Tommy Botten Jensen wrote: > Michelle Dupuis skrev: >> 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? > > Unfortunately, that is not possible.
As I understand it, the H.323 protocol, in most implementations, does not allow redirecting the media endpoints after the call is setup. In a pure proxy-type environment, where the media never goes through a switch at all, this would be possible, but for a B2BUA like Asterisk, it's not likely to be possible. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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
