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onsip / SIP.js

A simple, intuitive, and powerful JavaScript signaling library
https://sipjs.com
MIT License
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Microsoft Edge support #432

Closed chornyitaras closed 5 years ago

chornyitaras commented 7 years ago

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?

egreenmachine commented 7 years ago

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.

Klimashin commented 6 years ago

Hi! Does Edge 15 supported now in version 0.8.1?

egreenmachine commented 6 years ago

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.

c960657 commented 6 years ago

According to the release announcement, Microsoft Edge is supported in 0.8: https://www.onsip.com/blog/sipjs-v0.8.0-supports-all-major-browsers-and-renegotiation

egreenmachine commented 6 years ago

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.

egreenmachine commented 6 years ago

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.

aubalde commented 6 years ago

Hi, when you plan to have a stable version with the modifiers?

aubalde commented 6 years ago

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,

JimGreenberg commented 6 years ago

If you could post some logs of the issue, I might be able to point you in the right direction

aubalde commented 6 years ago

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,

JimGreenberg commented 6 years ago

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?

aubalde commented 6 years ago

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: ua.request('NOTIFY', ext + '@eng2.presenceco.com');

Now, I've new errors and warnings: WARNING: Timeout for addRemoteCandidate. Consider sending an end-of-candidates notification. ERROR: ORTC18615: ORTC RTCDtlsTransport: error de protocolo de enlace DTLS. hr=c004e00f.

Any idea?

JimGreenberg commented 6 years ago

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 keepAliveInterval UA configuration option. I can't say with certainty that that would fix the issue, full logs would confirm.

aubalde commented 6 years ago

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: ua.request('NOTIFY', ext + '@eng2.presenceco.com');

Is there any open issue about this?

Thanks,

ZikiBe commented 6 years ago

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

egreenmachine commented 6 years ago

@ZikiBe can you provide full logs with traceSip enabled? There are a few ways that we can possibly make this better.

ducdan commented 5 years ago

@egreenmachine I am using version 0.13.6 with adapter.js (https://github.com/webrtc/adapter) but it seems not to support for Edge? SCRIPT12030: SCRIPT12030: WebSocket Error: Network Error 12030, The connection with the server was terminated abnormally

egreenmachine commented 5 years ago

@ducdan your issue is almost certainly not a SIP.js issue. I cannot tell for sure without logs.

ducdan commented 5 years ago

HTML1300: Navigation occurred. 12 (1,1)

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 sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | createOffer sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | resetIceGatheringComplete sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | Setting local sdp. sip-0.13.6.min.js (1,66156)

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 sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | ICE is not complete. Returning promise sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | RTCIceGatheringState changed: gathering sip-0.13.6.min.js (1,66156)

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 sip-0.13.6.min.js (1,66156)

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 sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | RTCIceGatheringState changed: complete sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.inviteclientcontext | closing INVITE session ln8rdnchcjfs1b752qkvgfuuo251ub sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:26:35 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | closing PeerConnection sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:26:36 GMT+0700 (SE Asia Standard Time) | sip.invitecontext.sessionDescriptionHandler | resetIceGatheringComplete sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:27:06 GMT+0700 (SE Asia Standard Time) | sip.transport | Transport error: The Websocket had an error sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:27:06 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket disconnected (code: 1006) sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:27:06 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket closed without SIP.js requesting it sip-0.13.6.min.js (1,66156)

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) sip-0.13.6.min.js (1,66156)

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 sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:27:06 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket wss://webrtc.sagantel4.tk:8443 connected sip-0.13.6.min.js (1,66156)

**Mon Mar 11 2019 23:27:08 GMT+0700 (SE Asia Standard Time) | sip.transport | Transport error: The Websocket had an error sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:27:08 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket disconnected (code: 1006) sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:27:08 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket closed without SIP.js requesting it** sip-0.13.6.min.js (1,66156)

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) sip-0.13.6.min.js (1,66156)

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 sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:27:08 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket wss://webrtc.sagantel4.tk:8443 connected sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:27:38 GMT+0700 (SE Asia Standard Time) | sip.transport | Transport error: The Websocket had an error sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:27:38 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket disconnected (code: 1006) sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:27:38 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket closed without SIP.js requesting it sip-0.13.6.min.js (1,66156)

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) sip-0.13.6.min.js (1,66156)

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 sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:27:38 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket wss://webrtc.sagantel4.tk:8443 connected sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:28:08 GMT+0700 (SE Asia Standard Time) | sip.transport | Transport error: The Websocket had an error sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:28:08 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket disconnected (code: 1006) sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:28:08 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket closed without SIP.js requesting it sip-0.13.6.min.js (1,66156)

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) sip-0.13.6.min.js (1,66156)

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 sip-0.13.6.min.js (1,66156)

Mon Mar 11 2019 23:28:08 GMT+0700 (SE Asia Standard Time) | sip.transport | WebSocket wss://webrtc.sagantel4.tk:8443 connected sip-0.13.6.min.js (1,66156)

@egreenmachine here is full log running on Edge. The web socket has been connected but it showed the error above

egreenmachine commented 5 years ago

@ducdan There is nothing in the logs that points to a SIP.js issue. It appears there is an issue with your websocket server.

egreenmachine commented 5 years ago

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.