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

[Tito Cumpen]

http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1



On Mar 18, 2017 11:06 AM, "Michael Jerris" <mike at jerris.com> wrote:

> webrtc_media=true
>
> On Sat, Mar 18, 2017 at 8:50 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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>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
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>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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