<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Hi All !<br class=""><div apple-content-edited="true" class=""><div class=""><br class=""></div><div class="">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 class="">Maybe dear colleagues will share tips on how to make the connection stable? </div><div class=""><br class=""></div><div class="">PS: I use FreeSwitch 1.7 on Debian 8. </div><div class=""><br class=""></div><div class=""><br class=""></div><div class=""><br class=""><div style="orphans: 2; widows: 2; margin: 0px; font-size: 11px; font-family: Arial; color: rgb(192, 192, 192);" class=""><i class="">-- </i></div><div style="orphans: 2; widows: 2; margin: 0px; font-size: 11px; font-family: Arial; color: rgb(192, 192, 192);" class=""><i class="">S pozdravem,</i></div><div style="orphans: 2; widows: 2; margin: 0px; font-size: 11px; font-family: Arial; color: rgb(192, 192, 192);" class=""><i class="">Ing.Denis Jakovlev </i></div><div style="orphans: 2; widows: 2; margin: 0px; font-family: Arial; color: rgb(192, 192, 192);" class=""><i class="">mob.tel. 775-415-382</i></div></div>
</div>
<br class=""></body></html>