[Freeswitch-dev] FreeSWITCH SIP Adaptive codec switching
mike at jerris.com
Mon Oct 10 19:16:51 MSD 2016
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 at 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 at jerris.com <mailto:mike at 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 at netassist.ua <mailto:philippe46 at 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.
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