Asterisk 11 and WebRTC connection

1 Answer(s)

Install SRTP :

cd ~
git clone https://github.com/cisco/libsrtp.git
cd libsrtp/
autoconf
./configure CFLAGS=-fPIC --prefix=/usr
make
make runtest
sudo make install

Install PJPROJECT :

cd ~
git clone https://github.com/asterisk/pjproject pjproject
cd pjproject/
./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr --with-external-srtp
make
sudo make install

Install UUID Development Library (not required for Asterisk 12) :
cd ~ 
sudo apt-get install uuid-dev -y (for Debian & ubuntu, libuuid-devel for CentOS)

Install Asterisk 11 :

cd ~
wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz
tar -xzf asterisk-11-current.tar.gz

Generate and install DTLS certificates for SRTP :

cd asterisk*/contrib/scripts
sudo mkdir /etc/asterisk/
sudo mkdir /etc/asterisk/keys/

To generate a self-signed SSL certificate, use the following command :

### Replace 10.x.x.x by the IP adress of your server. 10.x.x.x is intern, use a public IP if your Asterisk will be exposed over Internet.
sudo ./ast_tls_cert -C 10.x.x.x -O "Your Company" -d /etc/asterisk/keys

During this process, you will be asked to choose a key. Type in the same key every time and valid by pressing Enter key.

The certificate path in this example is /etc/asterisk/keys/asterisk.pem

Install Asterisk (Yes, you need to compile Asterisk with PJPROJECT and LIBSRTP) :

cd ~
cd asterisk*
sudo ./configure --with-pjproject --with-ssl --with-srtp
make menuselect

Check that packages pbx_realtimeres_odbcres_http_websocketres_crypto and chan_sip are activated. This is a must have in order to use WebRTC over WS or WSS in Asterisk.

make
sudo make install
sudo make config
## Recommended demo conf files with : 
sudo make samples
cd ~

Activate WebSockets ans SecureWebSockets in /etc/asterisk/http.conf (file which manage the HTTP Apache Asterisk Web instance). If you use Asterisk Realtime (ODBC) then you will have to specify the file in each peer (Row dtlscertfile & dtlsprivatekey in table sippeers). :

enabled=yes;
bindport=8088;
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlsprivatekey=/etc/asterisk/keys/asterisk.pem

Ensure that rights on folders are good : (replace AsteriskUser by the user running Asterisk Service)

sudo chown AsteriskUser. /var/run/asterisk
sudo chown -R AsteriskUser. /etc/asterisk
sudo chown -R AsteriskUser. /var/{lib,log,spool}/asterisk
sudo chown -R AsteriskUser. /usr/lib/asterisk

Create your WebRTC peers in sip.conf (duplicate to make another user) :

[1060] ; This will be WebRTC client
type=friend ;
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=password ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
transport=udp,ws,wss ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS

[1061] ; This will be the legacy SIP client
type=friend;
username=1061;
host=dynamic;
secret=password;
context=default;

Edit extensions.conf to allow each peer to call :

[default]
exten => 1060,1,Dial(SIP/1060) ; Dialing 1060 will call the SIP client registered to 1060
exten => 1061,1,Dial(SIP/1061) ; Dialing 1061 will call the SIP client registered to 1061

Start Asterisk Service :

sudo service asterisk start

Open the required ports :

[Ubuntu] :
sudo ufw allow 5060 (or whatever port you have choosen in sip.conf `udpbindaddr=10.x.x.x:5060; tcpbindaddr=10.x.x.x:5060;tlsbindaddr=10.x.x.x:5061`)
sudo ufw allow 5061
sudo ufw allow 8088 (or whatever port you have choosen in http.conf : `bindport=8088`)
sudo ufw allow 8089 (or whatever port you have choosen in http.conf : `tlsbindaddr=10.x.x.x:8089`)
sudo ufw allow 10000:20000/udp (or whatever range you have choosen in rtp.conf : `rtpstart=10000; rtpend=20000`)
[or if you are on Debian] : 
sudo iptables -A INPUT -p tcp --dport 5060 -j ACCEPT
sudo iptables -A INPUT -p tcp --dport 5061 -j ACCEPT
sudo iptables -A INPUT -p tcp --dport 8088 -j ACCEPT
sudo iptables -A INPUT -p tcp --dport 8089 -j ACCEPT
sudo iptables -A INPUT -p tcp --match multiport --dports 10000:20000 -j ACCEPT

Restart (or start) the service : sudo service asterisk restart

Test WebSockets from another machine : Install WSCAT with sudo apt-get install wscat –y

## If error "connect ECONNREFUSED" it's not OK.
## If Echo service returns your messages, it's OK.
wscat -s echo -c ws://10.x.x.x:8088/ws
## The same command with WSS should work if you've installed WSS.

Test your SIP over WebSocket using a Javascript client such as JsSIP, sipML5, WebRTComm, …

Access the SIP console using sudo asterisk -vvvvvv -g -dddddd -r to debug and trace.

To do the same with Asterisk 12, simply replace Asterisk-11 by Asterisk-12 in Asterisk install.

Here you’ll find complete conf files for Asterisk 12 using Realtime, WS, WSS (ommitting ODBC conf). I post it because you may find usefull to check if some parameter is missing in your install :

http.conf

;
; Asterisk Builtin mini-HTTP server
;
[general]
enabled=yes;
bindaddr=10.x.x.x;
bindport=8088;
tlsenable=yes          ; enable tls - default no.
tlsbindaddr=10.x.x.x:8089    ; address and port to bind to - default is bindaddr and port 8089.
tlscertfile=/etc/asterisk/keys/asterisk.pem  ; path to the certificate file (*.pem) only.
tlsprivatekey=/etc/asterisk/keys/asterisk.pem    ; path to private key file (*.pem) only.

extensions.conf (made for Realtime !!)

[general]
[globals]
;
[default]
switch =>Realtime

modules.conf (made for Realtime !!!)

[modules]
autoload=yes
preload => res_odbc.so
preload => res_config_odbc.so
noload => pbx_gtkconsole.so
load => res_musiconhold.so
noload => chan_alsa.so
noload => chan_console.so

extconfig.conf (made for Realtime !!!)

[settings]
sippeers => odbc,YourAsteriskrealtimeDB,sippeers
sipusers => odbc,YourAsteriskrealtimeDB,sippeers
extensions => odbc,YourAsteriskrealtimeDB,extensions
ps_endpoints => odbc,YourAsteriskrealtimeDB,ps_endpoints
ps_auths => odbc,YourAsteriskrealtimeDB,ps_auths

asterisk.conf

[directories](!)
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdbdir => /var/lib/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk
astsbindir => /usr/sbin
[options];
verbose = 3;
debug = 3;
highpriority = yes      ; Run realtime priority (same as -p at startup).
initcrypto = yes        ; Initialize crypto keys (same as -i at startup).
[compat];
pbx_realtime=1.6;
res_agi=1.6;
app_set=1.6;

rtp.conf

;
; RTP Configuration
;
[general];
rtpstart=10000;
rtpend=20000;
icesupport=true;
stunaddr=stun.l.google.com:19302;

sip.conf

;
; SIP Configuration for Asterisk
;
[general] 
context=default ; Default context for incoming calls. Defaults to 'default' 
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
realm=YourAsteriskREALM             ; Realm for digest authentication
                                ; defaults to "asterisk". If you set a system name in
                                ; asterisk.conf, it defaults to that system name
                                ; Realms MUST be globally unique according to RFC 3261
                                ; Set this to your host name or domain name

udpbindaddr=10.x.x.x             ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
                                ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
videosupport=yes              ; Turn on support for SIP video. You need to turn this
                                ; on in this section to get any video support at all.
                                ; You can turn it off on a per peer basis if the general
                                ; video support is enabled, but you can't enable it for
                                ; one peer only without enabling in the general section.
                                ; If you set videosupport to "always", then RTP ports will
                                ; always be set up for video, even on clients that don't
                                ; support it.  This assists callfile-derived calls and
                                ; certain transferred calls to use always use video when
                                ; available. [yes|NO|always]
rtsavepath=yes                 ; If using dynamic realtime, store the path headers
send_diversion=no              ; Default "yes"     ; Asterisk normally sends Diversion headers with certain SIP
                                                    ; invites to relay data about forwarded calls. If this option
                                                    ; is disabled, Asterisk won't send Diversion headers unless
                                                    ; they are added manually.
rtpkeepalive=2            ; Send keepalives in the RTP stream to keep NAT open (default is off - zero)(secs)
;--------------------------- SIP DEBUGGING ---------------------------------------------------
sipdebug = yes                 ; Turn on SIP debugging by default, from
                                ; the moment the channel loads this configuration
icesupport = yes;
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
;
rtcachefriends=yes             ; Cache realtime friends by adding them to the internal list
                                ; just like friends added from the config file only on a
                                ; as-needed basis? (yes|no)

rtsavesysname=yes              ; Save systemname in realtime database at registration
                                ; Default= no

rtupdate=yes                   ; Send registry updates to database using realtime? (yes|no)
                                ; If set to yes, when a SIP UA registers successfully, the ip address,
                                ; the origination port, the registration period, and the username of
                                ; the UA will be set to database via realtime.
                                ; If not present, defaults to 'yes'. Note: realtime peers will
                                ; probably not function across reloads in the way that you expect, if
                                ; you turn this option off.
rtautoclear=yes                ; Auto-Expire friends created on the fly on the same schedule
                                ; as if it had just registered? (yes|no|<seconds>)
                                ; If set to yes, when the registration expires, the friend will
                                ; vanish from the configuration until requested again. If set
                                ; to an integer, friends expire within this number of seconds
                                ; instead of the registration interval.

[authentication]
;
; DTLS-SRTP CONFIGURATION
;
dtlsenable = yes                   ; Enable or disable DTLS-SRTP support
dtlsverify = no                   ; Verify that provided peer certificate and fingerprint are valid
dtlscertfile=/etc/asterisk/keys/asterisk.pem                ; Path to certificate file to present
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem              ; Path to private key for certificate file
dtlssetup = actpass                ; Whether we are willing to accept connections, connect to the other party, or both.

[basic-options](!)                ; a template
        dtmfmode=rfc2833
        context=from-office
        type=friend

[natted-phone](!,basic-options)   ; another template inheriting basic-options
        directmedia=no
        host=dynamic

[public-phone](!,basic-options)   ; another template inheriting basic-options
        directmedia=yes

[my-codecs](!)                    ; a template for my preferred codecs
        disallow=all
        allow=ilbc
        allow=g729
        allow=gsm
        allow=g723
        allow=ulaw

[ulaw-phone](!)                   ; and another one for ulaw-only
        disallow=all
        allow=ulaw

[1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=password ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS

[1061] ; This will be the legacy SIP client
type=friend
username=1061
host=dynamic
secret=password
context=default

res_odbc.conf (only for Asterisk Realtime !!!!)

[YourAsteriskrealtimeDB]
enabled => yes
dsn => your-asterisk-BD-connector-name-as-defined-in-file-odbcinst.ini
username => YourMySQLUser
password => YourMySQLPassword
pre-connect => yes

If you use Realtime, insert the following Generic Dialplan :

INSERT INTO `extensions`    (   `context`,  `exten`,    `priority`,     `app`,  `appdata`   ) 
VALUES    (   'default',  '_X.',  1,  'Dial',     'SIP/${EXTEN}'  );

Add the certificate details to http.conf. The following worked for me.

tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/path-to/cert.pem
tlsprivatekey=/path-to/privkey.pem

 

Answered on September 10, 2019.
Add Comment

Your Answer

By posting your answer, you agree to the privacy policy and terms of service.