[Freeswitch-dev] FreeSWITCH SIP Adaptive codec switching

[Steven Ayre]

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> 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> 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.
>>
>>
>> --
>> Philippe Duke
>> Network software engineer
>> System-level developer
>>
>> NetAssist LLC
>> Ukraine
>> Khreshchatyk Street, 10B, office 8
>> AS29632
>>
>> http://netassist.ua
>> Our GitHub Repository:
>> https://github.com/netassist-ua
>>
>>
>> _________________________________________________________________________
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>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
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>
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