我已经成功地设置sip.js使用标准的非安全ws：/ /使用firefox 43星号11服务器。我可以打电话给另一个ff浏览器/硬件。 但是这并不适用于最新的铬合金，铬47，我无法得到任何audio（铬没有问我想用哪个audio，它总是被困在“获取当地媒体”）。
sip-0.7.2.min.js:36 Sat Jan 16 2016 15:02:05 GMT-0500 (EST) | sip.invitecontext.mediahandler | acquiring local media 从我读的关于铬47， https://developers.google.com/web/updates/2015/10/chrome-47-webrtc?hl=en
从Chrome 47开始，getUserMedia（）请求只允许来自安全的来源：HTTPS或localhost。
所以我想configuration星号和sipjs开始使用安全的websockets（wss），并与wss连接有问题。 铬控制台输出如下：
sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | configuration parameters after validation: 2016-01-16 14:17:01.816 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · viaHost: "192.0.2.190" 2016-01-16 14:17:01.817 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · uri: sip:[email protected] 2016-01-16 14:17:01.819 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · wsServers: [{"ws_uri":"wss://192.168.10.145:8089/asterisk/ws","sip_uri":"<sip:192.168.10.145:8089;transport=ws;lr>","weight":0,"status":0,"scheme":"WSS"}] 2016-01-16 14:17:01.821 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · password: NOT SHOWN 2016-01-16 14:17:01.822 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · registerExpires: 600 2016-01-16 14:17:01.822 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · register: true 2016-01-16 14:17:01.823 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · registrarServer: sip:192.168.10.145 2016-01-16 14:17:01.823 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · wsServerMaxReconnection: 3 2016-01-16 14:17:01.823 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · wsServerReconnectionTimeout: 4 2016-01-16 14:17:01.823 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · connectionRecoveryMinInterval: 2 2016-01-16 14:17:01.824 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · connectionRecoveryMaxInterval: 30 2016-01-16 14:17:01.824 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · keepAliveInterval: 0 2016-01-16 14:17:01.824 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · extraSupported: [] 2016-01-16 14:17:01.824 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · usePreloadedRoute: false 2016-01-16 14:17:01.825 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · userAgentString: "SIP.js/0.7.2" 2016-01-16 14:17:01.825 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · iceCheckingTimeout: 5000 2016-01-16 14:17:01.825 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · noAnswerTimeout: 30000 2016-01-16 14:17:01.826 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · stunServers: ["stun:stun.l.google.com:19302"] 2016-01-16 14:17:01.826 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · turnServers: [] 2016-01-16 14:17:01.826 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · traceSip: true 2016-01-16 14:17:01.826 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · hackViaTcp: false 2016-01-16 14:17:01.827 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · hackIpInContact: true 2016-01-16 14:17:01.827 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · hackWssInTransport: true 2016-01-16 14:17:01.827 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · hackAllowUnregisteredOptionTags: false 2016-01-16 14:17:01.828 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · contactTransport: "wss" 2016-01-16 14:17:01.828 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · forceRport: false 2016-01-16 14:17:01.829 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · autostart: true 2016-01-16 14:17:01.829 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · rel100: "none" 2016-01-16 14:17:01.830 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · replaces: "none" 2016-01-16 14:17:01.830 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · mediaHandlerFactory: function (a,c){return new b(a,c)} 2016-01-16 14:17:01.831 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · authenticationFactory: undefined 2016-01-16 14:17:01.831 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · authorizationUser: "1001" 2016-01-16 14:17:01.831 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · displayName: "bobby laptop" 2016-01-16 14:17:01.832 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · instanceId: "4143f767-8b09-4e4f-b39e-dbe70a72605b" 2016-01-16 14:17:01.832 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · sipjsId: "e64e9" 2016-01-16 14:17:01.832 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · hostportParams: "192.168.10.145" 2016-01-16 14:17:01.833 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | · media: undefined 2016-01-16 14:17:01.865 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.ua | user requested startup... 2016-01-16 14:17:01.866 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:01 GMT-0500 (EST) | sip.transport | connecting to WebSocket wss://192.168.10.145:8089/asterisk/ws 2016-01-16 14:17:03.693 sip-0.7.2.min.js:39 WebSocket connection to 'wss://192.168.10.145:8089/asterisk/ws' failed: WebSocket opening handshake was canceled 2016-01-16 14:17:03.701 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:03 GMT-0500 (EST) | sip.transport | WebSocket connection error: {"isTrusted":true} 2016-01-16 14:17:03.704 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:03 GMT-0500 (EST) | sip.transport | WebSocket disconnected (code: 1006) 2016-01-16 14:17:03.705 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:03 GMT-0500 (EST) | sip.transport | WebSocket abrupt disconnection 2016-01-16 14:17:03.705 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:03 GMT-0500 (EST) | sip.ua | transport wss://192.168.10.145:8089/asterisk/ws failed | connection state set to 2 2016-01-16 14:17:03.706 sip-0.7.2.min.js:36 Sat Jan 16 2016 14:17:03 GMT-0500 (EST) | sip.ua | next connection attempt in 2 seconds
从星号CLI，我可以看到https已启用。
stations-desktop*CLI> http show status HTTP Server Status: Prefix: /asterisk Server Enabled and Bound to 0.0.0.0:8088 HTTPS Server Enabled and Bound to 0.0.0.0:8089 Enabled URI's: /asterisk/httpstatus => Asterisk HTTP General Status /asterisk/phoneprov/... => Asterisk HTTP Phone Provisioning Tool /asterisk/static/... => Asterisk HTTP Static Delivery /asterisk/ws => Asterisk HTTP WebSocket Enabled Redirects: None.
在http.conf
[general] enabled=yes bindaddr=0.0.0.0 bindport=8088 prefix=asterisk enablestatic=yes tlsenable=yes tlsbindaddr=0.0.0.0:8089 tlscertfile=/etc/asterisk/keys/asterisk.pem tlsprivatekey=/etc/asterisk/keys/asterisk.pem
sip.conf
[general] context=public allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 transport=udp,ws,wss srvlookup=yes qualify=yes tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 [authentication] [basic-options](!) ; a template dtmfmode=rfc2833 context=from-office type=friend [webrtc](!) type=friend host=dynamic encryption=yes avpf=yes icesupport=yes context=default directmedia=no transport=ws,wss force_avp=yes dtlsenable=yes dtlsverify=no dtlscertfile=/etc/asterisk/keys/asterisk.pem dtlsprivatekey=/etc/asterisk/keys/asterisk.key dtlssetup=actpass [1001](webrtc) secret=REDACTED [1002](webrtc) secret=REDACTED
extensions.conf文件
[general] exten => 1002,1,Dial(SIP/1002) exten => 1003,1,Dial(SIP/1003) exten => 2000,1,Answer() same => n,Playback(demo-congrats) same => n,Hangup()
rtp.conf
[general] rtpstart=10000 rtpend=20000 icesupport=true stunaddr=stun.l.google.com:19302
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我正在使用自签名证书。 自签名的证书不被认为是值得信赖的。 切换到有效的SSL证书的伎俩。 Chrome 47和更高版本将需要这个。