[Freeswitch-users] Webrtc

[Sanjay Soni]

Thanks. I also had this license thing in mind and didnt like this sipml business!
May be i should have framed my question differently. So, what i want to know is how the webrtc solution will look like with fs ? Because frankly there is no information about it anywhere except some of the mailing list old emails which also doesnt give this information.

Anthony Minessale <anthony.minessale at gmail.com> wrote:



I would be careful with sipml5.  All of the code related to it is GPLv3 which is the new version of the GPL that goes viral over a socket.  So if you make a product from this code and sell it as a service, you will be bound to the GPL license and forced to surrender your platform.

See the Web Services section in
http://neptune.billgatliff.com/GPL3.html

Also the project has changed its goals towards making a converter to change webrtc based connections into regular SIP for people who do not wish to implement the webrtc profile.......



On Mon, Jan 28, 2013 at 7:57 AM, soni4m <sanjay.sanjaysoni at gmail.com<mailto:sanjay.sanjaysoni at gmail.com>> wrote:
Hi Anthony,
Can you please explain how the overall solution will look like once this
webrtc support in FS effort completes ?
Will it be the way asterisk is used in many of these demos (One example
here
<http://highsecurity.blogspot.in/2012/12/webrtc-and-asterisk-11-using-sipml5.html>
)  In a sense that sipml5 (SIP stack in the browser) is necessarily there on
the browser side. My interest here is to understand the topology  here how
FS will help connecting the legacy SIP devices with browser and what will
come into the picture in between.
Thanks



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