[Freeswitch-users] how to improve the performance of Web RTC?

Hi All ! 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. Maybe dear colleagues will share tips on how to make the connection stable? PS: I use FreeSwitch 1.7 on Debian 8. -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382