[SOLVED] Asterisk 13 with Webrtc ... calls get dropped!

Asterisk Asterisk Support
21

WELL , chrome was working several hours ago … not its not working on webrtc and give me error :slight_smile:

Engine newRTCSession
VM64 html,chromewebdata:18getUserMedia() no longer works on insecure origins. To use this feature, you should consider switching your application to a secure origin, such as HTTPS. See https://goo.gl/rStTGz for more details.
(anonymous) @ VM64 html,chromewebdata:18
h @ VM64 html,chromewebdata:14
r.connect @ VM64 html,chromewebdata:14
r.call @ VM64 html,chromewebdata:17
call @ VM61:50
(anonymous) @ VM61:54
dispatch @ VM55:3
r.handle @ VM55:3
VM61:46Session failed

for the log on firefox ’ call from ex 300 webrtc to ex 100 on Xlite phone

OpenGL compositor Initialized Succesfully.
Version: 2.1 ATI-1.42.15
Vendor: ATI Technologies Inc.
Renderer: AMD Radeon R9 M370X OpenGL Engine
FBO Texture Target: TEXTURE_2D
OpenGL compositor Initialized Succesfully.
Version: 2.1 ATI-1.42.15
Vendor: ATI Technologies Inc.
Renderer: AMD Radeon R9 M370X OpenGL Engine
FBO Texture Target: TEXTURE_2D
1483046642706 addons.xpi WARN Add-on urla@kaspersky.com is not correctly signed.
1483046642706 addons.xpi WARN Add-on vkbd@kaspersky.com is not correctly signed.
1483046760378 addons.xpi WARN Add-on urla@kaspersky.com is not correctly signed.
1483046760379 addons.xpi WARN Add-on vkbd@kaspersky.com is not correctly signed.
TypeError: this.transport is null[Learn More] main.js:1429:5
Password fields present on an insecure (http://) page. This is a security risk that allows user login credentials to be stolen.[Learn More] ucp
Password fields present on an insecure (http://) page. This is a security risk that allows user login credentials to be stolen.[Learn More] ucp
The connection to ws://64.37.115.36:8088/ws was interrupted while the page was loading. jquery-1.11.1.min.js%20line%202%20%3E%20eval:19:10472
WebRTC interfaces with the “moz” prefix (mozRTCPeerConnection, mozRTCSessionDescription, mozRTCIceCandidate) have been deprecated. jquery-1.11.1.min.js%20line%202%20%3E%20eval:18:20312
onaddstream is deprecated! Use peerConnection.ontrack instead. jquery-1.11.1.min.js%20line%202%20%3E%20eval:18
navigator.mozGetUserMedia has been replaced by navigator.mediaDevices.getUserMedia jquery-1.11.1.min.js%20line%202%20%3E%20eval:18:13616
ReferenceError: webrtcDetectedType is not defined[Learn More] jsphp_90dee6bf5596e459deee6dc99ef95f36.js:45:1
WebrtcC<.manageSession/</< http://64.37.115.36/ucp/assets/js/compiled/modules/jsphp_90dee6bf5596e459deee6dc99ef95f36.js:45:1
[24]</r.prototype.emit http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:17:29756
b http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:14:19051
d http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:14:12307
h/this.receiveResponse http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:14:10994
[15]</r.prototype.receiveResponse http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:15:13146
[18]</s.prototype.receiveResponse http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:15:28343
[19]</r.prototype.onMessage http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:17:4004
[19]</r.prototype.connect/this.ws.onmessage http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:17:1835
OpenGL compositor Initialized Succesfully.
Version: 2.1 ATI-1.42.15
Vendor: ATI Technologies Inc.
Renderer: AMD Radeon R9 M370X OpenGL Engine
FBO Texture Target: TEXTURE_2D
onaddstream is deprecated! Use peerConnection.ontrack instead. jquery-1.11.1.min.js%20line%202%20%3E%20eval:18
ReferenceError: webrtcDetectedType is not defined[Learn More] jsphp_90dee6bf5596e459deee6dc99ef95f36.js:45:1
WebrtcC<.manageSession/</< http://64.37.115.36/ucp/assets/js/compiled/modules/jsphp_90dee6bf5596e459deee6dc99ef95f36.js:45:1
[24]</r.prototype.emit http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:17:29756
b http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:14:19051
d http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:14:12307
h/this.receiveResponse http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:14:10994
[15]</r.prototype.receiveResponse http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:15:13146
[18]</s.prototype.receiveResponse http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:15:28343
[19]</r.prototype.onMessage http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:17:4004
[19]</r.prototype.connect/this.ws.onmessage http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:17:1835
onaddstream is deprecated! Use peerConnection.ontrack instead. jquery-1.11.1.min.js%20line%202%20%3E%20eval:18
ReferenceError: webrtcDetectedType is not defined[Learn More] jsphp_90dee6bf5596e459deee6dc99ef95f36.js:45:1
WebrtcC<.manageSession/</< http://64.37.115.36/ucp/assets/js/compiled/modules/jsphp_90dee6bf5596e459deee6dc99ef95f36.js:45:1
[24]</r.prototype.emit http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:17:29756
b http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:14:19051
d http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:14:12307
h/this.receiveResponse http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:14:10994
[15]</r.prototype.receiveResponse http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:15:13146
[18]</s.prototype.receiveResponse http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:15:28343
[19]</r.prototype.onMessage http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:17:4004
[19]</r.prototype.connect/this.ws.onmessage http://64.37.115.36/ucp/assets/js/jquery-1.11.1.min.js%20line%202%20>%20eval:17:1835

22

Still no sip log there. You need to figure it out how to get the log from the browser this is a basic stuff to know what is happening. Without that we can go further…

So far based on what you provided there is something wrong with codecs or your peer isnt available when you call from sip to websocket. Because from wss to sip it works.

1 Like
23

is it here ?

i search a lot and i see many log types and not sure which one to choose .

24

chrome is probably complaining that your application isn’t over https

You need to serve your web application using https. webrtc won’t work in an application that is http, unless there is a flag somewhere.

25

thank you

can you just tell me how to serve my application with https ??

I’m sorry I’m newbie to webrtc stuff

cheers

26

first, you need to have a certificate, then you put it in your web server, then you configure apache to use your certificate. You’ll find plenty of tutorials to enable https in apache, juste google it.

After that, you’ll just have to access your application using https instead of http. You can redirect automaticaly all http requests to https if you want to. just google “apache redirect http to https”

27

Please go back and read the hundred post related to WEBRTC, this is like returning 3 years ago. I cannot see your image due to company policies but if you are not using HTTPS then we are wasting our time and for we I mean You and I.

1 Like
28

@navaismo thanks so much u have been great to help .

i was testing from firefox and it was ringing .

i will see if i can post logs of firefox

thank you very much

29

guys i will keep new post with new issue .

thank you so much for helping

30

Guys i wanna mark this issue as solved since the problem was in RSTP building .
i rebuilt SRTP and had all working .
@navaismo

thank you so much

31

Hi dr.x
Please tell me which version of RSTP that you have rebuilt with Asterisk v13 and how to rebuild it. I am newbie on Asterisk. Thanks a lot.

32

@dr.x I am facing with the same issue :frowning: could you please to tell me more about your solution?

33

be using chrome not firefox … i had issue with firefox … the call go and i answer it . but it keep ringing .
be using https not http for webrtc
make sure rstp is compiled before u go and install asterisk .
thats all i did and had it working .

to be honest i was using freepbx as backend

and many thanks again to @navaismo for helping me much