On Feb 29, 2012, at 7:25 AM, Jonas Kellens wrote: >> The Asterisk server still stays in the SIP Signaling path of the call, just >> media does not flow through the server. You can verify this by running a SIP >> debug and looking at the media endpoints. > > What is it that I should be looking for in the SIP debug information ? Is it > in the SDP-body ?
To set up direct media, Asterisk will send a re-invite with an SDP body containing the address of the other endpoint. RTP then flows directly between endpoints. If you just want to know if Asterisk is in the media path or not, you can also use "rtp set debug ..." and Asterisk will log a line for each RTP packet it handles. -- v: 248.893.0738 | f: 248.893.0747 http://macprofessionals.com/ find us on facebookall -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
