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Microsoft Edge support #432
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We are looking into this as part of 0.8.0 which is looking like it will be mostly the work of #426. I have tested Safari, but have no tested Edge yet. |
Hi! Does Edge 15 supported now in version 0.8.1? |
Sorry I have not tested yet. You will almost certainly need to use the WebRTC adapter. If you get it to work or need specific changes, let us know and we can try and make it happen. |
According to the release announcement, Microsoft Edge is supported in 0.8: |
I will have to have a discussion with our marketing team (hah). But I will honor their word and get some instructions for edge compatibility this afternoon. Stay tuned. |
Microsoft edge works with the WebRTC adapter and SIP.js. Simply load the adapter before SIP.js on the DOM and SIP.js has everything it needs. In my testing it appears that Edge is unhappy with SDP generated from anything that is not Edge. I am working on a base set of modifiers to get better compatibility with Edge. |
Hi, when you plan to have a stable version with the modifiers? |
Hi all, I'm testing SIP.js 0.9.2 with Edge v.41 including "adapter.js" before "SIP.js" and not works! The connection via web sockets is being disconnected continuously. Any idea? Thanks, |
If you could post some logs of the issue, I might be able to point you in the right direction |
Hi Jim, I am connecting to a FreeSWITCH server and every 30 seconds I get these traces in the Edge browser console after registration: SCRIPT12030: WebSocket Error: Network Error 12030, The connection with the server was terminated abnormally. Thanks, |
Based on only that- it seems like this is an issue with edge and websockets that isn't related to SIP.js at all. Without logs from FreeSWITCH or SIP.js I can't really be confident though- does it work in chrome? |
Hi JIm, Yes, it works in Chrome and Firefox. I've solved the problem with Edge periodically sending requests to web sockets to keep them active: Now, I've new errors and warnings: Any idea? |
Sending those requests manually seems like something the keepAlive system should take care of- websockets can have their own system to keep the connection alive, but in the event of that not working/existing SIP.js has its keepAlive system that's configurable with the |
Hi Jim, I've configured KeepAliveInterval to 20 (seconds) and it doesn't work. Every 30 seconds I've the same error reported 3 days ago. SIP.js with Edge only works sending requests to web sockets to keep them active: Is there any open issue about this? Thanks, |
Hi, Can someone confirm that Edge is supported by the last version 0.11.2 ? I'm trying to use it with the last adapter.js and I have the following error and no audio: Timeout for addRemoteCandidate. Consider sending an end-of-candidates notification Cyrille |
@ZikiBe can you provide full logs with traceSip enabled? There are a few ways that we can possibly make this better. |
@egreenmachine I am using version 0.13.6 with adapter.js (https://github.com/webrtc/adapter) but it seems not to support for Edge? |
@ducdan your issue is almost certainly not a SIP.js issue. I cannot tell for sure without logs. |
HTML1300: Navigation occurred. Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | configuration parameters after validation: Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · viaHost: "r0l4qtim4899.invalid" Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · uri: sip:100@webrtc.sagantel4.tk Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · custom: {} Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · displayName: "" Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · password: NOT SHOWN Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · register: true Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · registerOptions: {} Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · transportConstructor: r Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · transportOptions: {"wsServers":["wss://webrtc.sagantel4.tk:8443"]} Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · userAgentString: "SIP.js/0.13.6" Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · noAnswerTimeout: 60000 Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · hackViaTcp: false Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · hackIpInContact: false Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · hackWssInTransport: false Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · hackAllowUnregisteredOptionTags: false Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · sessionDescriptionHandlerFactoryOptions: {"constraints":{},"peerConnectionOptions":{}} Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · extraSupported: [] Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · contactName: "bo9u8if5" Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · contactTransport: "ws" Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · forceRport: false Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · autostart: true Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · autostop: true Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · rel100: "none" Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · dtmfType: "info" Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · replaces: "none" Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · sessionDescriptionHandlerFactory: function (e,t){return new r(e.ua.getLogger("sip.invitecontext.sessionDescriptionHandler",e.id),new h.SessionDescriptionHandlerObserver(e,t),t)} Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · authenticationFactory: undefined Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · allowLegacyNotifications: false Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · allowOutOfDialogRefers: false Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · authorizationUser: "100" Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · sipjsId: "ln8rd" Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | · hostportParams: "webrtc.sagantel4.tk" Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.registercontext | configuration parameters for RegisterContext after validation: Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.registercontext | · expires: 600 Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.registercontext | · extraContactHeaderParams: [] Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.registercontext | · instanceId: undefined Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.registercontext | · params: {"toUri":{"parameters":{},"type":38,"headers":{},"raw":{"scheme":"sip","user":"100","host":"webrtc.sagantel4.tk"},"normal":{"scheme":"sip","user":"100","host":"webrtc.sagantel4.tk"}},"toDisplayName":"","callId":"cmc6qn90cj832vlbj4064k","cseq":3550} Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.registercontext | · regId: undefined Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.registercontext | · registrar: {"parameters":{},"type":38,"headers":{},"raw":{"scheme":"sip","host":"webrtc.sagantel4.tk"},"normal":{"scheme":"sip","host":"webrtc.sagantel4.tk"}} Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.ua | user requested startup... Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.transport | configuration parameters after validation: Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.transport | · wsServers: [{"wsUri":"wss://webrtc.sagantel4.tk:8443","sipUri":"sip:webrtc.sagantel4.tk:8443;transport=ws;lr>","weight":0,"isError":false,"scheme":"WSS"}] Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.transport | · connectionTimeout: 5 Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.transport | · maxReconnectionAttempts: 3 Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.transport | · reconnectionTimeout: 4 Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.transport | · keepAliveInterval: 0 Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.transport | · keepAliveDebounce: 10 Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.transport | · traceSip: false Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.transport | connecting to WebSocket wss://webrtc.sagantel4.tk:8443 Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket wss://webrtc.sagantel4.tk:8443 connected Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | SessionDescriptionHandlerOptions: {"constraints":{},"peerConnectionOptions":{}} Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | initPeerConnection Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | New peer connection created Mon Mar 11 2019 23:26:22 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | acquiring local media Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | acquired local media streams Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | createOffer Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | resetIceGatheringComplete Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | Setting local sdp. Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | sdp is v=0 o=thisisadapterortc 06267784760916295 0 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE zy4u8vwgy7 a=ice-options:trickle m=audio 9 UDP/TLS/RTP/SAVPF 104 102 0 8 103 97 13 118 101 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 a=rtpmap:104 SILK/16000 a=rtcp-fb:104 x-cinfo a=rtcp-fb:104 x-bwe a=rtcp-fb:104 x-message app send:dsh recv:dsh a=rtpmap:102 opus/48000/2 a=rtcp-fb:102 x-cinfo a=rtcp-fb:102 x-bwe a=rtcp-fb:102 x-message app send:dsh recv:dsh a=rtpmap:0 PCMU/8000 a=rtcp-fb:0 x-cinfo a=rtcp-fb:0 x-bwe a=rtcp-fb:0 x-message app send:dsh recv:dsh a=rtpmap:8 PCMA/8000 a=rtcp-fb:8 x-cinfo a=rtcp-fb:8 x-bwe a=rtcp-fb:8 x-message app send:dsh recv:dsh a=rtpmap:103 SILK/8000 a=rtcp-fb:103 x-cinfo a=rtcp-fb:103 x-bwe a=rtcp-fb:103 x-message app send:dsh recv:dsh a=rtpmap:97 RED/8000 a=rtpmap:13 CN/8000 a=rtpmap:118 CN/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 events=0-16 a=maxptime:100 a=rtcp-mux a=extmap:1 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:3 http://skype.com/experiments/rtp-hdrext/fast_bandwidth_feedback#version_2 a=ice-ufrag:2Myt a=ice-pwd:Vo3vJgMOlfkQLZtfyjEXn6mR a=setup:actpass a=fingerprint:sha-256 0C:57:EE:95:0A:23:04:BD:46:E1:AA:15:4C:A6:86:63:53:51:D4:C6:01:1E:E0:B2:D9:3C:66:FD:B9:D9:AD:6B a=mid:zy4u8vwgy7 a=sendrecv a=msid:B22639FD-1282-43A6-8D00-A8CA62B0B292 998F71F8-6B66-4B72-8D41-00409E89C71B a=ssrc:1001 msid:B22639FD-1282-43A6-8D00-A8CA62B0B292 998F71F8-6B66-4B72-8D41-00409E89C71B a=ssrc:1001 cname:76hs4dkcj9 a=rtcp-rsize sip-0.13.6.min.js (1,66156) Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | waitForIceGatheringComplete Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | ICE is not complete. Returning promise Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | RTCIceGatheringState changed: gathering Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate received: candidate:1 1 UDP 2130706431 192.168.1.111 55618 typ host ufrag 2Myt Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate received: candidate:2 1 TCP 1684798975 192.168.1.111 55618 typ srflx raddr 192.168.1.111 rport 55618 tcptype active ufrag 2Myt Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | RTCIceGatheringState changed: complete Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.inviteclientcontext | closing INVITE session ln8rdnchcjfs1b752qkvgfuuo251ub Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | closing PeerConnection Mon Mar 11 2019 23:26:36 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | resetIceGatheringComplete Mon Mar 11 2019 23:27:06 GMT+0700 (SE Asia Standard Time) | sip.transport | Transport error: The Websocket had an error Mon Mar 11 2019 23:27:06 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket disconnected (code: 1006) Mon Mar 11 2019 23:27:06 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket closed without SIP.js requesting it Mon Mar 11 2019 23:27:06 GMT+0700 (SE Asia Standard Time) | sip.transport | trying to reconnect to WebSocket wss://webrtc.sagantel4.tk:8443 (reconnection attempt 1) SCRIPT12030: SCRIPT12030: WebSocket Error: Network Error 12030, The connection with the server was terminated abnormally Mon Mar 11 2019 23:27:06 GMT+0700 (SE Asia Standard Time) | sip.transport | connecting to WebSocket wss://webrtc.sagantel4.tk:8443 Mon Mar 11 2019 23:27:06 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket wss://webrtc.sagantel4.tk:8443 connected **Mon Mar 11 2019 23:27:08 GMT+0700 (SE Asia Standard Time) | sip.transport | Transport error: The Websocket had an error Mon Mar 11 2019 23:27:08 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket disconnected (code: 1006) Mon Mar 11 2019 23:27:08 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket closed without SIP.js requesting it** Mon Mar 11 2019 23:27:08 GMT+0700 (SE Asia Standard Time) | sip.transport | trying to reconnect to WebSocket wss://webrtc.sagantel4.tk:8443 (reconnection attempt 1) SCRIPT12157: SCRIPT12157: WebSocket Error: Network Error 12157, An error occurred in the secure channel support Mon Mar 11 2019 23:27:08 GMT+0700 (SE Asia Standard Time) | sip.transport | connecting to WebSocket wss://webrtc.sagantel4.tk:8443 Mon Mar 11 2019 23:27:08 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket wss://webrtc.sagantel4.tk:8443 connected Mon Mar 11 2019 23:27:38 GMT+0700 (SE Asia Standard Time) | sip.transport | Transport error: The Websocket had an error Mon Mar 11 2019 23:27:38 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket disconnected (code: 1006) Mon Mar 11 2019 23:27:38 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket closed without SIP.js requesting it Mon Mar 11 2019 23:27:38 GMT+0700 (SE Asia Standard Time) | sip.transport | trying to reconnect to WebSocket wss://webrtc.sagantel4.tk:8443 (reconnection attempt 1) SCRIPT12030: SCRIPT12030: WebSocket Error: Network Error 12030, The connection with the server was terminated abnormally Mon Mar 11 2019 23:27:38 GMT+0700 (SE Asia Standard Time) | sip.transport | connecting to WebSocket wss://webrtc.sagantel4.tk:8443 Mon Mar 11 2019 23:27:38 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket wss://webrtc.sagantel4.tk:8443 connected Mon Mar 11 2019 23:28:08 GMT+0700 (SE Asia Standard Time) | sip.transport | Transport error: The Websocket had an error Mon Mar 11 2019 23:28:08 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket disconnected (code: 1006) Mon Mar 11 2019 23:28:08 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket closed without SIP.js requesting it Mon Mar 11 2019 23:28:08 GMT+0700 (SE Asia Standard Time) | sip.transport | trying to reconnect to WebSocket wss://webrtc.sagantel4.tk:8443 (reconnection attempt 1) SCRIPT12030: SCRIPT12030: WebSocket Error: Network Error 12030, The connection with the server was terminated abnormally Mon Mar 11 2019 23:28:08 GMT+0700 (SE Asia Standard Time) | sip.transport | connecting to WebSocket wss://webrtc.sagantel4.tk:8443 Mon Mar 11 2019 23:28:08 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket wss://webrtc.sagantel4.tk:8443 connected @egreenmachine here is full log running on Edge. The web socket has been connected but it showed the error above |
@ducdan There is nothing in the logs that points to a SIP.js issue. It appears there is an issue with your websocket server. |
I just ran a bunch of scenarios with Microsoft Edge on Windows 10. You will need adapter.js but besides that it appears that everything is working properly. Due to that I feel that the original intent of this ticket is satisfied so I am going to close this. |
Hello team,
I know about #223 . However latest release of Edge has all required API implemented. (for instance Janus Webrtc gateway works perfectly in Edge https://janus.conf.meetecho.com/echotest.html )
Are you planning to add Edge support in nearest future?
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