<html><head><title>Re: [Freeswitch-users] how to improve the performance of Web RTC?</title>
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<span style=" font-family:'Courier New'; font-size: 9pt;">Dobrý den,<br>
<br>
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.<br>
The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can not I use them? <br>
<br>
<span style=" font-family:'calibri'; color: #c0c0c0;"><i>-- <br>
S pozdravem,<br>
Ing.Denis Jakovlev <br>
mob.tel. 775-415-382<br>
<br>
pátek 10. července 2015, 11:05:07, napsal jste:<br>
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<td><span style=" font-family:'courier new'; font-size: 9pt;">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. <br>
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пятница, 10 июля 2015 г. пользователь Denis Jakovlev написал:<br>
Hi All !<br>
<br>
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. <br>
Maybe dear colleagues will share tips on how to make the connection stable? <br>
<br>
PS: I use FreeSwitch 1.7 on Debian 8. <br>
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<br>
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<span style=" font-family:'arial'; font-size: 8pt; color: #c0c0c0;"><i>-- <br>
S pozdravem,<br>
Ing.Denis Jakovlev <br>
</i></span></span><a style=" font-family:'arial'; font-size: 9pt; font-style: Italic;" href="http://mob.tel">mob.tel</a><span style=" font-family:'arial'; font-size: 9pt; color: #c0c0c0;"><i>. 775-415-382<br>
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<span style=" font-family:'courier new'; font-size: 9pt;">-- <br>
Best regards,<br>
Igor</td>
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