[asterisk-dev] srtp

[Russell Bryant]

On 02/10/2010 05:28 PM, Kevin P. Fleming wrote:
> 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.
>

How easy is it to accomplish in our current architecture is another 
question.  It's probably easier to accomplish in Asterisk than the codec 
example, but still not easy.

-- 
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org