This was pretty much my point of my reply. Polling on those is not a good architecture, and will consume ram every time you call it. To do this would require a little mod that could realtime hook those values to be acted on. Its really just the last layer of glue.
On Oct 10, 2016, at 10:49 AM, Steven Ayre <steveayre@gmail.com> wrote:
You might be able to script it now.
There are a number of rtp_audio/video_in/out_ variables. Normally these are only in the CDR at the end of the call but you can set them at any time using uuid_set_media_stats. Once they're set you can read them out (uuid_getvar or uuid_dump) and if you need to change codec call uuid_media_reneg.
Wouldn't be event based or automatic, you'd need a script that periodically polls the current quality of the call.
On 10 October 2016 at 00:30, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote: i think FreeSWITCH already has all the bits in place to force a reinvite based renegotiation. we may not have all the bits to allow for real time monitoring of quality but the major pieces are in place already to track those.
On Sunday, October 9, 2016, Philippe Duke <philippe46@netassist.ua <mailto:philippe46@netassist.ua>> wrote: Hello, dear FreeSWITCH developers.
Would like to ask you what we need to implement adaptive codec switching using SIP re-invites in the application.
Adaptive codec switching is the codec parameters renegotiation (SIP re-invite) using network measurements of jitter and bandwidth between two legs. Would like to ask if we have a some docu defines this method.
We have a lack of clients support it, but we may rely on open project to make it work. I suggest to use some kind of protocol extension to achive such behavior.