[asterisk-users] directrtp with SIP + H.323

[Kevin P. Fleming]

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: kfleming at digium.com
Check us out at www.digium.com & www.asterisk.org