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

[Igor Olhovskiy]

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.

пятница, 10 июля 2015 г. пользователь Denis Jakovlev написал:

> 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 <http://mob.tel>. 775-415-382*
>
>
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-- 
Best regards,
Igor
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