[Freeswitch-users] Trickle ICE with SIP

[Michael Jerris]

I had not thought of tcp.  

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