Configure Asterisk FreePBX
Note for configuration, asterisk http server usually uses the ports: 8088, 8089 (nonSSL/SSL).
Note for SIPML5, it need to be in an HTTPS url in order to do calls. You can register in HTTP but you are not allowed to do calls.
Required FreePBX module
Install Required Certificate Manager module, need for ssl/tls connections.
Admin > Module Admin
Install the Certificate Manager module
Configure the Asterisk FreePBX for WebRTC.
At the glance it require:
- Enable TCP connections,
- Enable the HTTPS/S server,
- Specify an STUN for WebRTC,
- Enable TLS connections,
- Prepare extensions, with DTLS, APVF, ICE and WSS transport configured
Settings > Advanced Settings
Enable the mini-HTTP Server
Enable TLS for the mini-HTTP Server
Settings > Asterisk SIP Settings
General SIP Settings > WebRTC Settings
STUN Server Address = stun.l.google.com:19302
> Chan SIP Settings > TLS/SSL/SRTP Settings
Enable TLS = Yes
Certificate Manager = default
SSL Method = sslv2
> Chan SIP Settings > Advanced General Settings
Enable TCP = Yes
Individual Extension Settings
Create a Extension for use in SIPML5, for example: 1080, and don’t forget to include the following configuration:
Transport = All – WSS Primary
Enable AVPF = yes
Enable ICE Support = Yes
Enable DTLS = Yes
Install SIPML5
Get SIPML5 at: https://github.com/DoubangoTelecom/sipml5.
Unzip to a folder in your web server root.
Point your browser to that folder (ex: https://www.ptcommerce.net/sipml5/)
Fill the SIPML5 registration form
Display Name: 1080
Private Identity*: 1080
Public Identity*: sip:1080@serverip_or_domain
Password:
Realm*: localhost
Click on Expert mode
Disable Video: Yes
SIP outbound Proxy URL: udp://serverip_or_domain:5060
ICE Servers: [url:stun.l.google.com:19302]
Disable Call button options: Yes
Try some calls
Use the call control, insert your destination and press call
Some Links
http://blog.timmattison.com/archives/2014/01/02/how-to-get-sipml5-working-with-asterisk/