Probably if it actually sent a key frame when onr was desired.<div><br></div><div>We could pribably make a param to disable it. Or disable it on sip+webrtc though I still don't get why that is even a thing. Webrtc hates sip, their very core goal is to eliminaye it.<span></span></div><div><br></div><div>Yes we support the rtcp way.</div><div><br></div><div><br></div><div><br>On Friday, March 18, 2016, Fabio Margarido <<a href="mailto:fabiomargarido@gmail.com">fabiomargarido@gmail.com</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">I've now managed to get both sides to send a 200 OK response to the INFO request, but FreeSWITCH keeps sending them at a rate of more than one per second.<div><br></div><div>Is this expected behavior?</div></div><br><div class="gmail_quote"><div dir="ltr">On Wed, Mar 16, 2016 at 3:41 PM Fabio Margarido <<a href="javascript:_e(%7B%7D,'cvml','fabiomargarido@gmail.com');" target="_blank">fabiomargarido@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Anyone?<div><br></div><div>From what I understood reading other discussions regarding this request on the list, this is an obsolete way of requesting a picture refresh and that the ideal way would be via RTCP. Does FreeSWITCH support both? If that's the case, shouldn't there be a flag which allows to choose which method to use?</div><div><br></div><div>Thanks.</div></div><br><div class="gmail_quote"><div dir="ltr">On Wed, Mar 9, 2016 at 3:01 PM Fabio Margarido <<a href="javascript:_e(%7B%7D,'cvml','fabiomargarido@gmail.com');" target="_blank">fabiomargarido@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi there.<div><br></div><div>I'm trying a scenario in which FreeSWITCH sits between a WebRTC endpoint and an IMS network, performing media conversion.</div><div><br></div><div>When a call is established, FS starts to send INFO requests with an application/media_control+xml body related to RFC 5168. The WebRTC endpoint always responds with 415 Unsupported Media Type (the web application uses sip.js) and the IMS network sometimes responds with 500 Server Internal Error. The problem I have is that FS sends that request every second to both sides and keeps doing so for as long as the call is answered.</div><div><br></div><div>My question is whether it would stop or at least scale down the frequency if both sides replied with 200 OK, or alternatively if there is a way to disable the sending of these requests. I've looked at the documentation and the mailing list history but couldn't find anything conclusive.</div><div><br></div><div>Thanks in advance.</div></div></blockquote></div></blockquote></div>
</blockquote></div><br><br>-- <br><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr">Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬<div><br><div>☞ <a href="http://freeswitch.org/" target="_blank">http://freeswitch.org/</a> ☞ <a href="http://cluecon.com/" target="_blank">http://cluecon.com/</a> ☞ <a href="http://twitter.com/FreeSWITCH" target="_blank">http://twitter.com/FreeSWITCH</a></div><div><div>☞ <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch ☞ <u><a href="http://freeswitch.org/g+" target="_blank">http://freeswitch.org/g+</a></u><br><br></div><div>ClueCon Weekly Development Call <br></div><div>☎ <a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a> ☎ +19193869900 </div><div><br></div></div></div><div><a href="https://www.youtube.com/watch?v=9XXgW34t40s" style="color:rgb(17,85,204);font-size:12.8000001907349px" target="_blank">https://www.youtube.com/watch?v=9XXgW34t40s</a></div><div><a href="https://www.youtube.com/watch?v=NLaDpGQuZDA" target="_blank">https://www.youtube.com/watch?v=NLaDpGQuZDA</a><br></div></div></div></div></div></div><br>