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