[Configure Asterisk with webrtc support] Setting up asterisk for webrtc #asterisk #webrtc #sipml5 #configuration

asterisk_webrtc.md

Setting up Asterisk for webrtc

To set up with sipml5 I had been through the asterisk offiial site and I do recommand you to visit it.

We need to update several config file which are located on /etc/asterisk. Those filename are listed below

1. modules.conf
2. extensions.conf
3. http.conf
4. pjsip.conf
5. rtp.conf

I have posted how these file looks below with breif explaination.

1. modules.conf: Since we are using pjsip, we need to stop loading sip. As both of them cannot be used simultaneously. You can update manually or use the bash script below:

   sudo sh -c "echo 'noload => chan_sip.so' >> /etc/asterisk/modules.conf"

2. extension.conf:Add these things to the extension.conf at the end of the file. If you have just installed a fresh copy of asterisk you can even override the existing code.

   I have added two extensions, which are in fact dial plans.

   • Where helloworld just plays the hello-world music when we call in any number
   • Whereas the helloworld2, first plays the hello-world and then calls to another number, it also waits for the dtmf and plays its name based on whether the called number is registered one or not.

   [helloworld]
   exten => _X.,1,NoOp(${EXTEN})
   same => n,Playback(hello-world)
   same => n,Hangup()
   
   [helloworld2]
   exten => _X.,1,NoOp(${EXTEN})
   same => n,Playback(hello-world)
   same => n,Dial(PJSIP/${EXTEN},20)
   same => n,Read(Digits,,)
   same => n,Playback(you-entered)
   same => n,SayNumber(${Digits})
   

3. http.conf: Please update the file accordingly, or replace if you want.

   [general]
   enabled=yes
   bindaddr=0.0.0.0
   bindport=8088
   tlsenable=yes
   tlsbindaddr=0.0.0.0:8089
   tlscertfile=/etc/asterisk/keys/asterisk.pem
   

4. pjsip.conf: 199 is for web based phone 3002 and 3001 for sip clients: (like Linphone for desktop and CSipSimle for mobile)

   This file need to have:

   [transport-wss]
   type=transport
   protocol=wss
   bind=0.0.0.0
   
   [199]
   type=endpoint
   aors=199
   auth=199
   use_avpf=yes
   media_encryption=dtls
   dtls_ca_file=/etc/asterisk/keys/ca.crt
   dtls_cert_file=/etc/asterisk/keys/asterisk.pem
   dtls_verify=fingerprint
   dtls_setup=actpass
   ice_support=yes
   media_use_received_transport=yes
   rtcp_mux=yes
   context=helloworld2
   disallow=all
   allow=ulaw
   allow=opus
   
   [199]
   type=auth
   auth_type=userpass
   username=199
   password=199@pass1 
   
   [199]
   type=aor
   max_contacts=1
   remove_existing=yes
   
   
   [transport-udp]
   type=transport
   protocol=udp
   bind=0.0.0.0
   
   [3001]
   type=endpoint
   context=helloworld2
   disallow=all
   allow=ulaw
   auth=3001
   aors=3001
   
   [3001]
   type=auth
   auth_type=userpass
   password=3001pass
   username=3001
   
   [3001]
   type=aor
   max_contacts=1
   remove_existing=yes
   
   [3002]
   type=endpoint
   context=helloworld2
   disallow=all
   allow=ulaw
   auth=3002
   aors=3002
   
   [3002]
   type=auth
   auth_type=userpass
   password=3002pass
   username=3002
   
   [3002]
   type=aor
   max_contacts=1
   remove_existing=yes
   

5. rtp.conf: Need to have these on rtp.conf.

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

###Create Certificates Call the script as such:

cd /usr/local/src/asterisk-13.17.2/contrib/scripts
sudo ./ast_tls_cert -C pbx.example.com -O "My Super Company" -d /etc/asterisk/keys

• The "-C" option is used to define our host - DNS name or our IP address.
• The "-O" option defines our organizational name.
• The "-d" option is the output directory of the keys.

1. You'll be asked to enter a pass phrase for /etc/asterisk/keys/ca.key, put in something that you'll remember for later.
2. This will create the /etc/asterisk/keys/ca.crt file.
3. You'll be asked to enter the pass phrase again, and then the /etc/asterisk/keys/asterisk.key file will be created.
4. The /etc/asterisk/keys/asterisk.crt file will be automatically generated.
5. You'll be asked to enter the pass phrase a third time, and the /etc/asterisk/keys/asterisk.pem, a combination of the asterisk.key and asterisk.crt files, will be created.
6. You can then check your /etc/asterisk/keys directory to verify the new files were created, as such:

ls -w 1 /etc/asterisk/keys

And you should see:

asterisk.crt
asterisk.csr
asterisk.key
asterisk.pem
ca.cfg
ca.crt
ca.key
tmp.cfg

You can reload the asterisk by:

asterisk -rvvvvvv

or simply typing reload on Asterisk's cli.

To verify the web server is running, perform:

netstat -an | grep 8089

And you should see:

tcp        0      0 0.0.0.0:8089            0.0.0.0:*               LISTEN  

Next, to ensure these modules are loaded by Asterisk, you can perform:

asterisk -rx "module show like crypto"
asterisk -rx "module show like websocket"
asterisk -rx "module show like opus"

You should see something similar to:

# asterisk -rx "module show like crypto"
Module                         Description                              Use Count  Status      Support Level
res_crypto.so                  Cryptographic Digital Signatures         1          Running              core
1 modules loaded

# asterisk -rx "module show like websocket"
Module                         Description                              Use Count  Status      Support Level
res_http_websocket.so          HTTP WebSocket Support                   3          Running          extended
res_pjsip_transport_websocket.so PJSIP WebSocket Transport Support        0          Running              core
2 modules loaded
 
# asterisk -rx "module show like opus"
Module                         Description                              Use Count  Status      Support Level
codec_opus.so                  OPUS Coder/Decoder                       0          Running          extended
res_format_attr_opus.so        Opus Format Attribute Module             1          Running              core

Thanks for article.

1. Module opus is not loaded, how can I install it? I removed it from allow codecs. I don't know if that codec has a significant role to play

2. How can I optimize the video bandwidth? The video call sometimes buffer and freeze?

3. how can implement ringing tone in webrtc so that when webrtc calls an endpoint ,it will play ringing tone so the caller will be aware that the end point is ringing ?