Hello, http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly
The link above indicates that it is possible to setup RTP streams to directly flow between endpoints and completely bypass Asterisk. I would like to know if this configuration would work when, a) both endpoints are behind NAT, and/or b) both endpoints don't support same codecs with media flowing through a SIP+rtpproxy server that can do transcoding ? Thanks and Regards, Vikram. -- _____________________________________________________________________ -- 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
