Re: [rtcweb] New Draft - WebRTC JS Object API Model

Iñaki Baz Castillo <ibc@aliax.net> Mon, 01 July 2013 17:22 UTC

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 1 Jul 2013 19:22:24 +0200
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To: Ted Hardie <ted.ietf@gmail.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] New Draft - WebRTC JS Object API Model
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2013/7/1 Ted Hardie <ted.ietf@gmail.com>om>:
>> Ted, the media can be *perfectly* sent in-the-wire with any custom
>> format the web developer builts (JSON, XML, plain SDP or whatever).
>
>
> I'm sorry, I think I must misunderstand you.  You are planning to take media
> from a source  and encode it using json, xml, or SDP and send that over the
> wire?  Or you mean the media *signalling* can use json, xml, or SDP and go
> over the wire?

Of course, I mean "the media signaling". But that is not just about
"in-the-wire". Current WebRTC specs mandate that the information unit
exchanged between the JS app and the WebRTC stack must be a SDP blob.

What I mean is:

- The JS app uses the WebRTC API to retrieve a JS Object (or Objects)
with transport and media info.

- The JS app serializes such a information in any custom way defined
by the website developer (this can be a JSON body, XML, SDP! or
whatever).

- The JS app sends the information to the remote peer.

- The JS app of the remote peer decodes the information (since it is a
JS app designed by the same domain/website) and extracts all the
transport/media fields.

- The JS app of the remote peer creates a local PC and makes calls to
API methods of PC for passing those transport/media fields.

- ...and remove O/A mandate.

But honestly, all of this is really well described and exposed in the draft:

http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-api-rationale-00





>> The receiver JS app parses the information (i.e. converts from JSON to
>> an Object), extracts the media/transport fields and pass them to the
>> PeerConnection via API methods.
>>
>>
>> It is really so hard? this is how WWW works (in contrast to how SIP
>> works).
>>
>
> As has been repeatedly mentioned, you have a peer communication here that
> has to be implemented by the browser.  That means the downloaded javascript
> application must tell a browser what transports it needs so that it can
> create the appropriate connections.  If it is using RTP/UDP, it ends up
> needing to tell the browser what enough detail to construct the RTP/UPD
> streams.  What payload format, for example, is needed.  As I said before,
> that means that having no standard format across the API between the JS
> application and the browser has very poor scaling characteristics.  You
> might replace SDP as the interface between them, but not specifying anything
> there doesn't seem to work, at least in my personal opinion.

I invite you to read the draft which describes how your concerns can
be perfectly accomplished without mandating SDP or any other format:

http://tools.ietf.org/html/draft-raymond-rtcweb-webrtc-js-obj-api-rationale-00



Best regards.


--
Iñaki Baz Castillo
<ibc@aliax.net>