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 <[email protected]> wrote: > On 9/26/18 10:20 AM, Matthew Fredrickson wrote: > > > On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez <[email protected]> > 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 > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
