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

[Denis Jakovlev]

Dobrý den,

Ok. And there is some documentation for verto.js?
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S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pátek 10. července 2015, 13:28:27, napsal jste:


You can.
But this library add a support of a really not light protocol (SIP) in your browser. And I'm not sure about quality of these libraries. With Verto you got all from "one hands", as front-part (verto.js) and back part (mod_verto)

2015-07-10 12:40 GMT+03:00 Denis Jakovlev <yadenis at seznam.cz>:
Dobrý den,

About a month ago, I asked here about connection problems with the video. And to me strongly suggest it sipjs. I can not change the library every month.
The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can not I use them? 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pátek 10. července 2015, 11:05:07, napsal jste:


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. 775-415-382

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Igor
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