Hello. 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). 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. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | new listener added to event connecting sip-0.6.2.js:2655
||| <<< since answer call : 0 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.eventemitter | emitting event userMediaRequest sip-0.6.2.js:2655 ||| <<< since answer call : 41 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event userMediaRequest sip-0.6.2.js:2655 ||| <<< since answer call : 41 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.eventemitter | emitting event userMedia sip-0.6.2.js:2655 ||| <<< since answer call : 43 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event userMedia sip-0.6.2.js:2655 ||| <<< since answer call : 43 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | acquired local media stream sip-0.6.2.js:2655 ||| <<< since answer call : 44 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | emitting event connecting sip-0.6.2.js:2655 ||| <<< since answer call : 44 ms. Ожидание соединения: 47 ms. WebRTCCallCenter.js?r=1:212 Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | ICE connection state changed to "checking" sip-0.6.2.js:2655 ||| <<< since answer call : 47 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | ICE Gathering Completed sip-0.6.2.js:2655 ||| <<< since answer call : 47 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event iceComplete sip-0.6.2.js:2655 ||| <<< since answer call : 48 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event getDescription sip-0.6.2.js:2655 ||| <<< since answer call : 48 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.transaction.ist | emitting event stateChanged sip-0.6.2.js:2655 ||| <<< since answer call : 50 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | emitting event accepted sip-0.6.2.js:2655 ||| <<< since answer call : 51 ms.
Звонок начат Operator.js?r=3:1903 Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | no ACK received, attempting to retransmit OK sip-0.6.2.js:2655 ||| <<< since answer call : 60 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | no ACK received, attempting to retransmit OK sip-0.6.2.js:2655 ||| <<< since answer call : 80 ms. Wed Sep 17 2014 16:31:22 GMT+0600 | sip.invitecontext.mediahandler | ICE connection state changed to "connected" sip-0.6.2.js:2655 ||| <<< since answer call : 1216 ms.
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. In that demo with Verto https://webrtc.freeswitch.org/verto/index.html
>>>>>>>>>>Thu Sep 25 2014 16:13:14 GMT+0600 ICE connection state changed to "checking" verto-min.js:41 ||| <<< since answer call: 476 ms. verto-min.js:42 Dialog c46b6eb8-73de-62ce-df07-fb3d002e07eb: state change from trying to active verto-min.js:206 >>>>>>>>>>Thu Sep 25 2014 16:13:15 GMT+0600 ICE connection state changed to "connected" verto-min.js:41 ||| <<< since answer call: 2014 ms. How do I configure Freeswitch and javascript app to sound appeared within 200 ms?