SIP web socket

[samuel]

Desde aquí, desear suerte en el mundo de los dinosaurios del ietf a los autores del draft 
     WebSocket Transport for Session Initiation Protocol (SIP)
                   draft-ibc-rtcweb-sip-websocket-00
https://tools.ietf.org/html/draft-ibc-rtcweb-sip-websocket-00
aupa!
Samuel.

[Suman Gandham]

Hi ,

I been looking for something similar for one of my projects. I saw
Twilio , Tropo and Phono have a Jscript based websocket SIP dialer but
none are opensource.

Phono they say we can integrate with Freeswitch or any other opensource
but not sure . if anyone interested to help me out with this please let
me know and we can discuss the project in detail .

Thanks and Regards
Suman Gandham
+34-616162302

On 9/14/11 08:27 AM, samuel wrote:
> Desde aqu�, desear suerte en el mundo de los dinosaurios del ietf a los

> --
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[Iñaki Baz Castillo]

2011/9/14 Suman Gandham <su...@perplenu.com>:

> I saw Twilio , Tropo and Phono have a Jscript based websocket SIP dialer but
> none are opensource.

Hi, that's not true. Phono does not use WebSocket neither SIP. It's
just a Flash application (media goes to Phono servers via RTMP
protocol) and the signaling is based on a custom protocol on top of
HTTP (maybe they map XMPP, but not SIP).

Phono is a *private product/service*, not a technology, neither a
protocol or specification. And it's so far from being something
related to the SIP protocol. :)

> Phono they say we can integrate with Freeswitch or any other opensource but
> not sure . if anyone interested to help me out with this please let me know
> and we can discuss the project in detail .

Phono converts tha audio from RTMP (Adobe-Flash protocol) to RTP in
*their* server, and then the server sends the call to the real
destination using SIP and RTP. Think about the latency in case the
servers of Phono are far from your location and call destinations. ;)

Regards.

--
Iñaki Baz Castillo
<i...@aliax.net>