Use wrapper functions for webkitXXXXXX or mozXXXXXX named functions about jssip

[jmillan]

This is exactly how I was doing it ;-)

[saghul]

+1 to this. However, this doesn't qualify for 0.2.X. It doesn't change the API, but it adds new elements to the public API, so we need to bump the version to 0.3.0.

Btw, can we come up with a namespace for this? JsSIP.getUserMedia applies only to WebRTC stuff, we can do more than WebRTC, potentially.

[jmillan]

@saghul what would that namespace stand for?

[jmillan]

I guess, for new features/API changes, the best thing is doing this on a self branch. Continue with the 0.2.x development until the 0.3.0 release, when the branch would be merged in master before releasing.

Same with other features tagged for future releases.

This feature would provide interoperability with browsers that implement (http://dev.w3.org/2011/webrtc/editor/webrtc.html) other than Chrome. While it is not a big change in the code, I believe it is worth to be released as a 0.3.0 feature.

[saghul]

@jmillan YES.

[rjcoelho]

A few notes in firefox support I found:

• peerConnection.onicecandidate isn't called (https://bugzilla.mozilla.org/show_bug.cgi?id=827932), workaround is to call onSuccess inside createOffer/createAnswer, also comment out peerConnection.ononopen
• WebRTC setLocalDescription fails when offer has only G.711 codec (https://bugzilla.mozilla.org/show_bug.cgi?id=810220), workaround is to save sessionDescription into a variable
• interop w/ asterisk INVITE' faills because DTLS uses sha-256 and asterisk only supports sha-1, see http://lists.digium.com/pipermail/asterisk-dev/2013-January/058217.html

[ibc]

I propose a new file src/WebRTC.js which includes the following functions:

• JsSIP.WebRTC.getUserMedia() which calls to webkitGetUserMedia() or mozGetUserMedia().
• JsSIP.WebRTC.RTCPeerConnection() which calls to webkitRTCPeerConnection() or mozRTCPeerConnection().
• JsSIP.WebRTC.URL() which calls to webkitURL() or mozURL().

And then use those functions in the whole JsSIP code.

[ibc]

Let's check the JS WebRTC adapter code in the WebRTC demo app: https://apprtc.appspot.com/js/adapter.js

[ibc]

First approach in new "multi-browser" branch at commit 475cb16.

[jmillan]

@ibc,

This issue can be closed IMHO.

[ibc]

Done in d26448e