My strongest suggestion - switch to mod_verto. All sip implementation for JavaScript is buggy in many different ways. And with every new release of the browser you will get new surprises. <span></span><br><br>пятница, 10 июля 2015 г. пользователь Denis Jakovlev написал:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">Hi All !<br><div><div><br></div><div>I have a question. I use several different libraries (jssip, sip.js) for RTC Web communications. In both variants I observed during the freezing of the video communication. Sometimes it works great and without freezing. Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've tried various settings Dialplan and internal.xml. But the result is almost the same. </div><div>Maybe dear colleagues will share tips on how to make the connection stable? </div><div><br></div><div>PS: I use FreeSwitch 1.7 on Debian 8. </div><div><br></div><div><br></div><div><br><div style="margin:0px;font-size:11px;font-family:Arial;color:rgb(192,192,192)"><i>-- </i></div><div style="margin:0px;font-size:11px;font-family:Arial;color:rgb(192,192,192)"><i>S pozdravem,</i></div><div style="margin:0px;font-size:11px;font-family:Arial;color:rgb(192,192,192)"><i>Ing.Denis Jakovlev </i></div><div style="margin:0px;font-family:Arial;color:rgb(192,192,192)"><i><a href="http://mob.tel" target="_blank">mob.tel</a>. 775-415-382</i></div></div>
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