[asterisk-dev] Asterisk 13: Media improvements update

Saúl Ibarra Corretgé saghul at gmail.com
Mon Feb 24 10:58:55 CST 2014

On 02/24/2014 02:08 PM, Olle E. Johansson wrote:
> On 24 Feb 2014, at 13:59, Matthew Jordan <mjordan at digium.com> wrote:
>> So, today, it is possible to write a module that listens for RTCP
>> statistics from all channels and, on the fly, initiates a
>> re-INVITE/UPDATE request to the endpoints associated with a channel if
>> it feels like it.
> The cool thing is that we don't need to re-invite/update the session in
> many cases - we've already negotiated multiple codecs and can happily
> switch between them.

Heh, I guess you saw many of those devices when SIPit was held in 
Wonderland ;-) FTR, PJSIP itself (the PJSUA API more precisely) forces a 
reINVITE or UPDATE to lock down to a single codec if the reply contains 
more than one...


Saúl Ibarra Corretgé

More information about the asterisk-dev mailing list