W3C home > Mailing lists > Public > public-webrtc@w3.org > January 2013

Re: New version of editor's draft (20130116)

From: <piranna@gmail.com>
Date: Thu, 17 Jan 2013 22:45:52 +0100
Message-ID: <CAKfGGh34tQDdTCwFZzCvPPvZ3-F52QbkCMAwCDod9Yvp1WMyYA@mail.gmail.com>
To: Adam Bergkvist <adam.bergkvist@ericsson.com>
Cc: "public-webrtc@w3.org" <public-webrtc@w3.org>
Hi Adam. On the Peer-to-Peer Data API section is specified:

"Data channel setup signaling (signaling via SDP and application
specific signaling channel or first channel via SDP and consecutive
channels via internal signaling)."

The second option seems very nice since it's very probably you know in
advance if you'll need channels on your application, although use this
first channel as signaling between the two peers for media or to
create more DataChannels without requiring anymore the initial
signaling channel is really a great feature :-D I have two doubts
here:

* reading the changes done in the latest version of the specification
it seems that when a DataChannel object is created its PeerConnection
object is not needed anymore and could be closed and also garbage
collected (previously the DataChannel needed to check the status of
the PeerConnection object and now it have been removed all references
to it). Is that correct?
* Using the initial DataChannel for signaling is a great idea, but you
need an external initial signaling channel, and for pure client-side
webapps this is a problem. Has something been though about this? I
have had some time ago the idea that since to establish the connection
both ends need their SDP (that they have already) and the other end
SDP, you could interchange "by hand" the SDP of one of the ends and
this one automatically send its SDP to the other one, probably through
the TURN server, but currently there's no way on the API to detect
messages received from the TURN/STUN server. Is it feasible, or I'm
saying silly things? If it's not possible, what other options would be
possible (specially without having to modify the API)?

Greetings, Jesús Leganés Combarro "Piranna".


2013/1/16 Adam Bergkvist <adam.bergkvist@ericsson.com>
>
> Hi
>
> A new version of the editor's draft is available.
>
> Dated version: http://dev.w3.org/2011/webrtc/editor/archives/20130116/webrtc.html
> Living document: http://dev.w3.org/2011/webrtc/editor/webrtc.html
>
> Changes include:
> * Changed AudioMediaStreamTrack to RTCDTMFSender and gave it its own section. Updated text to reflect most recent agreements. Also added examples section.
> * Replaced the localStreams and remoteStreams attributes with functions returning sequences of MediaStream objects.
> * Added spec text for attributes and methods adopted from the WebSocket interface.
> * Changed the state ENUMs and transition diagrams.
> * Aligned the data channel processing model a bit more with WebSockets (mainly closing the underlying transport).
>
> Please review and provide feedback.
>
> Adam (for the editors)
>



--
"Si quieres viajar alrededor del mundo y ser invitado a hablar en un
monton de sitios diferentes, simplemente escribe un sistema operativo
Unix."
– Linus Tordvals, creador del sistema operativo Linux
Received on Thursday, 17 January 2013 21:46:39 UTC

This archive was generated by hypermail 2.3.1 : Monday, 23 October 2017 15:19:32 UTC