[asterisk-users] WebRTC as Softphone substitute ?

[Nasir Iqbal]

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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