Hi, everyone !
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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