On May 8, 2015, at 7:26 PM, Adam Ben-Ayoun <adam.ben.ayoun1@gmail.com> wrote:

Got it. Any way to do trickle ice with Freeswitch?

On 8 May 2015 at 20:51, Michael Jerris <mike@jerris.com> wrote:

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.

On May 8, 2015, at 1:19 PM, Adam Ben-Ayoun <adam.ben.ayoun1@gmail.com> wrote:

Can you clarify? Do you think there is no reason to use TURN in this case?

On 8 May 2015 at 20:06, Michael Jerris <mike@jerris.com> wrote:

I had not thought of tcp.  

On May 8, 2015, at 1:04 PM, Adam Ben-Ayoun <adam.ben.ayoun1@gmail.com> wrote:

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..

On 8 May 2015 at 20:00, Michael Jerris <mike@jerris.com> wrote:

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.

On May 8, 2015, at 11:43 AM, Adam Ben-Ayoun <adam.ben.ayoun1@gmail.com> wrote:

Well, it can get up to 30 seconds when using TURN, in other cases it's usually 2-4 seconds.

On 8 May 2015 at 18:35, Michael Jerris <mike@jerris.com> wrote:

>From your other post you said thats only when using turn?

> On May 8, 2015, at 7:36 AM, Adam Ben-Ayoun <adam.ben.ayoun1@gmail.com> wrote:
>
> Hi,
>
> 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?
>
> Thanks,
> Adam