[Freeswitch-users] SIP over Websocket VS SIP over TCP

Michael Jerris mike at jerris.com
Mon Jan 12 20:44:31 MSK 2015

The only caveat to this is, if you are registering to freeswitch and use that registration to send the browser a call, it will not know to send an sdp that will work with webrtc.  You will need to tell it to do so using media_webrtc=true var on originate.

> On Jan 10, 2015, at 1:38 PM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com> wrote:
> Great to hear that. Thanks again.
> On 10 January 2015 at 20:35, Carlos Ruiz Díaz <carlos.ruizdiaz at gmail.com <mailto:carlos.ruizdiaz at gmail.com>> wrote:
> WebRTC doesn't specify a signalling protocol. This means that you can use SIP over any transport you want to carry the webRTC enabled SDP.
> FS will receive the SDP, detect that has a RTP/SAVPF profile and start handling it accordingly.
> Take for example Jitsi or IMSDroid, they both support webRTC and do SIP over UDP/TCP/TLS.
> Regards, 
> Carlos
> On Jan 10, 2015 12:08 PM, "Adam Ben-Ayoun" <adam.ben.ayoun1 at gmail.com <mailto:adam.ben.ayoun1 at gmail.com>> wrote:
> Thanks Anthony. I assume that means I can use SIP over TCP/TLS for signalling? Also, will mandatory WebRTC requirements such as DTLS-SRTP work when communicating with FS (when stuff like fingerprint, etc)?
> On 10 January 2015 at 19:49, Anthony Minessale <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
> The WebRTC media engine is driven completely by the SDP, the transport will not make any difference.
> On Fri, Jan 9, 2015 at 5:26 PM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com <mailto:adam.ben.ayoun1 at gmail.com>> wrote:
> Hi,
> We are developing a mobile client that will use the WebRTC media stack and Freeswitch as an MCU (only for conference calls). My question is, since we build a native app, can we use SIP over TCP for signalling? In other words, if Freeswitch receives the WebRTC kind of SDP, will it be able to communicate in the same way as if we were using the SIP over Websocket (the other Freeswitch option)? Any corner cases/considerations with this? Our goal is to avoid implementing SIP over Websocket on the client as much as possible.
> Thanks,
> Adam

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150112/8608d51e/attachment.html 

Join us at ClueCon 2016 Aug 8-12, 2016
More information about the FreeSWITCH-users mailing list