- From: <bugzilla@jessica.w3.org>
- Date: Fri, 18 May 2012 17:49:29 +0000
- To: public-webrtc@w3.org
https://www.w3.org/Bugs/Public/show_bug.cgi?id=17109 Summary: TURN server API changes Product: WebRTC Working Group Version: unspecified Platform: PC OS/Version: All Status: NEW Severity: normal Priority: P2 Component: WebRTC API AssignedTo: public-webrtc@w3.org ReportedBy: prakashr.ietf@gmail.com CC: public-webrtc@w3.org These are some of the questions for TURN server based on read the spec here. http://dev.w3.org/2011/webrtc/editor/webrtc.html 1. The PeerConnection constructor takes in a configuration to specify TURN/STURN server configuration. Currently it states "The configuration string gives the address of a STUN or TURN server to use to establish the connection." Shouldn't we allow to pass two different IPs for STUN and TURN and not restrict either or? 2. The configuration parameter is taken in as a string. Isn't it easier to make this an object, like we did for "audio video" in getUserMedia? 3. There should be some way for PeerConnection to get a feedback back to Javascript layer in case of an error. A relay/stun server can throw different types of errors, like unauthorized, insufficient capacity. Also username/password have a life time and can expire in a relay server. When this happens, the app will have to refresh the relay server info in some way. So we need a callback/update mechanism for these things. This does not seem to exist currently? 4. How do we pass in a specific username/password for TURN or update it incase a username/password expires? -- Configure bugmail: https://www.w3.org/Bugs/Public/userprefs.cgi?tab=email ------- You are receiving this mail because: ------- You are on the CC list for the bug. You are the assignee for the bug.
Received on Friday, 18 May 2012 17:49:32 UTC