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

[Blake Priddy]

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 at 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 at 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 at 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 *
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>>>
>>> --
>>> Best regards,
>>> Igor
>>>
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>>
>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
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>>
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>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
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>
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