[Freeswitch-dev] FreeSWITCH SIP Adaptive codec switching
steveayre at gmail.com
Mon Oct 10 18:49:51 MSD 2016
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
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
> 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
>> Khreshchatyk Street, 10B, office 8
>> Our GitHub Repository:
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> Official FreeSWITCH Sites
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> Official FreeSWITCH Sites
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
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