyert de Boer <wdb-WqTeb0HIzk1M0PWSD34TJQ@public.gmane.org> wrote:
> The SIP you guys are talking about isn't the same thing as SIP of 3M
and
> mainly used by libraries?
I tend to believe we're talking about
http://en.wikipedia.org/wiki/Sessio...iation_Protocol 
--=20
Owen van Dijk
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