On Tue, Feb 23, 2010 at 8:22 AM, Michelle Dupuis <[email protected]> wrote: > We're creating a SIP gateway for a client that will take one leg of a call > in via SIP, and out the other side via H.323. To minimize load on the > gateway, we would like to have the RTP stream bypass the gatewayy altogether > (directrtp/reinvite). Is this possible with these to protocols? > > Thanks
Yate claims it can do this: http://yate.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
