<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="">Not currently, but its a feature we are interested in maybe supporting when ortc comes out with revised tricke standards that can more usefully allow handoff.<div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On May 8, 2015, at 7:26 PM, Adam Ben-Ayoun <<a href="mailto:adam.ben.ayoun1@gmail.com" class="">adam.ben.ayoun1@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">Got it. Any way to do trickle ice with Freeswitch?</div><div class="gmail_extra"><br class=""><div class="gmail_quote">On 8 May 2015 at 20:51, 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"><div style="word-wrap:break-word" class="">The TCP case is the only case, and I had not thought of it. Not sure it ever really makes sense to do it if we could add TCP \support, but we don't have that at the moment.<div class=""><div class="h5"><div class=""><br class=""><div class=""><blockquote type="cite" class=""><div class="">On May 8, 2015, at 1:19 PM, Adam Ben-Ayoun <<a href="mailto:adam.ben.ayoun1@gmail.com" target="_blank" class="">adam.ben.ayoun1@gmail.com</a>> wrote:</div><br class=""><div class=""><div dir="ltr" class="">Can you clarify? Do you think there is no reason to use TURN in this case?</div><div class="gmail_extra"><br class=""><div class="gmail_quote">On 8 May 2015 at 20:06, 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"><div style="word-wrap:break-word" class="">I had not thought of tcp. <div class=""><div class=""><div class=""><br class=""><div class=""><blockquote type="cite" class=""><div class="">On May 8, 2015, at 1:04 PM, Adam Ben-Ayoun <<a href="mailto:adam.ben.ayoun1@gmail.com" target="_blank" class="">adam.ben.ayoun1@gmail.com</a>> wrote:</div><br class=""><div class=""><div dir="ltr" class="">Can you explain how can I enable clients with UDP blocked if I don't use TURN? RTP ports on Freeswitch are UDP.. I haven't tried Verto yet, actually we are developing a mobile app, in this case it's easy for us to use something like pjsip for SIP signalling..</div><div class="gmail_extra"><br class=""><div class="gmail_quote">On 8 May 2015 at 20:00, 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"><div style="word-wrap:break-word" class="">okay.. so to confirm, there is no reason to ever be using turn, so we only are dealing w/ 2-4 seconds. What results do you get using sip.js and what results do you get using verto. I know there is some overhead in the browser and there is nothing we can do to get around that issue.<span class=""><div class=""><br class=""><div class=""><blockquote type="cite" class=""><div class="">On May 8, 2015, at 11:43 AM, Adam Ben-Ayoun <<a href="mailto:adam.ben.ayoun1@gmail.com" target="_blank" class="">adam.ben.ayoun1@gmail.com</a>> wrote:</div><br class=""><div class=""><div dir="ltr" class="">Well, it can get up to 30 seconds when using TURN, in other cases it's usually 2-4 seconds.</div><div class="gmail_extra"><br class=""><div class="gmail_quote">On 8 May 2015 at 18:35, 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">>From your other post you said thats only when using turn?<br class="">
<div class=""><div class=""><br class="">
> On May 8, 2015, at 7:36 AM, Adam Ben-Ayoun <<a href="mailto:adam.ben.ayoun1@gmail.com" target="_blank" class="">adam.ben.ayoun1@gmail.com</a>> wrote:<br class="">
><br class="">
> Hi,<br class="">
><br class="">
> We are currently using Freeswitch for voice conferencing, we use SIP for signalling. Call setup times are really slow at times (3sec-30sec), can we somehow do Trickle ICE with Freeswitch?<br class="">
><br class="">
> Thanks,<br class="">
> Adam<br class="">
<br class=""></div></div></blockquote></div></div></div></blockquote></div><br class=""></div></span></div></blockquote></div></div></div></blockquote></div></div></div></div></div></blockquote></div></div></div></blockquote></div></div></div></div></div></blockquote></div></div></div></blockquote></div><br class=""></div></body></html>