[asterisk-users] WebRTC phone

[Jarrod Cuzens]

For the client:
JSSIP and Sipml5.

If you are going to be coding something up yourself I like the JSSIP 0.5.x
javascript interfaces. If you are simply going to use a pre-canned one then
sipml5 works pretty well and remembers your settings in localstorage. I
haven't used any closed source versions since the above works really well
for us.

For the server:
If you are using Asterisk 1.8 you'll need to front it with Kamailio and
rtpengine (or webrtc2sip but I have had stability issues with that). If you
are using a more recent asterisk then the webrtc is built in but I haven't
used (we use Kamailio and rtpengine to bridge webrtc).

If you need the kamailio config I can send it to you (it gets complicated).
The rtpengine works very well if you can run the kernel module and
introduced very little cpu overhead.

Thanks,
Jarrod

On Thu, Feb 26, 2015 at 2:24 PM, Carlos Chavez <cursor at telecomabmex.com>
wrote:

>     Can anyone recommend a good WebRTC phone to use with Asterisk? I do
> not mind if it is commercial or open source.  Customers are starting to ask
> for web solutions and we need to start testing.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> +52 (55)9116-91161
>
>
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