You should have a working
chan_pjsip based Asterisk installation to start with and for purposes of this tutorial, it must be version 15.5 or higher. Either install Asterisk from your distribution's packages or, preferably, install Asterisk from source. Either way, there are a few modules over and above the standard ones that must be present for WebSockets and WebRTC to work:
codec_opus(optional but highly recommended for high quality audio)
Technically, a client can use WebRTC over an insecure WebSocket to connect to Asterisk. In practice though, most browsers will require a TLS based WebSocket to be used. You can use self-signed certificates to set up the Asterisk TLS server but getting browsers to accept them is tricky so if you're able, we highly recommend getting trusted certificates from an organization such as LetsEncrypt.
If you already have certificate files (certificate, key, CA certificate), whether self-signed or trusted, you can skip the rest of this section. If you need to generate a self-signed certificate, read on.
Asterisk provides a utility script,
ast_tls_cert in the
contrib/scripts source directory. We will use it to make a self-signed certificate authority and a server certificate for Asterisk, signed by our new authority.
From the Asterisk source directory run the following commands. You'll be prompted to set a a pass phrase for the CA key, then you'll be asked for that same pass phrase a few times. Use anything you can easily remember. The pass phrase is indicated below with "
********". Replace "
pbx.example.com" with your PBX's hostname or IP address. Replace "
My Organization" as appropriate.
$ sudo mkdir /etc/asterisk/keys $ sudo contrib/scripts/ast_tls_cert -C pbx.example.com -O "My Organization" -d /etc/asterisk/keys No config file specified, creating '/etc/asterisk/keys/tmp.cfg' You can use this config file to create additional certs without re-entering the information for the fields in the certificate Creating CA key /etc/asterisk/keys/keys/ca.key Generating RSA private key, 4096 bit long modulus ............................................................................++ .....................++ e is 65537 (0x010001) Enter pass phrase for /etc/asterisk/keys/ca.key:******** Verifying - Enter pass phrase for /etc/asterisk/keys/ca.key:******** Creating CA certificate /etc/asterisk/keys/ca.crt Enter pass phrase for /etc/asterisk/keys/ca.key:******** Creating certificate /etc/asterisk/keys/asterisk.key Generating RSA private key, 1024 bit long modulus ........++++++ ............++++++ e is 65537 (0x010001) Creating signing request /etc/asterisk/keys/asterisk.csr Creating certificate /etc/asterisk/keys/asterisk.crt Signature ok subject=CN = pbx.example.com, O = My Organization Getting CA Private Key Enter pass phrase for /etc/asterisk/keys/ca.key:******** Combining key and crt into /etc/asterisk/keys/asterisk.pem $ ls -l /etc/asterisk/keys total 32 -rw------- 1 root root 1204 Mar 4 2019 asterisk.crt -rw------- 1 root root 574 Mar 4 2019 asterisk.csr -rw------- 1 root root 887 Mar 4 2019 asterisk.key -rw------- 1 root root 2091 Mar 4 2019 asterisk.pem -rw------- 1 root root 149 Mar 4 2019 ca.cfg -rw------- 1 root root 1736 Mar 4 2019 ca.crt -rw------- 1 root root 3311 Mar 4 2019 ca.key -rw------- 1 root root 123 Mar 4 2019 tmp.cfg
We'll use the
asterisk.key files later to configure the HTTP server.
Configure Asterisk's built-in HTTP server
To communicate with WebSocket clients, Asterisk uses its built-in HTTP server. Configure
/etc/asterisk/http.conf as follows:
[general] enabled=yes bindaddr=0.0.0.0 bindport=8088 tlsenable=yes tlsbindaddr=0.0.0.0:8089 tlscertfile=/etc/asterisk/keys/asterisk.crt tlsprivatekey=/etc/asterisk/keys/asterisk.key
If you have not used the generated self-signed certificates produced in the "Create Certificates" section then you will need to set the "
Now start or restart Asterisk and make sure the TLS server is running by issuing the following CLI command:
*CLI> http show status HTTP Server Status: Prefix: Server: Asterisk/GIT-16-a84c257cd6 Server Enabled and Bound to [::]:8088 HTTPS Server Enabled and Bound to [::]:8089 Enabled URI's: /test_media_cache/... => HTTP Media Cache Test URI /guimohdir_rh => HTTP POST mapping /httpstatus => Asterisk HTTP General Status /phoneprov/... => Asterisk HTTP Phone Provisioning Tool /amanager => HTML Manager Event Interface w/Digest authentication /backups => HTTP POST mapping /arawman => Raw HTTP Manager Event Interface w/Digest authentication /manager => HTML Manager Event Interface /rawman => Raw HTTP Manager Event Interface /static/... => Asterisk HTTP Static Delivery /amxml => XML Manager Event Interface w/Digest authentication /mxml => XML Manager Event Interface /moh => HTTP POST mapping /ari/... => Asterisk RESTful API /ws => Asterisk HTTP WebSocket <there may be more>
Note that the HTTPS Server is enabled and bound to
[::]:8089 and that the
/ws URI is enabled.
If you're not already familiar with configuring Asterisk's
chan_pjsip driver, visit the
res_pjsip configuration page.
PJSIP WSS Transport
Although the HTTP server does the heavy lifting for WebSockets, we still need to define a basic PJSIP Transport:
[transport-wss] type=transport protocol=wss bind=0.0.0.0 ; All other transport parameters are ignored for wss transports.
PJSIP Endpoint, AOR and Auth
We now need to create the basic PJSIP objects that represent the client. In this example, we'll call the client
webrtc_client but you can use any name you like, such as an extension number. Only the minimum options needed for a working configuration are shown. NOTE: It's normal for multiple objects in
pjsip.conf to have the same name as long as the types differ.
[webrtc_client] type=aor max_contacts=5 remove_existing=yes [webrtc_client] type=auth auth_type=userpass username=webrtc_client password=webrtc_client ; This is a completely insecure password! Do NOT expose this ; system to the Internet without utilizing a better password. [webrtc_client] type=endpoint aors=webrtc_client auth=webrtc_client dtls_auto_generate_cert=yes webrtc=yes ; Setting webrtc=yes is a shortcut for setting the following options: ; use_avpf=yes ; media_encryption=dtls ; dtls_verify=fingerprint ; dtls_setup=actpass ; ice_support=yes ; media_use_received_transport=yes ; rtcp_mux=yes context=default disallow=all allow=opus,ulaw
An explanation of each of these settings parameters can be found on the Asterisk 15 Configuration_res_pjsip page. Briefly:
- Declare an endpoint that references our previously-made aor and auth.
- Notify Asterisk to expect the AVPF profile (secure RTP)
- Setup the DTLS method of media encryption.
- Specify which certificate files to use for TLS negotiations with this endpoint and our verification and setup methods.
- Enable ICE support
- Tell Asterisk to send media across the same transport that we receive it from.
- Enable mux-ing of RTP and RTCP events onto the same socket.
- Place received calls from this endpoint into an Asterisk Dialplan context called "default"
- And setup codecs by first disabling all and then selectively enabling Opus (presuming that you installed the Opus codec for Asterisk as mentioned at the beginning of this tutorial), then G.711 μ-law.
Restart Asterisk to pick up the changes and if you have a firewall, don't forget to allow TCP port 8089 through so your client can connect.
At this point, your WebRTC client should be able to register and make calls. If you've used self-signed certificates however, your browser may not allow the connection and because the attempt is not from a normal URI supplied by the user, the user might not even be notified that there's an issue. You may be able to get the browser to accept the certificate by visiting "
https://pbx.example.com:8089/ws" directly. This will usually result in a warning from the browser and may give you the opportunity to accept the self-signed certificate and/or create an exception. If you generated your certificate from a pre-existing local Certificate Authority, you can also import that Certificate Authority's certificate into your trusted store but that procedure is beyond the scope of this document.