[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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