[Freeswitch-users] Convert Sip to Webrtc-like call

[Anthony Minessale]

Kamil,

sip.js can register directly to FS and any calls to that user will
automatically be webrtc media.

Dave,

Maybe submit a talk for ClueCon......

On Sat, Mar 18, 2017 at 7:47 AM Dave Horton <daveh at beachdognet.com> wrote:

> I built something similar - a sip.js-based browser coupled with a webrtc
> proxy using freeswitch for the media proxy (no verto), with a separate SIP
> signaling engine (in my case instead of opensips I used my open open source
> framework, called drachtio https://github.com/davehorton/drachtio-server).
>
> TLDR; For the inbound call to the browser, send a 3pcc INVITE to the
> browser
>
> Meaning:
>
> 1. Send INVITE with no SDP to the browser
> 2.  Get 200 OK with SDP and ICE candidates
> 3. Send INVITE to freeswitch with that SDP
> 4. Get 200 OK from freeswitch with its SDP
> 5. Send ACK to browser with SDP from freeswitch
>
> Freeswitch then bridges the call from the webrtc side to the sip side.
>
> FYI, my browser app can be seen at https://webrtc.drachtio.org/, feel
> free to try it out, with a couple of notes:
> 1) only google authentication is supported
> 2) you need to have sip credentials from a hosted VoIP provider
> 3) when you first log in, select the ‘Settings’ menu from the upper right
> and add your sip credentials
> 4) its meant to be an operator type console where you can also monitor
> other extensions in your business using sip presence, do call pickup etc.
> 5) video is currently not supported
>
> Browser app is built using google polymer material design, and code is
> here: https://github.com/davehorton/webrtc-operator-console
> Webrtc proxy app code is here: https://github.com/davehorton/ws-proxy,
> and as noted depends on drachtio server and freeswitch (using
> drachtio-fsmrf module for controlling freeswitch).
>
> For more details or help, feel free to email me directly or open issues on
> any of those projects.
>
>
>
> On Mar 18, 2017, at 6:03 AM, Kamil Nigmatullin <
> kamil.nigmatullin at gmail.com> wrote:
>
> Maybe it is a silly question, but cannot find anything. I have the
> opensips that handles auth and all routing, and freeswitch is responsible
> for handling media, prepaid and so on.
>
> Now, I want to implement browser clinent based on sip.js and therefore i
> pass all sip signalling through opensips's implementetion for websockets.
> For outgoing invites from browser clients this works.
>
> But for calls that are directed to browser clients it doesn't. I beleive
> that this is due to simlpe sip call does not have ice-candidates. So, the
> question is how to make freeswitch convert simple sip call to web-rtc
> oritenteated. I didn't implement mod_verto.
>
> Thanks in advance
>
> --
> Kamil Nigmatullin
> Skype: kamil.nigmatullin
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-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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