Hello, Is there any plan to support webrtc in FS? Right I've found this post about asterisk: http://blogs.digium.com/2012/05/23/asterisk-11-webrtc/ T.
I think support for this should be added soon too ! On 7/17/12, Tamas Jalsovszky <jalsot@gmail.com> wrote:
Hello,
Is there any plan to support webrtc in FS? Right I've found this post about asterisk: http://blogs.digium.com/2012/05/23/asterisk-11-webrtc/
T.
-- Thanks, Prasd
Now accepting sponsors for this effort! On Thu, Jul 19, 2012 at 4:35 PM, Prasd D <prasd.d.b@gmail.com> wrote:
I think support for this should be added soon too !
On 7/17/12, Tamas Jalsovszky <jalsot@gmail.com> wrote:
Hello,
Is there any plan to support webrtc in FS? Right I've found this post about asterisk: http://blogs.digium.com/2012/05/23/asterisk-11-webrtc/
T.
-- Thanks, Prasd
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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In all seriousness for WebRTC support you need to start with full STUN/TURN/ICE support. FreeSWITCH already has mod_opus and mod_isac so audio is supported. On Thu, Jul 19, 2012 at 6:15 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Now accepting sponsors for this effort!
On Thu, Jul 19, 2012 at 4:35 PM, Prasd D <prasd.d.b@gmail.com> wrote:
I think support for this should be added soon too !
On 7/17/12, Tamas Jalsovszky <jalsot@gmail.com> wrote:
Hello,
Is there any plan to support webrtc in FS? Right I've found this post about asterisk: http://blogs.digium.com/2012/05/23/asterisk-11-webrtc/
T.
-- Thanks, Prasd
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Anthony Minessale II
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-- Kristian Kielhofner
Oops, I was actually trying to sound serious and garner some sponsorship for the effort ;) Also anyone who wants to see OPAL H.323 working, we're accepting pledges to get Robert from OPAL to the upcoming ClueCon so we can provide moral support. He's down under so it takes some hefty coin 3k.... Maybe we should start a how you can help section on our homepage! On Thu, Jul 19, 2012 at 5:19 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
In all seriousness for WebRTC support you need to start with full STUN/TURN/ICE support. FreeSWITCH already has mod_opus and mod_isac so audio is supported.
On Thu, Jul 19, 2012 at 6:15 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Now accepting sponsors for this effort!
On Thu, Jul 19, 2012 at 4:35 PM, Prasd D <prasd.d.b@gmail.com> wrote:
I think support for this should be added soon too !
On 7/17/12, Tamas Jalsovszky <jalsot@gmail.com> wrote:
Hello,
Is there any plan to support webrtc in FS? Right I've found this post about asterisk: http://blogs.digium.com/2012/05/23/asterisk-11-webrtc/
T.
-- Thanks, Prasd
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
On Thu, Jul 19, 2012 at 3:47 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Oops, I was actually trying to sound serious and garner some sponsorship for the effort ;)
Also anyone who wants to see OPAL H.323 working, we're accepting pledges to get Robert from OPAL to the upcoming ClueCon so we can provide moral support. He's down under so it takes some hefty coin 3k....
Maybe we should start a how you can help section on our homepage!
How about doing a kickstarter for each of these items? -MC
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 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Webrtc-tp7580875p7586790.html Sent from the freeswitch-users mailing list archive at Nabble.com.
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@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....
) 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
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Webrtc-tp7580875p7586790.html Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Hi Anthony, folks, personally, I agree that GPL is a pile of pox, but the ebola version is called Affero. I did look at the sipml5 home, and the web site appears to claim that it is covered by GPL3. Note -- not GNU Affero. Did a grep and can't find Affero anywhere in their source. I had thought that it was the combination with *any* GNU Affero licensed lump that makes it viral over the socket. [Section 13 of the GNUv3 license spells that out]. Thus, if none of it is Affero-licensed, it's just evil and wrong, not level 5 toxic. Afferophobia had me looking at their web site in double quick time. panic over. all the best, Lawrence p.s. agreed -- the potentially useful bit is the webrtc2sip converter gubbins On 28 Jan 2013, at 19:12, Anthony Minessale 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@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
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Webrtc-tp7580875p7586790.html Sent from the freeswitch-users mailing list archive at Nabble.com. -- Anthony Minessale II
participants (7)
-
Anthony Minessale
-
Kristian Kielhofner
-
Lawrence Conroy
-
Michael Collins
-
Prasd D
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soni4m
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Tamas Jalsovszky