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So, yeah, you’ll want to be a full-stack developer, plus a VoIP guru, if you want to
comfortably dive into WebRTC. We say this not to scare you off, but rather to assure
you that if you find it challenging, it’s not due to any shortcoming on your part, but
simply because this is complex, multilayered stuff.
Having said all that, it is possible to get a taste of WebRTC without all of that, and in
this chapter we’re going to configure Asterisk to support WebRTC, and run a pre-
built web application that will demonstrate the basic audio/video capabilities of
Asterisk’s WebRTC implementation. You’re still going to have that steep learning
curve, but hopefully we’ve delivered a foundation on which to build.
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Chapter 20: WebRTC
Configuring Asterisk for WebRTC
To pass calls through Asterisk using WebRTC, the PJSIP channel driver must be used.
The configuration will be similar to that of standard SIP telephones, but not identical.
For this we’ll need a transport type, which we’ll add to the /etc/asterisk/pjsip.conf file:
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0
cert_file=/home/asterisk/certs/self-signed.crt
priv_key_file=/home/asterisk/certs/self-signed.key
That’s all for editing the config file. For the rest of the PJSIP changes, we’ll be using
the database.1
We’re going to create two new subscribers named WS_PHONE_A and WS_PHONE_B. The
WebRTC client will use the credentials for these endpoints to communicate with the
PJSIP channel driver in Asterisk (i.e., to make phone calls).
Two records need to be added to the ps_aors table:
INSERT into asterisk.ps_aors
(id, max_contacts)
values ('WS_PHONE_A', 5),
('WS_PHONE_B', 5)
;
Corresponding ps_auth records are needed:
INSERT into asterisk.ps_auths
(id, auth_type, password, username)
values ('WS_PHONE_A','userpass','spiderwrench','WS_PHONE_A'),
('WS_PHONE_B','userpass','arachnoratchet','WS_PHONE_B')
;
We then create the endpoints themselves:
INSERT INTO asterisk.ps_endpoints
(id,aors,auth,context,
transport,dtls_auto_generate_cert,webrtc,disallow,allow)
VALUES
('WS_PHONE_A','WS_PHONE_A','WS_PHONE_A','sets',
'transport-tls','yes','yes','all','vp8,opus,ulaw'),
('WS_PHONE_B','WS_PHONE_B','WS_PHONE_B','sets',
'transport-tls','yes','yes','all','vp8,opus,ulaw');
1 Note that you can configure the PJSIP channel driver completely using the config file, but in this book we’re
only doing so where necessary, and otherwise are using the database for PJSIP channel configuration.
Configuring Asterisk for WebRTC
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345
In Chapter 4 we already generated our certificates, so we should be able to use them
here as well.
$ ls -l /home/asterisk/certs/
That should take care of the channel configuration for our WebRTC example.
We’ll now need to configure Asterisk’s web server to handle HTTPS.
$ sudo vim /etc/asterisk/http.conf
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/home/asterisk/certs/self-signed.crt
tlsprivatekey=/home/asterisk/certs/self-signed.key
Save and restart Asterisk.