[asterisk-users] WebRTC as Softphone substitute ?

Tue Oct 2 21:16:29 CDT 2018

WebRTC requires a few specific things to be in place. We have blog posts that talk about WebRTC based Thirdlane Connect, but most of the information applies to WebRTC applications in general.

https://www.thirdlane.com/blog/get-up-and-running-with-thirdlane-connect

https://www.thirdlane.com/blog/nat-stun-turn-and-ice

Best regards,
Alex

Alex Epshteyn
alex at thirdlane.com
+1 (415) 261 6601
www.thirdlane.com

> On Oct 2, 2018, at 6:08 PM, Nasir Iqbal <nasir at ictinnovations.com> wrote:
> 
> @Olivior
> I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations.
> 
> Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is security feature and not a bug.
> 
> Regards
> 
> On Tue, Oct 2, 2018, 13:03 Olivier <oza.4h07 at gmail.com <mailto:oza.4h07 at gmail.com>> wrote:
> @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 <mailto: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/ <http://www.ictbroadcast.com/>
> 
> 
> On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez <cursor at telecomab.mx <mailto: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 <mailto: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 <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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