<div dir="ltr">Can you clarify? Do you think there is no reason to use TURN in this case?</div><div class="gmail_extra"><br><div class="gmail_quote">On 8 May 2015 at 20:06, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">I had not thought of tcp. <div><div class="h5"><div><br><div><blockquote type="cite"><div>On May 8, 2015, at 1:04 PM, Adam Ben-Ayoun <<a href="mailto:adam.ben.ayoun1@gmail.com" target="_blank">adam.ben.ayoun1@gmail.com</a>> wrote:</div><br><div><div dir="ltr">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><div class="gmail_quote">On 8 May 2015 at 20:00, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">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><div><br><div><blockquote type="cite"><div>On May 8, 2015, at 11:43 AM, Adam Ben-Ayoun <<a href="mailto:adam.ben.ayoun1@gmail.com" target="_blank">adam.ben.ayoun1@gmail.com</a>> wrote:</div><br><div><div dir="ltr">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><div class="gmail_quote">On 8 May 2015 at 18:35, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span> wrote:<br><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>
<div><div><br>
> On May 8, 2015, at 7:36 AM, Adam Ben-Ayoun <<a href="mailto:adam.ben.ayoun1@gmail.com" target="_blank">adam.ben.ayoun1@gmail.com</a>> wrote:<br>
><br>
> Hi,<br>
><br>
> 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>
><br>
> Thanks,<br>
> Adam<br>
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