yes you have to start sending the rtp to FS when you are given the SDP FS can only detect the error and correct it if it gets several RTP packets. the other option is to use stun on the client so it can put the right ip in the SDP to begin with. On Tue, Mar 23, 2010 at 2:10 PM, Fabio Pietrosanti (naif) < lists@infosecurity.ch> wrote:
So which is the right way to "stimulate" FS to detect that condition quickly?
To send out RTP packets as early as possible from the SIP UA?
What i am still missing is if FS will always rewrite the SDP header to act as a media relay between the caller and the called or not, so forcing the clients to only send packets to FS.
Fabio
On 23/03/10 19.09, Anthony Minessale wrote:
No we don't have such a feature because the media and signaling are separated by design of SIP. It would be a horrible assumption to always think the media was coming from the same IP as the SIP messages. And even then you still need the port #
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900