Maybe I can plan a meeting time next week where we can all get together and plan out an approach for automated testing that can be done with little to no human intervention!

Who's up for that? Say next Tuesday?

On Fri, Jul 10, 2015 at 9:10 AM, Blake Priddy <bpriddy@bryantschools.org> wrote:

Brian I would love to be a part of that. I would also like to record it and post to YouTube such as a video "somewhat tutorial/best practices" about how this works. I think a freeswitch YouTube channel (which already exists) that had video about implementation of mods and how all these things work together would be great for the community! :)

On Jul 10, 2015 9:07 AM, "Brian West" <brian@freeswitch.org> wrote:
I need to have a POW WOW with the community and devise an automated testing approach that'll help us and the community.  

I think the community would benefit from a collaborative testing frame work, that allows us to present a set of machines public facing and some automation to help us test things on a continual basis.

Who's up for the challenge of starting with a clean slate and building a testing infrastructure from the ground up?



On Fri, Jul 10, 2015 at 8:44 AM, 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> 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?

--
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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,
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_________________________________________________________________________
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ClueCon 2015 Call for Speakers | Register TODAY! | Reddit: /r/freeswitch

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