11 Oct
2011
11 Oct
'11
5:14 a.m.
Hi folks,
I would like to show a project my colleage José Luis Millán and me are
working on. It's about SIP protocol running in a web browser. When
RTCweb (media capable web browsers) becomes a reality both
technologies together will allow real SIP endpoints coded in
JavaScript running in a web browser and interoperating with real SIP
networks. Bye bye Flash and custom protocol gateways.
http://sip-on-the-web.aliax.net/
Cheers.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
11 Oct
11 Oct
5:21 a.m.
11 okt 2011 kl. 11:14 skrev Iñaki Baz Castillo:
Hi folks,
I would like to show a project my colleage José Luis Millán and me are
working on. It's about SIP protocol running in a web browser. When
RTCweb (media capable web browsers) becomes a reality both
technologies together will allow real SIP endpoints coded in
JavaScript running in a web browser and interoperating with real SIP
networks. Bye bye Flash and custom protocol gateways.
http://sip-on-the-web.aliax.net/
I think this is awsome. If I understand this right, this is based on your
SIP over websocket draft, which basically adds a new transport to
SIP. So for Kamailio, we would need a new transport.
http://tools.ietf.org/html/draft-ibc-rtcweb-sip-websocket-00
/O
5:26 a.m.
2011/10/11 Olle E. Johansson <oej(a)edvina.net>et>:
I think this is awsome. If I understand this right,
this is based on your
SIP over websocket draft, which basically adds a new transport to
SIP. So for Kamailio, we would need a new transport.
http://tools.ietf.org/html/draft-ibc-rtcweb-sip-websocket-00
Right. All this stuff is based on a new transport for SIP: WebSocket.
This transport is a bit more exotic as it not a layer 4 transport
(WebSocket is a transport layer built on top of TCP/TLS, and it's a
message boundary transport as UDP or SCTP).
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
8:07 a.m.
This is really awesome.
Is any of that stuff on the video open to play with?
(I know it does not support audio but the SIP chat aspect is really great.
Thanks.
On 10/11/11 5:26 AM, Iñaki Baz Castillo wrote:
2011/10/11 Olle E. Johansson<oej(a)edvina.net>et>:
I think this is awsome. If I understand this
right, this is based on your
SIP over websocket draft, which basically adds a new transport to
SIP. So for Kamailio, we would need a new transport.
http://tools.ietf.org/html/draft-ibc-rtcweb-sip-websocket-00
Right. All this stuff
is based on a new transport for SIP: WebSocket.
This transport is a bit more exotic as it not a layer 4 transport
(WebSocket is a transport layer built on top of TCP/TLS, and it's a
message boundary transport as UDP or SCTP).
8:11 a.m.
2011/10/11 David <david(a)styleflare.com>om>:
This is really awesome.
Is any of that stuff on the video open to play with?
Not yet, but it will be.
(I know it does not support audio but the SIP chat
aspect is really great).
It's based on SIP MESSAGE. Implementing MSRP will be the next step :)
For audio/video we must wait for RTCweb to become an standard.
Thanks a lot.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
8:16 a.m.
Cool; I will wait until the code is available.
If I understand what I saw correctly you have implemented a JS SIP stack
which runs native in the browser over web-sockets.
On 10/11/11 8:11 AM, Iñaki Baz Castillo wrote:
2011/10/11 David<david(a)styleflare.com>om>:
This is really awesome.
Is any of that stuff on the video open to play with?
Not yet, but it will be.
(I know it does not support audio but the SIP
chat aspect is really great).
It's based on SIP MESSAGE. Implementing MSRP will
be the next step :)
For audio/video we must wait for RTCweb to become an standard.
Thanks a lot.
8:19 a.m.
2011/10/11 David <david(a)styleflare.com>om>:
If I understand what I saw correctly you have
implemented a JS SIP stack
which runs native in the browser over web-sockets.
Yes, it's a JavaScript code implementing a full SIP stack (parsing,
transactions, dialogs...) and uses a WebSocket connection for
sending/receiving SIP requests/responses.
There is also in the picture a SIP proxy implementing the WebSocket
transport, if not, the JavaScript client could not interoperate with
SIP networks ;)
Regards.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
5:51 a.m.
inaki,
i read your slides and have question about the example: is OverSIP sip
proxy still needed if Kamailio sip proxy/registrar would implement
websocket transport?
-- juha
10:22 a.m.
2011/10/11 Juha Heinanen <jh(a)tutpro.com>om>:
i read your slides and have question about the
example: is OverSIP sip
proxy still needed if Kamailio sip proxy/registrar would implement
websocket transport?
No, it isn't. But OverSIP is not just a proxy providing WebSocket transport ;)
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
6:05 a.m.
Iñaki Baz Castillo writes:
No, it isn't. But OverSIP is not just a proxy
providing WebSocket
transport ;)
i would prefer not to add more servers into my sip platform, because
there is already sip proxy, registrar, presence server, xcap server,
msrp server, and sems that provides several services. hopefully yours
would replace some of those rather than add yet another one.
-- juha
10:35 a.m.
2011/10/11 Juha Heinanen <jh(a)tutpro.com>om>:
i would prefer not to add more servers into my sip
platform, because
there is already sip proxy, registrar, presence server, xcap server,
msrp server, and sems that provides several services. hopefully yours
would replace some of those rather than add yet another one.
I plan to add a MSRP relay server on top of OverSIP (MSRP over TCP,
TLS and WebSocket, still to be defined) and also plan to build a
presence server, but not a SIMPLE/XCAP presence server, a new
specification instead:
http://sip-space.net/
Cheers.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
4691
days inactive
4691
days old
10 comments
4 participants
participants (4)
-
David -
Iñaki Baz Castillo -
Juha Heinanen -
Olle E. Johansson