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

[Ben Langfeld]

You might want to take a look at http://testrtc.com/ which Tsahi announced
this week. I'll be discussing it with him and his partners in detail on
Tuesday and I'll provide feedback here about how it might be relevant to
FreeSWITCH.

On 10 July 2015 at 11:06, 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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>
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>
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>
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