[Freeswitch-users] Trickle ICE with SIP

[Adam Ben-Ayoun]

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 at jerris.com> wrote:

> 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> 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>
>> 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> 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>
>>> 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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>
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>
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>
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