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List: freeswitch-dev
Subject: Re: [Freeswitch-dev] FreeSWITCH SIP Adaptive codec switching
From: Michael Jerris <mike () jerris ! com>
Date: 2016-10-10 15:16:51
Message-ID: 984B99B9-9BBA-4C4D-BB14-AFBFAD75A516 () jerris ! com
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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.
>
[Attachment #5 (unknown)]
<html><head><meta http-equiv="Content-Type" content="text/html \
charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: \
space; -webkit-line-break: after-white-space;" class="">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.<div \
class=""><br class=""><div><blockquote type="cite" class=""><div class="">On Oct 10, \
2016, at 10:49 AM, Steven Ayre <<a href="mailto:steveayre@gmail.com" \
class="">steveayre@gmail.com</a>> wrote:</div><br \
class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">You might be \
able to script it now.<div class=""><br class=""></div><div class="">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.</div><div class=""><br class=""></div><div \
class="">Wouldn't be event based or automatic, you'd need a script that periodically \
polls the current quality of the call.</div><div class="gmail_extra"><br \
class=""><div class="gmail_quote">On 10 October 2016 at 00:30, Michael Jerris <span \
dir="ltr" class=""><<a href="mailto:mike@jerris.com" target="_blank" \
class="">mike@jerris.com</a>></span> wrote:<br class=""><blockquote \
class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc \
solid;padding-left:1ex">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.<div class="m_8853309510867252752HOEnZb"><div \
class="m_8853309510867252752h5"><span class=""></span><br class=""><br class="">On \
Sunday, October 9, 2016, Philippe Duke <<a href="mailto:philippe46@netassist.ua" \
target="_blank" class="">philippe46@netassist.ua</a>> wrote:<br \
class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px \
#ccc solid;padding-left:1ex">Hello, dear FreeSWITCH developers.<br class=""> <br \
class=""> Would like to ask you what we need to implement adaptive codec switching<br \
class=""> using SIP re-invites in the application.<br class="">
<br class="">
Adaptive codec switching is the codec parameters renegotiation (SIP<br class="">
re-invite) using network measurements of jitter and bandwidth between<br class="">
two legs. Would like to ask if we have a some docu defines this method.<br class="">
<br class="">
We have a lack of clients support it, but we may rely on open project to<br class="">
make it work. I suggest to use some kind of protocol extension to achive<br class="">
such behavior.<br class="">
<br class=""></blockquote></div></div></blockquote></div></div></div></div></blockquote></div><br \
class=""></div></body></html>
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