Follow-up in case someone else runs into this problem.I tried my test on another machine and there was no delay before establishing the connection with JsSip (this is the first delay I reported that had to do with gathering ICE candidates). It turns out that I had VMWare Network adapters on my computer. Once I disabled them, the delay disappeared.Of course the second (post-dial) delay that Anthony explained is still there but it's short enough to not be a show-stopper for me.On Thu, Mar 13, 2014 at 9:29 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:It just sleeps whenever the channel answers so not much different.
On Mar 13, 2014 7:07 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:Thanks Anthony!!!I am using the "sleep" command in my dialplan after "answer". Is "answer_delay" preferrable?On Thu, Mar 13, 2014 at 4:54 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Its a combo of how many nic you have that are up with IP on them and the dtls handshake time which is 500ms to 1500 ms depending on the speed of the exchange which cannot be shortened. You cannot send media at all until you have dtls up, this is a known limitation of dtls and they chose to do it that way.WebRTC is not spending a lot of time pondering legacy telephony and post-dial-delay etc. Its centric on a peer-2-peer model and not even giving proper attention to use cases like FS where many clients are talking to one meta server with many WebRTC connections. Maybe when the drunken-buzz of trying to change the paradigm wears off, they will have some bloody marys and realize you can still change it while keeping the things that matter in tact that the entire human race has grown to expect (like communication channels establishing in a quick timeframe).Until then you can set up a catch-all extension at the top of your dialplan with:<action application="set" data="answer_delay=2500"/>or add it as a global to vars.xmlThis adds a 2500 ms sleep after answer to consume the time of the dtls negotiation.--On Thu, Mar 13, 2014 at 6:29 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:Thanks Markus, I might do that but first I want to get both JsSip and smpl5 working in my environment and confirm that simpl5 is faster for me.On Thu, Mar 13, 2014 at 1:17 PM, mbo <mbodbg@gmx.net> wrote:I’ve posted an issue regarding that to the jsip user group some time ago, but did not follow up yet.Maybe it makes sense to continue the discussion here as this problem is not related to freeswitch, maybe some Jssip developers can help if you post also details about your issue there.ThanksMarkusAm 13.03.2014 um 19:31 schrieb Oleg Stolyar <olegstolyar@gmail.com>:Thanks Michael!_________________________________________________________________________On Thu, Mar 13, 2014 at 11:19 AM, Michael Jerris <mike@jerris.com> wrote:
jssip didnt, at least at the time we made that site, support firefox, not sure if that has changed. sipml5 gets the candidates sent out much quicker than jssip does, but jssip is much simpler and nicer to use in other ways.On Mar 13, 2014, at 1:56 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:https://webrtc.freeswitch.org on chrome redirects to a JsJip demo.Same URL on Firefox redirects to smpl5 demo,so I was wondering if there is a reason for that and if the comparisons between JsSip and smpl5 were done on the same browser._________________________________________________________________________On Thu, Mar 13, 2014 at 10:45 AM, Michael Jerris <mike@jerris.com> wrote:
huh?On Mar 13, 2014, at 12:50 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:On the same browser?_________________________________________________________________________On Thu, Mar 13, 2014 at 9:43 AM, Michael Jerris <mike@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@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@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@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@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@freeswitch.org
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On Mar 13, 2014, at 4:29 AM, mbo <mbodbg@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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