I have faced an issue on integrating the demo of SIPML5 plugin on the Asterisks server. The Asterisks server version is "Asterisk 13.14.0". The new version of the asterisks server supports SRTP module. The plugin demo files are taken from Doubango's github repository. The demo integration files are placed on the root folder of our AWS server account.
Ref: https://github.com/DoubangoTelecom/sipml5/
The contents of the different configuration files for Asterisks server are as follows:
http.conf
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlsprivatekey=/etc/asterisk/keys/asterisk.pem
rtp.conf
[general]
rtpstart=10000
rtpend=20000
icesupport=yes
stunaddr=stun.l.google.com:19302
extensions.conf
[default]
exten => 100,1,Dial(SIP/1060)
exten => 101,1,Dial(SIP/1061)
exten => 102,1,Playback(1-for-am-2-for-pm)
exten => 205,1,Answer
exten => 205,2,Wait(2)
exten => 205,3,Record(asterisk-recording%d:ulaw)
exten => 205,4,Wait(2)
exten => 205,5,Playback(${RECORDED_FILE})
exten => 205,6,Wait(2)
exten => 205,7,Hangup
[from-internal]
exten => 1000,1,Answer()
same => n,Playback(demo-congrats)
same => n,Hangup()
sip.conf
[general]
udpbindaddr=0.0.0.0:5060
realm=x.x.x.x ;replace with your Asterisk server public IP address or host
transport=udp,ws,wss
externaddr=x.x.x.x ;replace with your Server's Public IP Address
websocket_enabled=true
[6001]
host=dynamic
secret=****
context=from-internal
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
The filed values set for the expert.htm page for the demo is as follows:-
WebSocket Server URL : wss://X.X.X.X:8088/ws
SIP outbound Proxy URL: udp://X.X.X.X:5060
ICE servers: [{ url: ‘stun:stun.l.google.com:19302’}]
The filed values set for the call.htm page for the demo is as follows:-
Display name: Rusty WEBRTC
Private identity: 6001
Public identity: sip:6001@X.X.X.X
Password; ****
Realm: X.X.X.X
Where X.X.X.X is my Asterisks server IP.
When click on the "Log in" button on the call.htm page an error message is displayed like
Disconnected: Failed to connect to the server
The linphone plugin is installed on the project right now. Due to the withdrawal of NPAPI support by Chrome/Mozilla browsers, the VoIP feature is completely outdated on the project. Now I want to replace the feature with SipML5 plugin which supports the WebRTC protocol. I need to check the registration, login, audio call features of the SIPML5 plugin. But all these functionalities can be implemented once i got connected to the server.
The Asterisks CLI is not displayed any error message. The console history of Chrome and Mozilla browsers are not displayed any vulnerable error messages. So I'm completely stuck on the project.
The demo files were worked fine on the Asterisks 11.8.1 server. But the SRTP module was not installed on the server so that the audio feature was not worked fine. Our supporting team is installed the library now and reinstalled the asterisks server with version 13. The firewall is open for the AWS server and the ports are open now.
Can anybody help me to fix the issue so that i can go further with the plug in integration to my project.Please let me know if any other details needed
