On Sun, Nov 5, 2017, at 07:16 AM, Saint Michael wrote: > Please correct me if I am wrong. With PJSIP there is no way for Asterisk > to > stay a OUT of the media path, while with the old SIP channel, using > directrtpsetup and directmedia, it just works. The issue I think is that > other servers do not accept reinvites or updates to redirect media, so > PJSIP will not be able to step out ever. Using the old sip channel, the > 200 > OK with SDP tells the calling side to talk direcly to the other side. > Is there a way to do this with PJSIP?
There is no "directrtpsetup" equivalent in PJSIP. Even in chan_sip it was experimental and could break things depending on the codec payloads in use. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
