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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100223/c067da2d/attachment.htm
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 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. - - Tommy -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEAREKAAYFAkuD42MACgkQ573V05EH/pbtrQCfY4ojpCKo6oTmKerJiB+s/14l qMAAn2PAmz2qCJI+W0EPqk8Khn9K1UKx =hrY5 -----END PGP SIGNATURE-----
On Tue, 2010-02-23 at 08:22 -0500, Michelle Dupuis 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 > -- > _____________________________________________________________________ > -- 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-usersIMHO, It's impossible ;) -- Best regards, Vince Mallow xmpp: wins at jabber.slan.ru web: http://gentoo-way.blogspot.com
On Tue, Feb 23, 2010 at 8:22 AM, Michelle Dupuis <support at ocg.ca> 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? > > ThanksYate 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