- From: <piranna@gmail.com>
- Date: Thu, 17 Jan 2013 22:45:52 +0100
- 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