WebRTC

WebRTC / WebSockets (WS/WSS)

WebRTC is supported from MOR X18 with PJSIP enabled.

Requirements

• Asterisk-20
• Rocky OS 9
• MOR X18 (or later)
• Domain in Asterisk server
• TLS (SSL) certificate

Enabling WebSockets

1. In Asterisk servers edit /etc/asterisk/http.conf and make sure that these settings are configured under [general] section:

[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/path/to/certificate
tlsprivatekey=/path/to/private_key

Note: these settings should be already present in http.conf, but commented out. Just uncomment required settings.

Set proper path to TLS certificate in tlscertfile and private key in tlsprivatekey.

Example with Let's Encrypt. Generate certificates as in manual and set the proper path:

[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/letsencrypt/live/your.domain.com/fullchain.pem
tlsprivatekey=/etc/letsencrypt/live/your.domain.com/privkey.pem

2. Restart Asterisk:

systemctl restart asterisk

3. Check if Asterisk HTTP server is running:

asterisk -rx "http show status"

The output should be similar to this:

HTTP Server Status:
Prefix: 
Server: Asterisk/20
Server Enabled and Bound to 0.0.0.0:8088

HTTPS Server Enabled and Bound to 0.0.0.0:8089

Enabled URI's:
/httpstatus => Asterisk HTTP General Status
/ws => Asterisk HTTP WebSocket

The important lines are:

HTTPS Server Enabled and Bound to 0.0.0.0:8089
/ws => Asterisk HTTP WebSocket

If these are present, it means that Asterisk is configured to accept calls via WebSockets protocol.

4. Check if Asterisk is accepting HTTP requests:

In browser, open the following address (with your own domain name):

https://YOUR_DOMAIN:8089/httpstatus

The output should be similar to this:

If this page is shown, Asterisk is configured to work with WebSockets protocol and WebRTC calls can be initiated using address:

https://YOUR_DOMAIN:8089/ws

Enabling WebRTC in PJSIP Devices

Only PJSIP Devices are allowed to make WebRTC calls.

In PJSIP Device settings, enable WebRTC option under Advanced settings:

Tracing WebRTC calls

Since WebRTC calls are encrypted, it is difficult to capture calls directly using standard tools such as sngrep, wireshark, etc.
Asterisk allows to capture these calls using internal SIP logger by dumping UDP packets to PCAP file (even for encrypted calls) that can be later opened in sngrep or wireshark.

Enable SIP capture for all traffic

1. Run the following asterisk command to enable SIP logging:

asterisk -rx "pjsip set logger on"

2. Save output to PCAP file:

asterisk -rx "pjsip set logger pcap /root/asterisk_sip_capture.pcap

3. When done, disable SIP logging:

asterisk -rx "pjsip set logger off"

4. Open PCAP file inwireshark or in sngrep by specifying input file:

sngrep -I /root/asterisk_sip_capture.pcap

Enable SIP capture for specific IPs

1. Run the following asterisk command to enable SIP logging for specific IP:

asterisk -rx "pjsip set logger host 192.168.0.111"

2. If needed, add additional IPs to capture (for example Provider IPs so that LegB side would be captured as well):

asterisk -rx "pjsip set logger host 192.168.0.222 add"

Note the additional keyword add at the end.

3. Save output to PCAP file:

asterisk -rx "pjsip set logger pcap /root/asterisk_sip_capture.pcap

4. When done, disable SIP logging:

asterisk -rx "pjsip set logger off"

5. Open PCAP file in wireshark or in sngrep by specifying input file:

sngrep -I /root/asterisk_sip_capture.pcap

Testing with WebRTC phone

WIP

MOR X17 (or older) WebRTC/WebSockets/WSS

WebRTC together with MOR can be integrated using 3rd party solutions (Example: https://www.mizu-voip.com/Software/WebRTCtoSIP.aspx).

With Asterisk 1.8: Please note that only audio could work, because in our experience Asterisk 1.8 which MOR uses does not work properly with video over WebRTC.

https://janus.conf.meetecho.com/