[asterisk-dev] srtp

Kevin P. Fleming kpfleming at digium.com
Wed Feb 10 17:28:58 CST 2010


Hans Witvliet wrote:
> While on the subject of srtp...
> 
> Would it be possible  to change halfway a configuration from "normal" to
> "secure" ?
> 
> (I presume it is probably just as impossible as changing codecs during a
> call...)

Neither are impossible; SIP/SDP supports changing pretty much anything
during the lifetime of a session.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming at digium.com
Check us out at www.digium.com & www.asterisk.org



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