webrtc-rails 0.3.4 → 0.3.5

checksums.yaml

@@ -1,7 +1,7 @@
 ---
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-  data.tar.gz: 0b83828c0c610f8bc2479da73bba332205f1d384
+  metadata.gz: 83913fb945b62320d8d09c983930387175fdda6c
+  data.tar.gz: 23ad77bb9c1d76b682b12b44147cdbf00a725d29
 SHA512:
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+  data.tar.gz: 19c44ccc0c347e78292252f13703231ca6344d974169832823bc311ec6690fca358777a29c493b338c52340768d0a1e41e92123f974009923dfe83bfcb6ff6c7

data/lib/assets/javascripts/webrtc_rails/main.js.coffee

@@ -42,7 +42,7 @@ class @WebRTC
         =>
           unless @_callAnswerReceived
             if @_webRTCReconnecting
-              @connect(remoteUserIdentifier)
+              @call(remoteUserIdentifier)
             else
               @onWebRTCConnectFailed(WebRTC.TIMEOUT)
         5000
@@ -90,6 +90,10 @@ class @WebRTC
     'OfferToReceiveAudio': true
     'OfferToReceiveVideo': true
 
+  _RTCIceCandidate: window.RTCIceCandidate || window.mozRTCIceCandidate || window.webkitRTCIceCandidate || window.msRTCIceCandidate
+  _RTCSessionDescription: window.RTCSessionDescription || window.mozRTCSessionDescription || window.webkitRTCSessionDescription || window.msRTCSessionDescription
+  _RTCPeerConnection: window.RTCPeerConnection || window.mozRTCPeerConnection || window.webkitRTCPeerConnection || window.msRTCPeerConnection
+
   _webSocketInitialize: (url, userToken) ->
     @_userToken = userToken
     @_webSocket = new WebSocket(url)
@@ -98,7 +102,7 @@ class @WebRTC
       @_sendValue('setMyToken')
       if @_wantWebRTCReconnecting
         @_wantWebRTCReconnecting = false
-        @connect(@_remoteUserIdentifier)
+        @call(@_remoteUserIdentifier)
 
     @_webSocket.onclose = (event) =>
       unless @_isWebSocketReconnectingStarted
@@ -201,7 +205,8 @@ class @WebRTC
 
   _startOutput: (localOutput) ->
     isVideo = (@localOutput? && @localOutput.tagName.toUpperCase() == 'VIDEO')
-    navigator.webkitGetUserMedia(
+    navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia || navigator.msGetUserMedia
+    navigator.getUserMedia(
       video: isVideo
       audio: true
       (stream) =>
@@ -225,7 +230,7 @@ class @WebRTC
     @_setAnswer(event)
 
   _onCandidate: (event) ->
-    candidate = new RTCIceCandidate(
+    candidate = new @_RTCIceCandidate(
       sdpMLineIndex: event.sdpMLineIndex
       sdpMid: event.sdpMid
       candidate: event.candidate
@@ -239,7 +244,7 @@ class @WebRTC
     @_sendMessage(candidate)
 
   _prepareNewConnection: ->
-    pcConfig = 'iceServers': [ "url": "stun:stun.l.google.com:19302" ]
+    pcConfig = 'iceServers': [ "urls": "stun:stun.l.google.com:19302" ]
     peer = null
 
     onRemoteStreamAdded = (event) =>
@@ -249,7 +254,7 @@ class @WebRTC
       @remoteOutput.src = ''
 
     try
-      peer = new webkitRTCPeerConnection(pcConfig)
+      peer = new @_RTCPeerConnection(pcConfig)
     catch e
       console.log('Failed to create peerConnection, exception: ' + e.message)
 
@@ -288,7 +293,7 @@ class @WebRTC
     if @_isCaller
       @_webRTCReconnecting = true
       if @_webSocket.readyState == WebSocket.OPEN
-        @connect(@_remoteUserIdentifier)
+        @call(@_remoteUserIdentifier)
       else 
         @_wantWebRTCReconnecting = true
 
@@ -307,7 +312,7 @@ class @WebRTC
     if @_peerConnection
       console.error('peerConnection alreay exist!')
     @_peerConnection = @_prepareNewConnection()
-    @_peerConnection.setRemoteDescription(new RTCSessionDescription(event))
+    @_peerConnection.setRemoteDescription(new @_RTCSessionDescription(event))
 
   _sendAnswer: (event) ->
     if !@_peerConnection
@@ -326,7 +331,7 @@ class @WebRTC
     if !@_peerConnection
       console.error('peerConnection NOT exist!')
       return
-    @_peerConnection.setRemoteDescription(new RTCSessionDescription(event))
+    @_peerConnection.setRemoteDescription(new @_RTCSessionDescription(event))
 
   _hangUp: ->
     @_stop()
@@ -335,7 +340,6 @@ class @WebRTC
 
   _stop: ->
     if @_peerConnection?
-      @_peerConnection.removeStream(@_peerConnection.getRemoteStreams()[0])
       @_peerConnection.close()
       @_peerConnection = null
     @_peerStarted = false

data/lib/webrtc_rails/daemon.rb

@@ -56,7 +56,7 @@ module WebrtcRails
             if data[:event] != 'heartbeat'
               token = data[:token]
               if token.present?
-                user = @user_class.send(@fetch_user_by_token_method, token)
+                user = @user_class.send(@fetch_user_by_token_method, token.to_s)
                 my_user_identifier = user ? user.send(@user_identifier).to_s : nil
                 if my_user_identifier.present?
                   case data[:event]

data/lib/webrtc_rails/version.rb

@@ -1,3 +1,3 @@
 module WebrtcRails
-  VERSION = "0.3.4"
+  VERSION = "0.3.5"
 end

metadata

@@ -1,7 +1,7 @@
 --- !ruby/object:Gem::Specification
 name: webrtc-rails
 version: !ruby/object:Gem::Version
-  version: 0.3.4
+  version: 0.3.5
 platform: ruby
 authors:
 - Koji Murata
@@ -144,8 +144,9 @@ required_rubygems_version: !ruby/object:Gem::Requirement
       version: '0'
 requirements: []
 rubyforge_project: 
-rubygems_version: 2.4.5.1
+rubygems_version: 2.4.5
 signing_key: 
 specification_version: 4
 summary: Simple Ruby on Rails WebRTC integration.
 test_files: []
+has_rdoc: