The KEY thing you are missing is that IAX does NOT use RTP for audio.
IAX uses IAX for audio and IAX for signaling. You CANNOT "reinvite"
between a SIP/RTP endpoint and an IAX endpoint.
Since SIP, H323, MGCP, and SCCP/Skinny all use RTP for audio, in theory
you could do RTP reinvites between these protocols. I have no idea if
Asterisk supports this or not.
On Sat, 2004-09-18 at 00:52, Carlos Arnt wrote:> Looking at this explanation :
> "When SIP initiates the call, the INVITE message contains the
information
> on where to send the media streams. Asterisk uses itself as the end-points
> of media streams when setting up the call. Once the call has been accepted,
> Asterisk sends another (re)INVITE message to the clients with the
> information necessary to have the two clients send the media streams
> directly to each other."
>
> So if i really understand this using this option i can make the RTP packets
> flow from one device to another when they connect leaving only the SIP to
> asterisk .
> So for example if then I put my Grandstream with a real ip address and use
> * with a real ip address i can make my calls from nufone flow direct to my
> grandstream leaving my * bandwidth free .
>
> Like this :
>
> Grandstream begin call ----SIP---> Asterisk |
> | ----->
> Nufone.
>
> Open RTP Channel
>
> Grandstream Real IP <------> Nufone IP
>
> Right ??
> If i'm right , i try this and with tcpdump see the even with everyone
using
> real ip's, the RTP still going over asterisk using my bandwidth .
> (Note, I force grandstream to use the same codec then Nufone, G729)
>
> Can someone give-me some light ?? ;)
> Can i make this ??? Use asterisk only to begin the call and let the RTP
> flow over the client and nufone network ??
>
> Thanks alot !
>
> Carlos.
>
>
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--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
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