SIPJS using Internet explorer 11

[Gustavo Calderon]

Hello,

i have the following setup:

Freeswitch with webRTC ready

Internet Explorer 11

web app with sip.js

FYI, before this, i tried on chrome browser and i could get it working.

So from Internet explorer im trying to crea a new User Agent

and i get the following input:

Thu Oct 15 2015 18:58:48 GMT-0400 (BOT) | sip.ua | · authenticationFactory: undefined

sip-0.7.0.js:2817 Thu Oct 15 2015 18:58:48 GMT-0400 (BOT) | sip.ua | · authorizationUser: "sipb_gustavo"

sip-0.7.0.js:2817 Thu Oct 15 2015 18:58:48 GMT-0400 (BOT) | sip.ua | · displayName: "Silviu"

sip-0.7.0.js:2817 Thu Oct 15 2015 18:58:48 GMT-0400 (BOT) | sip.ua | · instanceId: "9b807962-7572-4de8-bd5f-a82c8b08c9d3"

sip-0.7.0.js:2817 Thu Oct 15 2015 18:58:48 GMT-0400 (BOT) | sip.ua | · sipjsId: "73b04"

sip-0.7.0.js:2817 Thu Oct 15 2015 18:58:48 GMT-0400 (BOT) | sip.ua | · hostportParams: "208.x.x.x"

sip-0.7.0.js:2817 Thu Oct 15 2015 18:58:48 GMT-0400 (BOT) | sip.ua | · media: undefined

sip-0.7.0.js:2817 Thu Oct 15 2015 18:58:48 GMT-0400 (BOT) | sip.ua | user requested startup...

sip-0.7.0.js:2817 Thu Oct 15 2015 18:58:48 GMT-0400 (BOT) | sip.transport | connecting to WebSocket ws://208.x.x.x:5066

Thu Oct 15 2015 18:58:49 GMT-0400 (BOT) | sip.transport | WebSocket ws://208.x.x.x:5066 connected

sip-0.7.0.js:2817 Thu Oct 15 2015 18:58:49 GMT-0400 (BOT) | sip.ua | connection state set to 0

SCRIPT5007: Unable to get property 'resolve' of undefined or null reference

File: sip-0.7.0.js, Line: 9174, Column: 7

Also, when i try it on the Demo sipjs website (http://sipjs.com/demo-phone/)

i actually get sipjs register the extension, but after a couple seconds i get:

SCRIPT12030: WebSocket Error: Network Error 12030, The connection with the server was terminated abnormally

Thu Oct 15 2015 15:34:35 GMT-0700 (Pacific Daylight Time) | sip.transport | WebSocket connection error: {}

Thu Oct 15 2015 15:34:35 GMT-0700 (Pacific Daylight Time) | sip.transport | WebSocket disconnected (code: 1006)

Thu Oct 15 2015 15:34:35 GMT-0700 (Pacific Daylight Time) | sip.transport | WebSocket abrupt disconnection

Thu Oct 15 2015 15:34:35 GMT-0700 (Pacific Daylight Time) | sip.ua | connection state set to 1

Thu Oct 15 2015 15:34:35 GMT-0700 (Pacific Daylight Time) | sip.transport | Connection to WebSocket ws://208.x.x.x:5066 severed, attempting first reconnect

Thu Oct 15 2015 15:34:35 GMT-0700 (Pacific Daylight Time) | sip.transport | connecting to WebSocket ws://208.x.x.x:5066

Thu Oct 15 2015 15:34:37 GMT-0700 (Pacific Daylight Time) | sip.transport | WebSocket ws://208.x.x.x:5066 connected

Thu Oct 15 2015 15:34:37 GMT-0700 (Pacific Daylight Time) | sip.ua | connection state set to 0

SCRIPT5007: Unable to get property 'resolve' of undefined or null reference

File: sip-0.7.1.js, Line: 9232, Column: 7

Does anyone have a clue why this happens??

Actually im tied up to IE, so not using IE is not an option :(, does anyone made this work with internet explorer???

thanks alot!!

[Gustavo Calderon]

Ok so been doing some reading and 0.7.x only support "pormises" that IE doesnt have... :(((

is there a specific version that could work with IE???

[Gustavo Calderon]

Alright 0.6.4 got me connected!!!

One more thing now...

i have set in place a "getUserMedia" using flash, since IE doesnt support webRTC, my big question here is:

how do i get to pass the stream to sipjs so it uses that one???

[Gustavo Calderon]

Hmmm it actually disconnects with 

SCRIPT12030: WebSocket Error: Network Error 12030, The connection with the server was terminated abnormally

no other error shown, not even on freeswitch.

Any ideas?

El jueves, 15 de octubre de 2015, 19:14:29 (UTC-4), Gustavo Calderon escribió:

[Joseph Frazier]

i have set in place a "getUserMedia" using flash, since IE doesnt support webRTC, my big question here is: 

how do i get to pass the stream to sipjs so it uses that one???

Unless the flash thing you're using provides a WebRTC API, I strongly doubt this could work, unfortunately.

SCRIPT12030: WebSocket Error: Network Error 12030, The connection with the server was terminated abnormally

Not sure what's going on here, but note that in general, IE 11 (Microsoft Edge) is currently unsupported. See Github Issue 223 for more info: https://github.com/onsip/SIP.js/issues/223 

[Gustavo Calderon]

Thanks for answering, i appreciate it.

hmmm that a bummer, well waht im using is this https://github.com/maciejzasada/flashGetUserMedia

i believe it donestt have that webrtc api...

Do you have any suggetions tho? for this to work...? im kinda desperate right now...

[Will Mitchell]

You may want to checkout out WebRTC-specific plugins.  Your users will have to install the plugin to get it to work.  We don't officially support any, and I wouldn't be comfortable recommending any one in particular.  If you google around, you'll find something worth trying.  I *think* most of them should work more or less out of the box with SIP.js.  Although I'll say again, not officially supported, I haven't tried any in a while, your mileage may vary.

-Will

[Gustavo Calderon]

Ok, thank you!! i really wanted to make it work with this, but apparently is a dead end... :/

Here guys doesnt want to install plugins, so i think i will have to fallback to flash D:

Thanks for your time.