Enabling WebRTC: Difference between revisions
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* Set the path for your certificate in the Web interface - Admin/Settings, Security section, WebRTC SSL path field | * Set the path for your certificate in the Web interface - Admin/Settings, Security section, WebRTC SSL path field | ||
* In your extension enable Opus Codec, RTP Encryption and Any Transport | * In your extension enable Opus Codec, WebRTC support, RTP Encryption and Any Transport | ||
== Verifying WebRTC == | == Verifying WebRTC == | ||
Revision as of 17:42, 23 March 2020
WebRTC is not enabled by default and you need to follow carefully the following steps to make it work.
- Install an SSL certificate on Asterisk. This needs to be a real SSL certificate. Check the Asterisk FAQ on how to install an SSL Certificate in Asterisk
- Install the Opus codec and add to the web interface. You can check the instructions in the Setup Guides section Installing OPUS codec
- Activate the http daemon for asterisk by editing the http.conf file as following:
[general] enabled=yes bindaddr=0.0.0.0 bindport=8088 tlsenable=yes tlsbindaddr=0.0.0.0:8089 tlscertfile=/etc/asterisk/certificates/demo.mirtapbx.com.pem
The tlscertfile needs to point to your certificate in pem format
- Activate the ws and wss transport in sip.conf, set the realm
transport=udp,tls,ws,wss realm=demo.mirtapbx.com
- Set the path for your certificate in the Web interface - Admin/Settings, Security section, WebRTC SSL path field
- In your extension enable Opus Codec, WebRTC support, RTP Encryption and Any Transport
Verifying WebRTC
To verify the WebRTC configuration, you can try to register and place calls using the SipML5 by visiting: https://www.doubango.org/sipml5/
