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

[Brian West]

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


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