Re: [rtcweb] Interest and need for Websocket subprotocol - JSEP over websockets

Christer Holmberg <christer.holmberg@ericsson.com> Thu, 04 December 2014 04:46 UTC

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "ranjit@ranjitvoip.com" <ranjit@ranjitvoip.com>, "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
Thread-Topic: [rtcweb] Interest and need for Websocket subprotocol - JSEP over websockets
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Date: Thu, 04 Dec 2014 04:46:39 +0000
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Subject: Re: [rtcweb] Interest and need for Websocket subprotocol - JSEP over websockets
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Hi Ranjit,

For WebRTC access to IMS, there is no such thing as a "default protocol".

It is true that, inside IMS, SIP is used, but you can use whatever protocol you want to access   IMS, as long as the eP-CSCF (acting as a WebRTC Gateway) can map it into SIP.

Regards,

Christer


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________________________________
From: ranjit@ranjitvoip.com<mailto:ranjit@ranjitvoip.com>
Sent: ‎03/‎12/‎2014 21:02
To: Makaraju, Maridi Raju (Raju)<mailto:Raju.Makaraju@alcatel-lucent.com>
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] Interest and need for Websocket subprotocol - JSEP over websockets

Hello all

While I agree SIP over Websockets is default signaling protocol for
WebRTC while working with IMS, there could be scenarios where WebRTC
calls can get initiated from non SIP UAs like web browsers which do not
support SIP. Then in such cases, the following things could happen
1) the WebRTC client on the browser can use JSEP to send its signaling
information over WebSocket,
2) the JSEP message would then land on the WebRTC GW over WS.
3) This JSEP message would then be converted to a SIP message and then
sent to IMS core.
4) within IMS core, its a regular SIP message
5) Again in the reverse direction, WebRTC GW would convert SIP to JSEP
6) JSEP message is sent over Websocket to UE.

now we see JSEP messages getting exchanged over Websockets. so if the
websocket sub-protocol does not define the type as "jsep", then the
WebRTC GW would not know the incoming message type and hence may discard
it or its behavior may be uncertain.

Also the JSEP message needs to be enhanced to add more message types
(along with current OFFER / ANSWER) to be able to map it with standard
signaling protocol like SIP as defined in
https://tools.ietf.org/html/draft-partha-rtcweb-jsep-sip-01

Regards
Ranjit

On 2014-12-03 12:40 pm, Makaraju, Maridi Raju (Raju) wrote:
> + 1 for using SIP over WebSocket.
>
> FROM: rtcweb [mailto:rtcweb-bounces@ietf.org] ON BEHALF OF Roman
> Shpount
>  SENT: Wednesday, December 03, 2014 12:38 PM
>  TO: ranjit@ranjitvoip.com
>  CC: rtcweb@ietf.org
>  SUBJECT: Re: [rtcweb] Interest and need for Websocket subprotocol -
> JSEP over websockets
>
> Is there any reason you cannot use SIP over WebSocket
> (https://tools.ietf.org/html/rfc7118 [1])?
>
> Call signaling will require a lot more information then what is
> provided in JSEP. JSEP mostly deals with offer and answer processing.
> Signaling will also need to deal with things like who is calling, why
> they are calling, transfers, other application specific details. In
> other words, I think this is a very bad idea.
>
> _____________
>  Roman Shpount
>
> On Wed, Dec 3, 2014 at 1:31 PM, <ranjit@ranjitvoip.com> wrote:
>
> Hi
>  With websockets as a de-facto transport protocol for WebRTC signaling
> and JSEP being the format of encoding information, there is a need for
> a defining a websocket sub-protocol : jsep. So I would like to know if
> there is any interest in the community and also the views from experts
> about the need for a websocket-sub protocol for JSEP.
>
>  The main purpose of defining the sub protocol (jsep) is to make sure
> that the WebRTC client (WIC) and WebRTC server (E-CSCF) are receiving
> JSEP encoded messages.
>
>  Thanks
>  Ranjit
>
>  _______________________________________________
>  rtcweb mailing list
>  rtcweb@ietf.org
>  https://www.ietf.org/mailman/listinfo/rtcweb [2]
>
>
>
> Links:
> ------
> [1] https://tools.ietf.org/html/rfc7118
> [2] https://www.ietf.org/mailman/listinfo/rtcweb

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