certainly sip.js is not the devil, but it has some distinct issues we address in verto. It's a hack with little value. Jssip however IS the devil, everyone using it has problems due to bugs that are fixed in sip.js that have remained open for many months in jssip with no action. We should not be recommending either sip js stack, but I would expect sip.js to generally be functional for most thingsz On Friday, July 10, 2015, Ben Langfeld <ben@langfeld.co.uk> wrote:
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 <javascript:_e(%7B%7D,'cvml','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 <http://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 * _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Best regards, Igor
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