once you get media stream established with webrtc, if you have issues after that with the media, it likely has nothing at all to do with the signaling method

On Friday, July 10, 2015, Denis Jakovlev <yadenis@seznam.cz> wrote:
Dobrý den,

Thanks. I also think that perhaps Verto is a good choice, but why all break if working. The only thing I want is a little tips that I can do to improve.

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

pátek 10. července 2015, 15:44:28, napsal jste:


You absolutely can use SIP.js, and infact I do. There are bugs, of course, but all software has bugs. It seems like the scare-mongering about how awful SIP in the browser is is more of a marketing effort for Verto than reality.

Verto may be great, and given the FreeSWITCH core team's track record of quality I don't doubt it, but it's not the great saviour that will fix all of your problems while making you a cup of tea, nor is SIP.js the devil that should be avoided at all costs.

There are tradeoffs to be made on both sides, and I've used SIP.js on several occasions and been very happy with my choice.

More specifically, it's very unlikely that your choice of signalling protocol will impact the stability of media flow. The suggestion to switch to Verto in this case seems premature.

On 10 July 2015 at 06:40, Denis Jakovlev <
yadenis@seznam.cz> wrote:
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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Best regards,
Igor


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