[asterisk-users] WebRTC as Softphone substitute ?

Olivier oza.4h07 at gmail.com
Tue Oct 2 03:02:21 CDT 2018


@Nasir:
Thanks for replying here.

Did you met in your deployments, the kind of stability issues Carlos
reported earlier ?

Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal <nasir at ictinnovations.com> a
écrit :

> Hi Olivior,
>
> We have recently worked on a WebRTC based agent panel. As based on my
> experience I think that WebRTC based phones are far better and cheaper then
> those soft / sip phone. the big plus is that they are easy to customize and
> developer can use the power of browser and web to build / offer features
> which are not possible with regular phones.
>
> Regarding your concern about BLF or call history, for me as a being
> developer it is just a matter of customization.
>
> Regards
>
> Nasir Iqbal
>
> ICTBroadcast - an Auto Dialer software for ITSP
> <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition>
> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns
> http://www.ictbroadcast.com/
>
>
> On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez <cursor at telecomab.mx> wrote:
>
>> On 9/26/18 10:20 AM, Matthew Fredrickson wrote:
>>
>> > On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez <cursor at telecomab.mx>
>> wrote:
>> >> On 9/26/2018 4:46 AM, Olivier wrote:
>> >>
>> >>> Hello,
>> >>>
>> >>> This morning, I asked myself if WebRTC could be a viable alternative
>> >>> to softphone deployment.
>> >>>
>> >>> For me, main issue with Softphones is the amount of work needed for
>> >>> installation and configuration.
>> >>> Also, Softphones must be carefully choosen if Deskphone-like quality
>> >>> is expected.
>> >>>
>> >>> Now that WebRTC becomes ubiquitous, it might make sense to trade
>> >>> Softphone features (call history, BLF, ...) for WebRTC deployment
>> >>> simplicity.
>> >>>
>> >>> What do you think of this ?
>> >>> What kind of experience did you met with such WebRTC deployments ?
>> >>> What about classic telephony features (CallTransfer) ?
>> >>> Have you tried Cyber Maga Phone 2K ?
>> >>>
>> >>       If you can get it to work WebRTC is a good option.  The problem
>> is
>> >> that any changes in your network may disrupt it and even trying to
>> >> replicate your installation is difficult.  I have it working fine on my
>> >> website so customers can call us directly from our web page but I never
>> >> could get Cyber Mega Phone 2K to work on the same server.  We used
>> JSSIP
>> >> to create the webrtc phone on our website.
>> > We just updated the documentation for how to get CMP2K working on the
>> > wiki [1].  We'd love some feedback if you still have issues getting it
>> > setup so that we can improve the docs.
>> >
>> > [1]
>> https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone
>> >
>> > Best wishes,
>> > Matthew Fredrickson
>> >
>>      I followed the procedure indicated in the link but I cannot get
>> remote video.  I can only see my own feed.  We do have audio for a
>> little while.  For some reason the users get disconnected after a few
>> minutes even though you can still see your video feed on screen.  This
>> was done with Asterisk 15.6.0
>>
>> --
>> Telecomunicaciones Abiertas de México S.A. de C.V.
>> Carlos Chávez
>> +52 (55)8116-9161
>>
>>
>> --
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>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
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>
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