[asterisk-users] WebRTC as Softphone substitute ?

Tue Oct 2 21:39:55 CDT 2018

Thanks for sharing this, Alex. It sounds like TURN, as a media repeater,
wouldn't work if the media must be secured (via SRTP). Is that right?

On Wed, 3 Oct 2018, 3:17 pm alex epshteyn, <alex at thirdlane.com> wrote:

> 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> 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> 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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>>>>
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>>
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>> _____________________________________________________________________
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>> 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
>>
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>
> --
> _____________________________________________________________________
> -- 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
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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>
> --
> _____________________________________________________________________
> -- 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
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