yes - a tutorial was linked in the announcement which includes setting
up rtpengine with opensips+websockets to do rtp<->dtls-srtp interop.
http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1
-Eric
On 03/19/2015 02:32 PM, Tito Cumpen wrote:
Great news,
Are there any media engines that can be used in conjunction with
opensips that would allow the interop between webrtc sip clients and
standard sip? I am aware that freeswitch will currently do this.
Thanks,
Tito
On Wed, Mar 18, 2015 at 3:36 AM, Saúl Ibarra Corretgé
<[email protected] <mailto:[email protected]>> wrote:
Congrats to the team! Great work!
On 17 Mar 2015, at 20:00, Răzvan Crainea <[email protected]
<mailto:[email protected]>> wrote:
> Hello, all!
>
> Aaand, we're finally making it official: OpenSIPS 2.1 will have
*WebSockets* support!
>
> Are you planning to use (or perhaps you're already using) WebRTC
based SIP clients, but you are having hard time setting up the
platform? Starting from now, it has never been simpler - based on
your needs and feedback[1] we decided to implement a WebSocket
server directly in OpenSIPS.
>
> And we're doing it now! Starting with the new OpenSIPS 2.1
release you will be able to plug your web-based SIP clients
directly in your OpenSIPS server using the new WebSocket transport
protocol[2].
>
> We've also setup a short tutorial[3] for you to integrate this
feature in your platform easier. Many thanks to Eric Tamme
(lirakis) for all his help with the tutorial as well as for the
intensive tests.
>
> [1] http://www.opensips.org/Community/IRCmeeting20141029
> [2] http://www.opensips.org/html/docs/modules/2.1.x/proto_ws
> [3] http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1
>
> Best regards,
> --
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> _______________________________________________
> Users mailing list
> [email protected] <mailto:[email protected]>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Saúl Ibarra Corretgé
AG Projects
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