[Freeswitch-users] Media delay when calling FS from a WebRTC client

[Michael Jerris]

huh?

On Mar 13, 2014, at 12:50 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> On the same browser?
> 
> 
> On Thu, Mar 13, 2014 at 9:43 AM, Michael Jerris <mike at jerris.com> wrote:
> sipml5 seems to handle this part much much better than jssip.
> 
> On Mar 13, 2014, at 12:34 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> 
>> So, this is WebRTC browser behavior, right?  Not something specific to JsSip?
>> 
>> 
>> On Thu, Mar 13, 2014 at 9:20 AM, mbo <mbodbg at gmx.net> wrote:
>> I also have seen that it sometimes takes up to 10 seconds before the websocket message is sent, depending on the router I’m behind, but I’ve no solution for it yet.
>> 
>> 
>> Am 13.03.2014 um 16:58 schrieb Oleg Stolyar <olegstolyar at gmail.com>:
>> 
>>> Thanks guys!  Another related question.  It's not FreeSWITCH but perhaps someone here knows.
>>> 
>>> There is a delay in JsSip demo when gathering candidates.  The delay occurs after the last candidate is received and before sending the websocket message.  From looking around the web I found that it may be due to the WebRTC trying to get more candidates and only sending the message when that effort times out.
>>> 
>>> Does anyone know if there is a way around it?
>>> 
>>> 
>>> 
>>> On Thu, Mar 13, 2014 at 6:27 AM, Brian West <brian at freeswitch.org> wrote:
>>> I don’t think you can speed this up, if you try DTLS from FS to FS its instant, no delays but no matter what browser you selected there is some delay.
>>> --
>>> Brian West
>>> brian at freeswitch.org
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>>> 
>>> On Mar 13, 2014, at 4:29 AM, mbo <mbodbg at gmx.net> wrote:
>>> 
>>> > we also face that problem. Is there any solution to speed up DTLS handshake so far?
>>> >
>>> > Thanks
>>> >
>>> > Markus
>>> >
>>> 
>>> 
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>> 
>> 
>> _________________________________________________________________________
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> 
> 
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> 
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