On 24 Feb 2014, at 17:58, Saúl Ibarra Corretgé <[email protected]> wrote:
> On 02/24/2014 02:08 PM, Olle E. Johansson wrote: >> >> On 24 Feb 2014, at 13:59, Matthew Jordan <[email protected]> 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... > That is an implementation choice. Sometimes there are limitations in how many active codecs a device can have, due to licenses or hardware activation or something else. The fun part is that Asterisk actually can switch codecs mid-call without a re-invite today. The next SIPit will be in Europe in October. Plan for it! /O -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
