<div dir="ltr">
<p class=""><span class="">vars.xml</span></p><p class=""><span class=""><</span><span class="">X-PRE-PROCESS</span><span class=""> </span><span class="">cmd</span><span class="">=</span>"set"<span class=""> </span><span class="">data</span><span class="">=</span>"suppress_cng=true"<span class="">/></span></p><p class=""><span class="">If that doesn't work get better network conditions or use lower bitrate codec.</span></p></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Sep 25, 2014 at 5:51 AM, Алексей <a href="http://kob.kz">kob.kz</a> Целищев <span dir="ltr"><<a href="mailto:a-altair@mail.ru" target="_blank">a-altair@mail.ru</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div><div><div><p>Hello.<br>I work with browser with WebRTC with FreeSWITCH Version 1.5.14b+git~20140730T044231Z~b1f382c67e~64bit (git b1f382c 2014-07-30 04:42:31Z 64bit).<br>Since answer call until the sound is appeared (ICE connection state changed to "connected" in the log) takes ~ 1216ms, which is monstrous for my problems. Have any ideas on this? STUN, as you can see, is disabled.<br><em><br>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | new listener added to event connecting sip-0.6.2.js:2655</em></p><p><em>>>> ||| <<< since answer call : 0 ms. </em><br><em>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.eventemitter | emitting event userMediaRequest sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 41 ms. </em><br><em>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event userMediaRequest sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 41 ms. </em><br><em>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.eventemitter | emitting event userMedia sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 43 ms. </em><br><em>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event userMedia sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 43 ms. </em><br><em>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | acquired local media stream sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 44 ms. </em><br><em>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | emitting event connecting sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 44 ms. </em><br><em>Ожидание соединения: 47 ms. WebRTCCallCenter.js?r=1:212</em><br><em>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | ICE connection state changed to "checking" sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 47 ms. </em><br><em>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | ICE Gathering Completed sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 47 ms. </em><br><em>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event iceComplete sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 48 ms. </em><br><em>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event getDescription sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 48 ms. </em><br><em>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.transaction.ist | emitting event stateChanged sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 50 ms. </em><br><em>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | emitting event accepted sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 51 ms. </em><br><em>>>>>> Звонок начат Operator.js?r=3:1903</em><br><em>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | no ACK received, attempting to retransmit OK sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 60 ms. </em><br><em>Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | no ACK received, attempting to retransmit OK sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 80 ms. </em><br><em>Wed Sep 17 2014 16:31:22 GMT+0600 | sip.invitecontext.mediahandler | ICE connection state changed to "connected" sip-0.6.2.js:2655</em><br><em>>>> ||| <<< since answer call : 1216 ms.<br><br></em>Nuance: if the operator and the server are in the local network and connected to an internal address, then until the sound appeared goes ~ 280ms, which is a stretch of more or less acceptable. But if I connect to the browser through external network (router with port forwarding), then the delay increases by 5 times.<br><br>In that demo with Verto <a href="https://webrtc.freeswitch.org/verto/index.html" target="_blank">https://webrtc.freeswitch.org/verto/index.html</a><br><br></p><p><em>>>>>>>>>>>>>>>>>Thu Sep 25 2014 16:13:14 GMT+0600 ICE connection state changed to "checking" verto-min.js:41</em><br><em>>>> ||| <<< since answer call: 476 ms. verto-min.js:42</em><br><em>Dialog c46b6eb8-73de-62ce-df07-fb3d002e07eb: state change from trying to active verto-min.js:206</em><br><em>>>>>>>>>>>>>>>>>Thu Sep 25 2014 16:13:15 GMT+0600 ICE connection state changed to "connected" verto-min.js:41</em><br><em>>>> ||| <<< since answer call: 2014 ms.</em></p><p>How do I configure Freeswitch and javascript app to sound appeared within 200 ms?</p></div></div></div></div>
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<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div dir="ltr">Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬<div><br><div>☞ <a href="http://freeswitch.org/" target="_blank">http://freeswitch.org/</a> ☞ <a href="http://cluecon.com/" target="_blank">http://cluecon.com/</a> ☞ <a href="http://twitter.com/FreeSWITCH" target="_blank">http://twitter.com/FreeSWITCH</a></div><div><div>☞ <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch ☞ <u><a href="http://freeswitch.org/g+" target="_blank">http://freeswitch.org/g+</a></u><br><br></div><div>ClueCon Weekly Development Call <br></div><div>☎ <a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a> ☎ +19193869900 </div><div><br></div></div></div></div>
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