Subject: Re: [rtcweb] Consensus call on JSEP/ROAP

From: Iñaki Baz Castillo <[email-masked]>
Date: Thu, 1 Mar 2012 10:22:10 +0100

2012/3/1 Ravindran, Parthasarathi <[email-masked]>:
> Inaki,
>
> I agree that we have two different architecture in mind.

Partha, 4 months ago you talked all the time about "SIP to ROAP
gateway". Now you talk about "SIP to JSEP" gateway. I'm not sure
whether you properly understand that ROAP/JSEP is just a signaling
protocol/mechanism/API between JavaScript and the browser's WebRTC
stack to handle *WebRTC media* sessions:

http://tools.ietf.org/html/draft-uberti-rtcweb-jsep-02#section-2
------------------------------------------------------------------
   JSEP's handling of session descriptions is simple and
   straightforward. Whenever an offer/answer exchange is needed, the
   initiating side creates an offer by calling a createOffer() API on
   PeerConnection. The application can do massaging of that offer, if it
   wants to, and then uses it to set up its local config via a
   setLocalDescription() API. The offer is then sent off to the remote
   side over its preferred signaling mechanism (e.g. WebSockets); upon
   receipt of that offer, the remote party installs it using a
   setRemoteDescription() API.
--------------------------------------------------------------------

Please read http://tools.ietf.org/html/draft-sipdoc-rtcweb-open-wire-protocol-00
(just replace ROAP with JSEP everywhere), and then please let me know
if you understand that ROAP/JSEP information can be sent in
***multiple*** ways between web browser and HTTP/WebSocket server (and
vice versa), but "something more" than a media negotiation body is
*required*.

If you still think that JSEP is "mapeable" to SIP, then why does SIP
exist? why don't you directly send an SDP body over UDP or TCP?

Response: you need SIP because you need to tell the server who you
want to call to, and that information is NOT in the SDP, but in the
INVITE's Request URI. So ok, JSEP has NOT "Request URI", so the
desired destination of the media session must be indicated in some
other way (not always needed, see example 5.3. "Poker Game" in the
above draft) and you CANNOT standarize that because each web site has
his own HTTP logic, his own users identificator format (numeric ID,
URI, token...), his own everything. Please see the examples in the
above draft.

I hope this is clear after 4 months repeating it.

> Ranjit/Inaki,
>
> After IETF-83, I can write the informational mapping document to show how to implement simple standard SIP using JSEP. Let implementers choose whether JSEP-SIP gateway or SIP over WebSocket is the way to go.

See above (JSEP is not mapeable to SIP), so you cannot build a JSEP to
SIP gateway.

It seems that you want to build the "standard-and-unique-Web-to-SIP-Gateway".
I'm sorry but this is NOT a competition. I'm not a vendor and I don't
sell gateways. And people can use SIP over WebSocket or a custom
protocol with a SIP gateway or whatever they want.

Best regards.

-- 
Iñaki Baz Castillo
<[email-masked]>