[asterisk-dev] srtp

[Olle E. Johansson]

11 feb 2010 kl. 00.31 skrev 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.
> 
The question is how we should handle it by design. A reinvite to change media encapsulation would not go through the dialplan, so we have to have code that either accepts it or denies it, based on requirements on the bridge.

/O