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.<span></span><br><br>On Sunday, October 9, 2016, Philippe Duke <<a href="mailto:philippe46@netassist.ua">philippe46@netassist.ua</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello, dear FreeSWITCH developers.<br>
<br>
Would like to ask you what we need to implement adaptive codec switching<br>
using SIP re-invites in the application.<br>
<br>
Adaptive codec switching is the codec parameters renegotiation (SIP<br>
re-invite) using network measurements of jitter and bandwidth between<br>
two legs. Would like to ask if we have a some docu defines this method.<br>
<br>
We have a lack of clients support it, but we may rely on open project to<br>
make it work. I suggest to use some kind of protocol extension to achive<br>
such behavior.<br>
<br>
<br>
--<br>
Philippe Duke<br>
Network software engineer<br>
System-level developer<br>
<br>
NetAssist LLC<br>
Ukraine<br>
Khreshchatyk Street, 10B, office 8<br>
AS29632<br>
<br>
<a href="http://netassist.ua" target="_blank">http://netassist.ua</a><br>
Our GitHub Repository:<br>
<a href="https://github.com/netassist-ua" target="_blank">https://github.com/netassist-<wbr>ua</a><br>
<br>
<br>
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</blockquote>