[asterisk-dev] Asterisk 13: Media improvements update
Olle E. Johansson
oej at edvina.net
Mon Feb 24 11:21:54 CST 2014
On 24 Feb 2014, at 17:58, Saúl Ibarra Corretgé <saghul at gmail.com> wrote:
> 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...
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!
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