Setting up a VoIP communication between a Raspberry Pi and an IP phone using an Asterisk IP PBX server

Context

Telegraph, analogue telephone, digital telephone, IP telephone with wired or wireless modes, telephony is currently a widely used and very convenient global link, indispensable for fast and real-time exchanges, for all purposes. It is quite reliable, not too expensive, and offers ever more extensive possibilities at an ever faster rate.

The aim of this project is to use the IP protocol used in the Internet, to make devices communicate with each other (commonly called endpoints). This project will therefore implement two endpoints with the following hardware characteristics:

• endpoint 1: Alcatel IP Touch 4018 EE phone;
• endpoint 2: Raspberry Pi 3 Model B+.

Below is a global diagram of the infrastructure architecture in the 192.168.1.0 network (the router is not shown for readability reasons):

(Figure 1 - Overall infrastructure diagram)

Simply, without going into detail, in this diagram there are two clients: the end points which are the Alcatel IP Touch 4018 EE and the Raspberry Pi. When there is a telephone communication between these two devices, the protocol used for the initiation of the communication is SIP (Session Initiation Protocol) for the application layer and UDP for the transport layer of the Internet protocol stack. This initiation goes through an intermediary: the SIP initiation server (Asterisk) which is accessible on the Debian Asterisk Server virtual machine on port 5060 (in blue). Once the initiation is done, the two devices connect to each other directly to let the audio pass via the RTP protocol (in green).

An important point concerns the Alcatel phone part, which needs at each startup to check its firmware and configuration files allowing it to connect to the Asterisk server. These files are stored in the Debian virtual machine and are available to the Alcatel via the HTTP server on port 80 (in orange).

This project details the architecture of the system as well as the implementation of the entire IP telephony infrastructure allowing communication between these two devices. Firstly, some theoretical elements on the protocols used will be looked at, in particular on the role and functioning of SIP in VoIP and RTP. Secondly, the interest of using the Asterisk system in the design of this infrastructure and its implementation will be evaluated, then the two clients (Raspberry Pi and Alcatel phone) will be configured and tested. Finally a user-friendly SIP client in JavaScript for the Raspberry Pi will be designed.

1. SIP protocol and VoIP communication

The first questions to ask before starting the implementation of this infrastructure are: What is VoIP? What is SIP and how does it relate to VoIP? What is the relationship between SIP and RTP?

Answering the first question requires understanding the internet stack (Internet protocol suite).

(Figure 2 - Internet protocol stack)

VoIP stands for Voice over Internet Protocol, voice is specifically an analogue signal which can be acquired by means of a microphone and then digitally encoded using an analogue-to-digital converter. Once encoded, the data can be passed through the Internet protocol stack. In this diagram, it is therefore possible to transfer voice encoded using the HTTP protocol, transported in TCP, in an IP network via an Ethernet link to another device using this same protocol stack. Note that not all application protocols are necessarily suitable for transferring encoded voice. Some are suitable for real-time communication, e.g. transmission of encoded voice (RTP), others for establishing a communication (SIP)...

SIP (Session Initiation Protocol) is the protocol that allows establishing a communication between endpoints. Typically, an endpoint initially acts as a SIP client to contact another endpoint. It first queries a SIP server which provides it with information about the identity of this other endpoint and how to access it in the IP network. Once the information has been received, communication takes place directly between the two endpoints via the Real-time Transport Protocol (RTP). This protocol transmits the encoded voice. Once the communication is over, SIP takes over again to close the connection (session).

(Figure 3 - Establishment and termination of a VoIP communication using SIP - attribution: 3cx.com)

Going back to our very first diagram, for the establishment of a communication (e.g. Raspberry Pi calls Alcatel):

1. RPi sends a contact request to Alcatel (Invite) through Asterisk. Alcatel receives the Invite response, rings and indicates RPi that it is ringing. Someone picks up the Alcatel, the contact is established, so...
2. ...RTP takes over and transfers the voices between the two devices in direct connection (without going through Asterisk).

(Figure 4 - Establishing a telephone call between the Raspberry Pi and the Alcatel IP Touch 4018 EE telephone)

Once the call is over (one person hangs up), SIP takes care of closing the session between the two participants.

2. Implementation of an Asterisk IP PBX server

In large organisations (companies, public services, etc.), there are mainly two types of telephone communication: internal communication and external communication. The agents therefore have an internal fixed telephone capable of making calls to another internal fixed telephone or an external (public) telephone. For this to work, a PBX, also known as a Private Branch Exchange, is needed.

The PBX (below) has many advantages, including:

• financially, the bill is reduced, as internal calls do not go through the public network (Orange, formerly France Telecom in France);
• more internal numbers can be allocated without difficulty;
• it is possible to offer services such as conference calls, call forwarding, call transfer;
• and, of course, linking an internal line to an external line.

(Figure 5 - PBX Matra MC6500 serie)
attribution: the original uploader was After310 at French Wikipedia, CC BY-SA 3.0, via Wikimedia Commons

A comprehensive list of the functions of a traditional PBX can be found on Wikipedia (1).

Now that the overall operation of this system has been presented, how can communication using SIP protocol be achieved?

Historically, there have been three main phases in the evolution of telephone communications:

1. The public switched telephone network (PSTN) which is analogue. In this network, the SIP protocol cannot be used because SIP is a digital protocol.
2. The Integrated Services Digital Network (ISDN) is a digital network whose network stack is based on the OSI model. Starting at Layer 3 (transport), the following protocols are used: Q.931 (2), X.25 layer 3 (3). Here it is Layer 3 which is problematic: SIP relies on IP to function.
3. Voice over IP, over the Internet, is digital. SIP can work in this case. The associated PBX is an IP PBX (Internet Protocol private branch exchange).

There are many IP PBXs in the world. Asterisk has the distinction of being the world's number one in terms of usage. It has a free version and a proprietary version. And it offers interoperability with older networks (PSTN and ISDN) by means of hardware cards and software modules. This last point is probably a major factor in its success: companies that still use old equipment, and that, in a perspective of modernisation, migrate to recent equipment, probably use several systems (PSTN, ISDN or VoIP), Asterisk will make it possible to manage these three systems simultaneously.

So let's proceed with the installation of Asterisk.

Technology choices

• OS: Debian 10, it is free and is mainly used as a server in the computer world. It is a system with regularly updated packages in terms of stability and security.
• Asterisk: 18 LTS release compiled from source. The version in the Debian repositories is old (16 for Debian 10) and although it is also an LTS version, the fact that it is already compiled offers less flexibility in terms of adding modules. In addition, the module that handled SIP in the 16 release has been deprecated since the 17 release (4) in favour of a new module (PJSIP) that can handle SIP as well as NAT traversal functions with SIP (5).

Prerequisite

Have an up-to-date Debian 10 machine (without GUI) connected to the local network and to the internet, also with SSH access (see appendix A1 of the PDF report for the detailed implementation of this prerequisite).

Consider in this section the following information from this machine:

Installation of Asterisk

Note: the libraries allowing the management of telephone interface cards from Digium (the company maintaining Asterisk) and the management of protocols used in ISDN networks will not be installed.

1. Connect via SSH to the asterisk machine.

# ssh login@vm_ip_address -p 22
ssh asterisktz@192.168.1.80 -p 22

2. Install the packages allowing the compilation of Asterisk and the prerequisites.

sudo apt update && sudo apt install linux-headers-$(uname -r) build-essential autoconf libglib2.0-dev libtool net-tools

3. Reboot the Debian machine and connect again via SSH.

sudo reboot

# SSH reconnection
ssh asterisktz@192.168.1.80 -p 22

4. Go to the directory /usr/src and download Asterisk 18 at this address: https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz.

cd /usr/src
sudo wget https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz

5. Decompress the archive.

sudo tar -zxvf asterisk-18-current.tar.gz

6. Run the install_prereq script installing the prerequisites. Note: To find out the minor version number, type ls. In this report, it is Asterisk 18.2.

cd asterisk-18.2.0/contrib/scripts/
sudo ./install_prereq install

During the installation of the prerequisites a window appears asking to select the telephone number code. Select 33 for France and validate with <Ok>. You can find an international list of area codes on this site: https://countrycode.org/.

(Figure 6 - Area code setting)

7. Checking the required dependencies.

cd ../..
sudo ./configure

If the operation is successful, you will get a message similar to this one with the Asterisk logo:

                .$$$$$$$$$$$$$$$=..      
              .$7$7..        .7$$7:.    
            .$7$7..           .7$$7:.
          .$$:.                 ,$7.7
        .$7.     7$$$$           .$$77
     ..$$.       $$$$$            .$$$7
    ..7$   .?.   $$$$$   .?.       7$$$.
   $.$.   .$$$7. $$$$7 .7$$$.      .$$$.
 .777.   .$$$$$$77$$$77$$$$$7.      $$$,
 $$$~      .7$$$$$$$$$$$$$7.       .$$$.
.$$7          .7$$$$$$$7:          ?$$$.
$$$          ?7$$$$$$$$$$I        .$$$7
$$$       .7$$$$$$$$$$$$$$$$      :$$$.
$$$       $$$$$$7$$$$$$$$$$$$    .$$$.
$$$        $$$   7$$$7  .$$$    .$$$.
$$$$             $$$$7         .$$$.
7$$$7            7$$$$        7$$$
 $$$$$                        $$$
  $$$$7.                       $$  (TM)
   $$$$$$$.           .7$$$$$$  $$
     $$$$$$$$$$$$7$$$$$$$$$.$$$$$$
       $$$$$$$$$$$$$$$$.

configure: Package configured for:
configure: OS type : linux-gnu
configure: Host CPU : x86_64
configure: build-cpu:vendor:os: x86_64 : pc : linux-gnu :
configure: host-cpu:vendor:os: x86_64 : pc : linux-gnu :

If it went wrong, here is a link to the documentation:
https://wiki.asterisk.org/wiki/display/AST/Checking+Asterisk+Requirements.

8. Selecting the options to be installed for Asterisk. Ensure that the terminal is at least 80x27 in size.

sudo make menuselect

A window appears allowing you to choose the options.

Go to the Core Sound Package section to install the French sounds (or another language if you prefer):

• Deselect the CORE-SOUNDS-EN-GSM. These are the English sounds with the GSM audio codec.
• Select the CORE-SOUNDS-FR-ULAW and CORE-SOUNDS-FR-ALAWpackages.These are the French sounds with ULAW and ALAW codecs (supported by the Alcatel IP Touch 4018EE (6)). These codecs are defined by the ITU G.711 standard: https://www.itu.int/rec/T-REC-G.711-198811-I/en. The ALAW codec is used in Europe and Africa. The ULAW codec in North America and Japan (7).

(Figure 7 - Selection of Asterisk sound modules)

Then in the Extra Sound Packages section, select the EXTRA-SOUNDS-FR-ULAW and EXTRA-SOUNDS-FR-ALAW packages.

(Figure 8 - Selection of Asterisk extra sound modules)

Confirm by pressing the F12 key.

9. Compile the program with the previously chosen options and install Asterisk.

sudo make

The compilation takes time depending on the power of the machine. Once successful...

+--------- Asterisk Build Complete ---------+
+ Asterisk has successfully been built, and +
+ can be installed by running:              +
+                                           +
+                make install               +
+-------------------------------------------+
+--------- Asterisk Build Complete ---------+

...the installation of Asterisk can begin.

sudo make install

The installation ends with this message:

 +---- Asterisk Installation Complete -------+
 +                                           +
 +    YOU MUST READ THE SECURITY DOCUMENT    +
 +                                           +
 + Asterisk has successfully been installed. +
 + If you would like to install the sample   +
 + configuration files (overwriting any      +
 + existing config files), run:              +
 +                                           +
 + For generic reference documentation:      +
 +    make samples                           +
 +                                           +
 + For a sample basic PBX:                   +
 +    make basic-pbx                         +
 +                                           +
 +                                           +
 +-----------------  or ---------------------+
 +                                           +
 + You can go ahead and install the asterisk +
 + program documentation now or later run:   +
 +                                           +
 +               make progdocs               +
 +                                           +
 + **Note** This requires that you have      +
 + doxygen installed on your local system    +
 +-------------------------------------------+

10. Create the sample configuration files in the /etc/asterisk folder.

sudo make samples

11. Install the start-up scripts.

sudo make config
sudo ldconfig

12. Setting up the automatic start of the Asterisk service when the machine is launched.

cd
sudo groupadd asterisk
sudo useradd -r -d /var/lib/asterisk -g asterisk asterisk
sudo usermod -aG audio,dialout asterisk
sudo chown -R asterisk.asterisk /etc/asterisk
sudo chown -R asterisk.asterisk /var/{lib,log,spool}/asterisk
sudo chown -R asterisk.asterisk /usr/lib/asterisk

sudo nano /etc/default/asterisk

Uncomment the lines AST_USER="asterisk" and AST_GROUP="asterisk" (remove the # before each line). Save with Ctrl + O. Exit with Ctrl + X.

(Figure 9 - Setting asterisk as the default user of the Asterisk service)

sudo nano /etc/asterisk/asterisk.conf

Uncomment the lines (remove the semicolon ; before each line). Save with Ctrl + O. Exit with Ctrl + X.

runuser = asterisk ; The user to run as.
rungroup = asterisk ; The group to run as.

(Figure 10 - Setting asterisk as the default user of the Asterisk service)

13. Start the Asterisk service

sudo systemctl start asterisk

You can now check its current status with the following command:

sudo systemctl status asterisk

(Figure 11 - Launch of the Asterisk service with errors)

At this stage, if there are these two red lines, they can be corrected in this way (fix from https://www.clearhat.org/blog/post/a-fix-for-apt-install-asterisk-on-ubuntu-18-04):

sudo systemctl stop asterisk

sudo sed -i 's";\[radius\]"\[radius\]"g' /etc/asterisk/cdr.conf

sudo sed -i 's";radiuscfg => /usr/local/etc/radiusclient-ng/radiusclient.conf"radiuscfg => /etc/radcli/radiusclient.conf"g' /etc/asterisk/cdr.conf

sudo sed -i 's";radiuscfg => /usr/local/etc/radiusclient-ng/radiusclient.conf"radiuscfg => /etc/radcli/radiusclient.conf"g' /etc/asterisk/cel.conf

sudo systemctl start asterisk

(Figure 12 - Launch of the Asterisk service without errors)

14. Automatic start of the Asterisk service.

sudo /lib/systemd/systemd-sysv-install enable asterisk

Asterisk is operational. Let's move on to user creation and SIP configuration!

SIP configuration and user creation

So we have an operational server, we must now configure SIP and create users. We will create 3 users with the following characteristics:

There are two configuration files to edit:pjsip.conf and extensions.conf. The first one is used to create accounts, configure the operation of SIP (UDP/TCP), the authentication systems... and the second to define the behaviour of the system, more precisely the dialling plan (similar to routing if we were talking about IP packets). This "routing" is done according to telephone numbers (identifiers). These files are present in /etc/asterisk and in the asterisk_sip directory of the Git repository.

SIP configuration - pjsip.conf

1. Rename the pjsip.conf configuration file to pjsip_original.conf.

sudo mv /etc/asterisk/pjsip.conf /etc/asterisk/pjsip_original.conf

2. Create a pjsip.conf file...

sudo nano /etc/asterisk/pjsip.conf

...and write the following content:

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
 
; Basic templates, they will be copied for each user
 
[endpoint_basic](!)
type=endpoint         ; endpoint (phone/rpi/pc...)
context=plan-num      ; uses the dial plan defined in extensions.conf
disallow=all          ; disabling all audio codecs
allow=ulaw            ; except the ULAW codec
allow=alaw            ; and the ALAW codec
language=fr
 
[authentication](!)
type=auth             ; type of section: authentication
auth_type=userpass    ; password authentication
 
[aor_template](!)
type=aor              ; find out where the endpoint can be contacted 
max_contacts=1
 
; Definitions of user accounts associated with equipment
 
[alcatel](endpoint_basic)
auth=alcatel
aors=alcatel
callerid="Alcaltel IP Touch" <5001> ; to have the name of the caller displayed
[alcatel](authentication)
password=11111111
username=alcatel
[alcatel](aor_template)
 
[rpi](endpoint_basic)
auth=rpi
aors=rpi
callerid="Raspberry Pi" <5002>
[rpi](authentication)
password=22222222
username=rpi
[rpi](aor_template)
 
[guillaume](endpoint_basic)
auth=guillaume
aors=guillaume
callerid="Guillaume Nibert" <5003>
[guillaume](authentication)
password=33333333
username=guillaume
[guillaume](aor_template)

The creation of the accounts and the configuration of SIP is complete. Let's move on to the dial plan.

Dial plan - extensions.conf

1. Rename the extensions.conf configuration file to extensions_original.conf.

sudo mv /etc/asterisk/extensions.conf /etc/asterisk/extensions_original.conf

2. Create an extensions.conf file...

sudo nano /etc/asterisk/extensions.conf

...and write the following content:

[plan-num]
exten => 5001,1,Answer(500)
exten => 5001,2,Dial(PJSIP/alcatel,25)
exten => 5001,3,Hangup()
 
exten => 5002,1,Answer(500)
exten => 5002,2,Dial(PJSIP/rpi,25)
exten => 5002,3,Hangup()
 
exten => 5003,1,Answer(500)
exten => 5003,2,Dial(PJSIP/guillaume,25)
exten => 5003,3,Hangup()
 
; 1, 2 and 3 correspond to the priorities of the Answer(),
; Dial() and Hangup() application calls. 1 being the highest priority.
; We can also write 1,n,n where the first n corresponds to 2 and the second
; to 3.

"The Answer() application takes a delay (in milliseconds) as its first parameter. Adding a short delay is often useful to ensure that the endpoint has time to start processing the audio before starting the communication via the Dial() application. Otherwise, you may not hear the very beginning" (8). Hangup() as the name suggests hangs up the current call.

The dial plan is complete. The SIP accounts have been created. We can now proceed to the configuration of a SIP client on the Raspberry Pi.

3. Installation and configuration of a SIP client on the Raspberry Pi

The installation and configuration of a SIP client on the Raspberry Pi is necessary to communicate with VoIP. This client will connect to the Asterisk server and depending on the number the client is calling, the server will use the dial plan defined in extensions.conf to contact the right endpoint (Alcatel phone for example).

Technology choices

• OS: Raspberry Pi OS Buster (arm64), the 64-bit version is only very recent but seems to be promising. Indeed, it is more powerful than the 32 bits version (armhf) (9). This is a significant advantage, especially on a Raspberry Pi 3B+ which will be used in desktop mode. Any improvement in performance is worthwhile. Furthermore, Raspberry Pi OS is a system officially maintained by the Raspberry Pi Foundation.
• SIP client: linphonec, the command line version of Linphone. It is stable and works perfectly on Raspberry Pi OS. It is open source. Unfortunately, the popular client Jami (formerly GNU Ring), developed by Savoir-faire Linux does not seem to work well with Raspberry Pi OS.

Prerequisites

Having an up-to-date Raspberry Pi 3B+ running Raspberry Pi OS Buster (64-bit) connected to the local network and the internet, also with SSH access (see appendix A2 of the PDF report for the detailed implementation of this requirement). The Asterisk server must be running.

Consider in this section the following information from this machine:

Installation and configuration of the Linphone SIP client

1. Start the Raspberry Pi then launch a terminal and install Linphone.

sudo apt install linphone -y

2. Once the installation is complete, register the associated rpi user on the server.

linphonec

#linphonec> register sip:login@domain domain <password>

linphonec> register sip:rpi@192.168.1.80 192.168.1.80 22222222

The rpi client is connected to the Asterisk server. It can therefore contact the Alcatel telephone and vice versa.

Client-side testing

If the registration on the server was successful, the previously executed command returns this:

linphonec> Refreshing on sip:rpi@192.168.1.80…
linphonec> Registration on sip:192.168.1.80 successful.
linphonec>

Server-side testing

On a server console, type the command sudo asterisk -rvvv.

Once entered, type the command pjsip show endpoints. If the Raspberry Pi's SIP client is connected then the console will return this:

asterisktz*CLI> pjsip show endpoints

 Endpoint:      
    I/OAuth:  
        Aor:    
      Contact:     
  Transport:          
   Identify:  
        Match:  
    Channel:      
        Exten:   CLCID: 
===============================================================================

 Endpoint:  alcatel/5001                                 Unavailable   0 of inf
     InAuth:  alcatel/alcatel
        Aor:  alcatel                                        1

 Endpoint:  guillaume/5003                               Unavailable   0 of inf
     InAuth:  guillaume/guillaume
        Aor:  guillaume                                      1

 Endpoint:  rpi/5002                                     Not in use    0 of inf
     InAuth:  rpi/rpi
        Aor:  rpi                                            1
      Contact:  rpi/sip:rpi@192.168.1.82;transport=udp  cec2f9dd2f NonQual  nan


Objects found: 3

It is clear that the client has an IP address of 192.168.1.82 and is connected. The information "Not in use" indicates that there is no call in progress.

The Alcatel phone is still not connected, it is time to integrate it into the system.

4. IP phone configuration

The Alcatel IP Touch 4018 EE is a telephone with two operating modes:

1. NOE (New Office Environment) mode: proprietary communication protocol developed by Alcatel/Lucent;
2. SIP mode.

First, it must be configured in SIP mode. Then, it has the particularity to configure itself automatically by downloading its configuration files and its firmware files on a TFTP, HTTP or HTTPS server at startup.

Among the three protocols, the most secure is HTTPS. We tried to implement it with 2048 bit RSA certificates and cipher suites compatible with older equipment. Unfortunately, being in a local network, the Alcatel phone seems not to accept self-signed certificates (according to this forum, it seems that Alcatel IP Touch phones only trust the certification authority issued by the Alcatel-Lucent company. The root certificate from Alcatel-Lucent seems to be inside the phone:
https://www.alcatelunleashed.com/viewtopic.php?t=28822. We do not have the means to get a certificate from the Alcatel-Lucent certification authority). Given this problem, there are only two choices: TFTP or HTTP. These application protocols are not very secure: no authentication, transfer of data in clear text over the network... The choice is therefore made at the level of the transport protocols: TFTP necessarily uses UDP, a non-connected mode, whereas HTTP can be configured to use TCP, a connected mode, which includes error detection and correction. The programs transmitted are firmware, if data is corrupted due to an error, it could brick the phone. So let's use the HTTP protocol associated with the TCP protocol. We will therefore set up a lightweight NGINX HTTP server, whose sole purpose is to provide firmware and configuration files to the Alcatel IP Touch phone. It will be accessible at the address 192.168.1.80 on port 80.

Of course, if possible, for production use, we recommend using HTTPS.

(Figure 13 - Retrieving Alcatel phone configuration files via HTTP)

Configuration information

Configuring the phone in SIP mode - on the device

Connect the phone to the PoE injector and follow the instructions in the installation manual below (screenshots of the procedure).

(Figure 14 - Alcatel-Lucent, 3. SIP stand-alone mode In: IP Touch 4008/4018 Extended Edition - SIP Phone Installation Guide - 8AL90824AAAA ed02, p.4-5, August 2010, Available at: https://www.cluster2.hostgator.co.in/files/writeable/uploads/hostgator136107/file/iptouchsipphoneinstallationguide-ed02.pdf)

Once in SIP mode, the phone's IP address must be configured to match the one defined in the router (192.168.1.81), and it must also be provided with the IP address of the HTTP server and its port.

1. Connect the phone to the PoE injector.
2. At phase 2/5 network setup, press " i " then " #" until the " MAC address " menu appears.
3. If there is already a password configured, enter 000000.
4. Scroll down to IP Parameters, then click OK.
5. Scroll down, then select IP mode: Static.
6. Scroll down, then enter the IP address: IP@: 192.168.1.81
7. Scroll down, then enter the subnet mask: Subnet: 255.255.255.0
8. Scroll down, then enter the router IP address: Router: 192.168.1.254
9. Scroll down, then select from DL Scheme: HTTP.
10. Scroll down, then select Use Defaultport.
11. Scroll down, then select the HTTP server address: DL Addr: 192.168.1.80
12. Scroll down, then select the HTTP server port: DL Port: 80
13. Scroll down, do not select a VLAN.
14. Scroll down to save save, then click OK.

The configuration on the device is complete, however for now it will keep restarting in a loop as the HTTP server does not yet exist, nor do the associated configuration files and phone firmwares.

HTTP server installation

So let's go back to our Debian virtual machine and install an HTTP server.

1. Connect via SSH to the asterisktz machine.

ssh asterisktz@192.168.1.80 -p 22

2. Install nginx HTTP server..

sudo apt install nginx
sudo systemctl status nginx

If the output shows active then it is working, otherwise you need to run it sudo systemctl start nginx. Normally it is configured to start automatically at startup, if this is not the case then run the following command:

sudo /lib/systemd/systemd-sysv-install enable nginx

The HTTP server is ready, all that remains is to transfer the firmware, the information on the SIP account dedicated to the Alcatel phone and the IP address of the Asterisk server.

Transfer of configuration files and firmware to the Alcatel phone

In practise, there are 4 files to transfer to the HTTP server (10):

• sipconfig.txt: global configuration file, contains the settings to be applied to all Alcatel IP Touch 4018EE phones connected to the network.
• sipconfig-MacAddress.txt: configuration file specific to a single Alcatel IP Touch 4018EE phone. This is the MAC address of the phone in question which must be written in lower case.
• noesip4018: proprietary firmware containing the SIP protocol application for the Alcatel IP Touch 4018EE.
• datsip4018: resources containing ringtones and different melodies.

As the Alcatel-Lucent website does not necessarily provide the appropriate documentation for the structure of the sipconfig.txt files, we have based ourselves on an example configuration file created by Florian Duraffourg, a graduate of Télécom SudParis: https://github.com/fduraffourg/utils/blob/master/iptouch/sipconfig-reynoud.txt.

Concerning the firmwares, also difficult to find on the official Alcatel-Lucent website, we found them on the Alcatel Unleashed forum through fbird's post on March 21st 2016: https://www.alcatelunleashed.com/viewtopic.php?p=95015#p95015.

The most important file is sipconfig-MacAddress.txt whose important content in bold and green is as follows (we have voluntarily removed the non-essential parts, you will find the complete file in the project repository available in the following paragraph):
Note: we added comments in this report, to avoid mistakes, you should take the one located in the repository.

[...]


[dns]

###########################################################################
## The primary DNS IP address HAS TO BE FILLED
## If no DNS, use the SIP proxy address instead
###########################################################################

   dns_addr=192.168.1.254  #  DNS of the Freebox (or another router)
   dns2_addr=
   hostname=192.168.1.254

[sip]

###########################################################################
## Domain name : IP address, FQDN or domain name (see the SIP proxy config)
###########################################################################

   domain_name=192.168.1.80  # Asterisk server's domain

###########################################################################
## Primary SIP proxy and SIP registrar settings
##
## Proxy address : IP address, FQDN or domain name
## Registrar address : IP address, FQDN or domain name (usually, the proxy)
## SIP proxy UDP port : usually 5060
## SIP registrar UDP port : by default 5060
###########################################################################

   proxy_addr=192.168.1.80
   proxy_port=5060
   registrar_addr=192.168.1.80
   registrar_port=5060
   outbound_proxy_addr=
   outbound_proxy_port=

###########################################################################
## Redundancy settings
##
## Proxy address : IP address, FQDN or domain name
## Registrar address : IP address, FQDN or domain name (usually, the proxy)
## SIP proxy UDP port : usually 5060
## SIP registrar UDP port : by default 5060
## sip_transport_mode_survi : Transport mode in PCS mode
##          0 = UDP or TCP
##          1 = UDP
##          2 = TCP
###########################################################################

   proxy2_addr=192.168.1.80
   proxy2_port=5060
   registrar2_addr=192.168.1.80
   registrar2_port=5060
   outbound_proxy2_addr=
   outbound_proxy2_port=
   pcs_addr=192.168.1.80
   pcs_port=5060
   sip_transport_mode_survi=0 # in our case, it will be UDP
   option_timer=120

###########################################################################
## Global SIP parameters
## Transport mode : 0 = UDP or TCP
##                  1 = UDP
##                  2 = TCP
## local_rtp_port : RFC3605 is not supported in this release, so
##                  only default value can be used
## PRACK type : 0 = PRACK supported
##              1 = PRACK required
##              2 = PRACK disabled
## Codec settings : 0 = G711 (PCMU)
##                  4 = G723.1
##                  8 = G711 (PCMA)
##                 18 = G729A
###########################################################################

   register_expire=3600
   register_retry=300
   local_sip_port=
   sip_transport_mode=0 # dans notre cas, ce sera de l’UDP
   local_rtp_port=42000
   local_rtcp_port=42001
   prack_type=0
   preferred_vocoder=8,0,4,18

###########################################################################
## SIP authentication.
##
## Realm : If no authentication, leave empty
## Authentication name : HAS TO BE FILLED
##                       If no authentication, PUT A VALUE LIKE none
## Authentication password : If no authentication, leave empty
###########################################################################

   authentication_realm=192.168.1.80  # authentication is done on the Asterisk server
   authentication_name=alcatel        # username and password
   authentication_password=11111111
   user_name=alcatel
   display_name=Alcatel IP Touch      # display name when calling

[...]

[sntp]

###########################################################################
## SNTP server settings (can be OXE or an external server)
##
## Timezone construction : UT::60:032802:103103  (Paris - 2021)
##          GMT delta : 60 = + sixty minutes from GMT time
##          Daylight saving start (mmddhh) : 032902 = 28 March 2am
##          Daylight saving end (mmddhh) : 103103 = 31 October 3am
## The daylight saving settings HAVE to be changed each year.
###########################################################################

   sntp_addr=192.168.1.254       # To synchronise the phone's time
   timezone=UT::60:032802:103103 # with the Freebox's built-in NTP server
                                 # The time zone is to be changed every year,
                                 # it is set according to the GMT zone and
                                 # manages summer and winter time.

[...]

[init]

###########################################################################
## For IP Touch with SIP binary in 1.xx, 2.00.10 and 2.00.20, equal
## or greater than 2.00.81
##     mode 0 = SIP
##     mode 1 = NOE
##
## For IP Touch with SIP binary 2.00.30 to 2.00.80
##     mode 0 = NOE
##     mode 1 = SIP
###########################################################################

   application_mode=0     # SIP mode (depending on firmware version)

[audio]

###########################################################################
## Tone country : 0 = English
##                1 = French
##                2 = German
##                3 = Italian
##                4 = Spanish
##                5 = Dutch
##                6 = Portuguese
## DTMF type : 0 = RFC2833
##             1 = In-band
##             2 = SIP INFO
## DTMF level / RLR handset / SLR handset / Sidetone handset :
## 0 = 0db, 1 = +3db, 2 = +6db, 3 = -3db, 4 = -6db
## VAD / DTMF feedback / Hearing Aid :
##       0 = VAD not used
##       1 = VAD used
###########################################################################

   tone_country=1    # to be set according to the standards used in France
   dtmf_type=1
   dtmf_level=0
   dtmf_avt_payload_type=96
   vad=0
   dtmf_feedback_enable=0
   rlr_handset=0
   slr_handset=0
   sidetone_handset=2
   hearing_aid_enable=0

[appl]

###########################################################################
## Password to access the administrator menu on the phone (digits only)
## Power priority : 1 = critical
##                  2 = high
##                  3 = low
## Time format : 0 = 24 hours format
##               1 = AM / PM
## Speed dial numbers (first and last name, URI)
###########################################################################

   admin_password=000000   # admin password when pressing i then #.
   bluetooth_parameters=blue
   supported_language=0
   remote_forward_code=
   remote_forward_deactive_code=
   power_priority=
   asset_id=
   time_format=0
   speed_dial_1_first_name=
   speed_dial_1_last_name=
   speed_dial_1_uri=
   speed_dial_2_first_name=
   speed_dial_2_last_name=
   speed_dial_2_uri=
   speed_dial_3_first_name=
   speed_dial_3_last_name=
   speed_dial_3_uri=
   speed_dial_4_first_name=
   speed_dial_4_last_name=
   speed_dial_4_uri=
[...]
    

1. Download all these 2 firmware files (noesip4018 and datsip4018, found on the Alcatel Unleashed forum) and the 2 configuration files (sipconfig.txt and sipconfig-MacAddress.txt), available in the project repository: https://github.com/guillaumenibert/VoIP-Asterisk-WebRTC-SIP/tree/main/iptouch4018ee.
2. Place them in the /var/www/html of the Debian virtual machine.
3. Connect the phone to the PoE injector to turn it on. The Alcatel IP Touch will fetch the latest firmware from the HTTP NGINX server, the configurations and will be connected to the Asterisk server.

Client-side testing

If the phone displays the date and time and does not restart in a loop, it is connected to the Asterisk server.

Server-side testing

On a server console, type the command sudo asterisk -rvvv.

Once entered, type the command pjsip show endpoints. If the Raspberry Pi's SIP client is connected then the console will return this:

asterisktz*CLI> pjsip show endpoints

 Endpoint:      
    I/OAuth:  
        Aor:    
      Contact:     
  Transport:          
   Identify:  
        Match:  
    Channel:      
        Exten:   CLCID: 
===============================================================================

 Endpoint:  alcatel/5001                                 Not in use    0 of inf
     InAuth:  alcatel/alcatel
        Aor:  alcatel                                        1
      Contact:  alcatel/sip:alcatel@192.168.1.81       8c02c0558c NonQual  nan

 Endpoint:  guillaume/5003                               Unavailable   0 of inf
     InAuth:  guillaume/guillaume
        Aor:  guillaume                                      1

 Endpoint:  rpi/5002                                     Not in use    0 of inf
     InAuth:  rpi/rpi
        Aor:  rpi                                            1
      Contact:  rpi/sip:rpi@192.168.1.82;transport=udp cec2f9dd2f NonQual  nan


Objects found: 3

The client has an IP address of 192.168.1.81 and is connected. The information "Not in use" indicates that there is no call in progress.

5. Communication tests

All endpoints are connected to the Asterisk server. It is therefore possible to call from the Raspberry Pi to the Alcatel or from the Alcatel to the Raspberry Pi.

In order to carry out the communication, you must first have launched the Asterisk server, the TFTP server, turned on all the peripherals and connected an audio output on the Raspberry Pi (audio jack, Bluetooth or HDMI) to be able to listen to the audio stream.

Raspberry Pi to Alcatel IP Touch

1. Launch a terminal and run linphonec.

linphonec

2. Call the Alcatel IP Touch phone, it has the number 5001 (see part 2 - SIP configuration and user creation).

linphonec> call 5001

Output (with comments):

# Error message, not important, video is disabled, we are only doing VoIP.
2021-02-09 13:30:36:367 ortp-error-LinphoneCore has video disabled for both capture
and display, but video policy is to start the call with video. This is a possible
mis-use of the API. In this case, video is disabled in default LinphoneCallParams

# Linking to the Alcatel phone
Establishing call id to sip:5001@192.168.1.80, assigned id 1
# The Alcatel phone has been found, it is contacted, it rings on the Alcatel
# side.
Contacting sip:5001@192.168.1.80
linphonec> Call 1 to sip:5001@192.168.1.80 in progress.
linphonec> Call 1 with sip:5001@192.168.1.80 connected.
# We picked up the Alcatel phone.
Call answered by sip:5001@192.168.1.80
# Communication is in progress, audio is playing, settings are adjusted.
linphonec> 2021-02-09 13:30:36:563 ortp-error-no such method on filter MSPulseWrite,
fid=16394 method index=2
Media streams established with sip:5001@192.168.1.80 for call 1 (audio).
Call is updated by remote.
linphonec> 2021-02-09 13:30:40:761 ortp-error-no such method on filter MSPulseWrite,
fid=16394 method index=2
Call parameters were successfully modified.
linphonec> Media streams established with sip:5001@192.168.1.80 for call 1 (audio).
Call is updated by remote.
linphonec> Call parameters were successfully modified.
linphonec> Media streams established with sip:5001@192.168.1.80 for call 1 (audio).
# The call has just ended, someone has hung up one of the devices.
Call terminated.
linphonec> Call 1 with sip:5001@192.168.1.80 ended (No error).

When the call is initiated, the Alcatel phone screen displays the following:

(Figure 15 - Call from Raspberry Pi to Alcatel IP Touch 4018 EE)

The communication therefore works in one direction. Let's see what happens if the Alcatel calls the Raspberry Pi.

Alcatel IP Touch to Raspberry Pi

1. From the phone, call the Raspberry Pi's number 5002 (see part 2 - SIP configuration and user creation).

(Figure 16 - Call from Alcatel IP Touch 4018 EE to Raspberry Pi)

2. From the Raspberry Pi terminal, make sure that linphonec is active. When the Alcatel launches its call, it is received in the terminal:

Output:

linphonec> Receiving new incoming call from "Alcaltel IP Touch" <sip:5001@192.168.1.80>, assigned id 3

To answer it, just type answer and the call ID:

answer 3

The communication is launched and works in the same way.

Output:

linphonec> Receiving new incoming call from "Alcaltel IP Touch" , assigned id 3
answer 3
Connected.
linphonec> Call 3 with "Alcaltel IP Touch"  connected.
2021-02-09 13:46:28:345 ortp-error-no such method on filter MSPulseWrite, fid=16394
method index=2
Media streams established with "Alcaltel IP Touch"  for call 3 (audio).
linphonec> Call is updated by remote.
linphonec> 2021-02-09 13:46:28:424 ortp-error-no such method on filter MSPulseWrite,
fid=16394
method index=2
Call parameters were successfully modified.
linphonec> Media streams established with "Alcaltel IP Touch"  for call 3 (audio).
Call is updated by remote.
linphonec> Call parameters were successfully modified.
linphonec> Media streams established with "Alcaltel IP Touch"  for call 3 (audio).
Call terminated.
linphonec> Call 3 with "Alcaltel IP Touch"  ended (No error).

The communications therefore work in both directions. The objective of the next part is to create a graphical interface in JavaScript on the Raspberry Pi side, more user-friendly than the linphonec command line client.

6. JavaScript SIP client using WebRTC

The establishment of the communication (SIP) as well as the communication itself (RTP) works correctly. The development of a SIP client program in JavaScript will require modifications to our environment. Indeed, we are going to make a call from a web browser supporting JavaScript to another device (having a JavaScript SIP client or not). Before starting the implementation. It is necessary to understand what the WebRTC, WebSocket APIs are.

WebRTC & WebSocket

The WebRTC (Web Real-Time Communication) API is a software interface whose purpose is to link two devices so that they can communicate directly. This connection requires opening a communication channel between a client and a server: the technology that allows this is the WebSocket API.

In concrete terms, establishing a connection works in a similar way to SIP. Below is an explanatory diagram from Wikipedia.

(Figure 17 - Establishing a connection between two clients)
Attribution: adapted from the original work of Feyd-Aran, CC BY-SA 3.0, via Wikimedia Commons

• 1: A asks the server for a connection with B.
• 2: The server relays the request from A to B.
• 3: If B accepts, it sends a connection request to A.
• 4: The server relays the request to A.
• 5 and 6: Bidirectional PeerConnection is established.

— Wikipédia

The PeerConnection corresponds in our case to the RTP flow between the two clients. Once the communication is established, as with SIP, the communication between the two clients is direct and the media flows do not pass through the application web server.

Let's imagine now:

• Client A: Raspberry Pi, with Mozilla Firefox web browser supporting WebRTC.
• Client B: Alcatel IP Touch 4018 EE, not supporting WebRTC.

How can the two endpoints be made to communicate?

The Alcatel phone cannot support WebRTC, it is proprietary hardware, the source code is closed.

On the Raspberry Pi side, however, it is possible to use WebRTC and SIP by encapsulating the SIP protocol in a WebSocket. This is defined in RFC 7118 and requires a server that can handle WebRTC/SIP for client A and only SIP for client B. The Asterisk server supports WebRTC with SIP. Therefore, modifications are required to make the server capable of supporting WebRTC.

(Figure 18 - Raspberry Pi calls Alcatel IP Touch from a WebRTC client)

At the Raspberry Pi client level, web browsers such as Mozilla Firefox, Safari or those based on Chromium natively implement the WebRTC API. In this project, Mozilla Firefox will be used as a client using WebRTC.

Configuring the Asterisk server to support the WebRTC API

In order to improve the security between the WebRTC client and the Asterisk server, a secure web socket (WSS) via TLS will be set up. We will therefore first generate a self-signed certificate.

Generating a self-signed SSL/TLS certificate

In order to improve security and modernise, we adopt a certificate generated with ECDSA algorithms, which are much more efficient than the classic RSA algorithms (11). We will use the ECDSA P-521 algorithm, recommended by the ANSSI (French National Agency for the Security of Information Systems) (12) and compatible with Mozilla Firefox.

To perform this operation, you must first have started the Debian virtual machine and have the OpenSSL tool.

1. Connect via SSH to the asterisktz machine.

# ssh login@vm_ip_address -p 22
ssh asterisktz@192.168.1.80 -p 22

2. Create the folders in which the Certificate Authority's certificate, called the root certificate (ca), the certificate associated with the IP 192.168.1.80 (certs) and the certificate signing request file to the authority (csr) are stored.

mkdir ca && mkdir certs && mkdir csr

3. Creation of certificates.

Create the private key of the root certificate (certification authority)

openssl ecparam -genkey -name secp521r1 -out ca/TZVoIP-Root-CA.key

Generate the root certificate from its private key

openssl req -x509 -new -nodes -key ca/TZVoIP-Root-CA.key -sha384 -days 3650 -utf8 -out ca/TZVoIP-Root-CA.crt

Information to be filled in:

You are about to be asked to enter information that will be incorporated
into your certificate request.
What you are about to enter is what is called a Distinguished Name or a DN.
There are quite a few fields but you can leave some blank
For some fields there will be a default value,
If you enter '.', the field will be left blank.
-----
Country Name (2 letter code) [AU]:FR
State or Province Name (full name) [Some-State]:Hauts-de-France
Locality Name (eg, city) []:Compiègne
Organization Name (eg, company) [Internet Widgits Pty Ltd]:Université de Technologie de Compiègne
Organizational Unit Name (eg, section) []:TZ VoIP
Common Name (e.g. server FQDN or YOUR name) []:TZ VoIP Root
Email Address []:guillaume.nibert@etu.utc.fr

Generate the private key of the IP address certificate and its signature request file.

openssl req -new -sha384 -nodes -utf8 -out csr/asterisktz.csr -newkey ec:<(openssl ecparam -name secp521r1) -keyout certs/asterisktz.key

Information to be filled in:

Generating an EC private key
writing new private key to 'certs/asterisktz.key'
-----
You are about to be asked to enter information that will be incorporated
into your certificate request.
What you are about to enter is what is called a Distinguished Name or a DN.
There are quite a few fields but you can leave some blank
For some fields there will be a default value,
If you enter '.', the field will be left blank.
-----
Country Name (2 letter code) [AU]:FR
State or Province Name (full name) [Some-State]:Hauts-de-France
Locality Name (eg, city) []:Compiègne
Organization Name (eg, company) [Internet Widgits Pty Ltd]:Université de Technologie de Compiègne
Organizational Unit Name (eg, section) []:TZ VoIP
Common Name (e.g. server FQDN or YOUR name) []:192.168.1.80
Email Address []:guillaume.nibert@etu.utc.fr

Please enter the following 'extra' attributes
to be sent with your certificate request
A challenge password []:
An optional company name []:

Create the file containing the parameters of the certificate to be created.

nano csr/openssl-v3.cnf

csr/openssl-v3.cnf

authorityKeyIdentifier=keyid,issuer
basicConstraints=CA:FALSE
keyUsage = digitalSignature, nonRepudiation, keyEncipherment, dataEncipherment
subjectAltName = @alt_names

[alt_names]
IP.1 = 192.168.1.80

Generate the certificate and signature with the certification authority.

openssl x509 -req -in csr/asterisktz.csr -CA ca/TZVoIP-Root-CA.crt -CAkey ca/TZVoIP-Root-CA.key -CAcreateserial -out certs/asterisktz.crt -days 365 -sha384 -extfile csr/openssl-v3.cnf

Produce the full-chain certificate.

cat certs/asterisktz.crt certs/asterisktz.key > certs/asterisktz.pem

Change the permissions.

chmod a+r certs/asterisktz.pem

The self-signed certificate has been created. For more details about the certification process for a non-self-signed certificate, see: https://letsencrypt.org/how-it-works/. Let's move on to enabling the HTTP server built into Asterisk.

Enabling the Asterisk HTTP server

On the same machine:

1. Edit the configuration file for Asterisk's built-in HTTP server.

sudo nano /etc/asterisk/http.conf

2. Replace the contents with:

/etc/asterisk/http.conf

[general]
enabled=no
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/home/asterisktz/certs/asterisktz.crt
tlsprivatekey=/home/asterisktz/certs/asterisktz.key
enablestatic=no
sessionlimit=1000

Here only encrypted connections (tlsenable=yes) are allowed on port 8089. Unencrypted connections would be enabled to yes on port 8088.

Restart the Asterisk service to enable the built-in HTTP server.

sudo systemctl restart asterisk

To check that the HTTP server is enabled, simply type the command sudo asterisk -rvvv, then the command http show status. It should return this:

asterisktz*CLI> http show status
HTTP Server Status:
Prefix:
Server: Asterisk/18.2.0
Server Disabled

Enabled URI's:
/httpstatus => Asterisk HTTP General Status
/phoneprov/... => Asterisk HTTP Phone Provisioning Tool
/metrics/... => Prometheus Metrics URI
/ari/... => Asterisk RESTful API
/ws => Asterisk HTTP WebSocket

asterisktz*CLI>

The element we are interested in: the use of WebSocket for SIP (/ws) Let's move on to the configuration of pjsip.conf and extensions.conf to take into account both WebRTC and SIP. We have based ourselves on the Browser Phone project and have adapted the configuration files in question. The files are available in the asterisk_webrtc directory of the Git repository.

Editing pjsip.conf to support WebRTC

1. Edit the pjsip.conf configuration file.

sudo nano /etc/asterisk/pjsip.conf

2. Replace content with:

[global]
max_forwards=70
user_agent=AsteriskTZ
default_realm=192.168.1.80
keep_alive_interval=300

; == Transport
 
[udp_transport]
type=transport
protocol=udp
bind=0.0.0.0
tos=af42
cos=3
 
[wss_transport]
type=transport
protocol=wss
bind=0.0.0.0
 
[tcp_transport]
type=transport
protocol=tcp
bind=0.0.0.0
 
[tls_transport]
type=transport
protocol=tls
bind=0.0.0.0
cert_file=/home/asterisk/certs/asterisktz.crt
priv_key_file=/home/asterisk/certs/asterisktz.key
cipher=ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-ECDSA-CHACHA20-POLY1305:ECDHE-RSA-CHACHA20-POLY1305:DHE-RSA-AES128-GCM-SHA256:DHE-RSA-AES256-GCM-SHA384;
method=tlsv1_2
 
; == ACL
 
[acl]    ; Communications are only allowed in Class A, B and C local
type=acl ; networks.
deny=0.0.0.0/0.0.0.0
permit=10.0.0.0/255.0.0.0
permit=172.16.0.0/255.240.0.0
permit=192.168.0.0/255.255.0.0
 
; Templates
 
[single_aor](!)
max_contacts=1
qualify_frequency=120
remove_existing=yes
 
[userpass_auth](!)
auth_type=userpass
 
[basic_endpoint](!)
moh_suggest=default
context=from-extensions
inband_progress=no
rtp_timeout=120
message_context=textmessages
allow_subscribe=yes
subscribe_context=subscriptions
direct_media=yes
dtmf_mode=rfc4733
device_state_busy_at=1
disallow=all
 
[phone_endpoint](!)
allow=ulaw,alaw
 
[webrtc_endpoint](!)
transport=wss_transport
allow=ulaw,alaw
dtls_auto_generate_cert=yes
webrtc=yes
 
; Users
 
[alcatel](basic_endpoint,phone_endpoint)
type=endpoint
callerid="Alcatel IP Touch" <5001>
auth=alcatel
aors=alcatel
[alcatel](single_aor)
type=aor
[alcatel](userpass_auth)
type=auth
username=alcatel
password=11111111
 
[rpi](basic_endpoint,webrtc_endpoint)
type=endpoint
callerid="Raspberry Pi" <5002>
auth=rpi
aors=rpi
[rpi](single_aor)
type=aor
[rpi](userpass_auth)
type=auth
username=rpi
password=22222222
 
[guillaume](basic_endpoint,webrtc_endpoint)
type=endpoint
callerid="Guillaume Nibert" <5003>
auth=guillaume
aors=guillaume
[guillaume](single_aor)
type=aor
mailboxes=guillaume@default
[guillaume](userpass_auth)
type=auth
username=guillaume
password=33333333
    

For more details, please refer to the PJSIP documentation:
https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Sections+and+Relationships.

Editing extensions.conf to support WebRTC

3. Edit the configuration file extensions.conf.

sudo nano /etc/asterisk/extensions.conf

4. Replace the contents with:

[general]
static=yes
writeprotect=yes
priorityjumping=no
autofallthrough=no
 
[globals]
ATTENDED_TRANSFER_COMPLETE_SOUND=beep
 
[textmessages] ; Allows you to send text for WebRTC clients
exten => 5002,1,Gosub(send-text,s,1(rpi))
exten => 5003,1,Gosub(send-text,s,1(guillaume))
 
[subscriptions] ; Allows to know the status of an endpoint (in call or
exten => 5001,hint,PJSIP/alcatel ; available)
exten => 5002,hint,PJSIP/rpi
exten => 5003,hint,PJSIP/guillaume
 
[from-extensions]
; When you call 5000, you get music.
exten => 5000,1,Gosub(moh,s,1)
; Extensions
exten => 5001,1,Gosub(dial-extension,s,1,(alcatel))
exten => 5002,1,Gosub(dial-extension,s,1,(rpi))
exten => 5003,1,Gosub(dial-extension,s,1,(guillaume))
; If you have anything other than 5000, 5001, 5002 or 5003 then it is a
; wrong number, so hang up.
exten => _[+*0-9].,1,NoOp(You called: ${EXTEN})
exten => _[+*0-9].,n,Hangup(1)
 
exten => e,1,Hangup()
 
[moh] ; "function" for music (see 5000). Note: "function" is an abusive
; term, it is actually called "context".
exten => s,1,NoOp(Music On Hold)
exten => s,n,Ringing()
exten => s,n,Wait(2)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,MusicOnHold()
 
[dial-extension] ; "function" to call an endpoint.
exten => s,1,NoOp(Calling: ${ARG1})
exten => s,n,Set(JITTERBUFFER(adaptive)=default)
exten => s,n,Dial(PJSIP/${ARG1},30)
exten => s,n,Hangup()
 
exten => e,1,Hangup()
 
[send-text] ; "function" to send text.
exten => s,1,NoOp(Sending Text To: ${ARG1} From: ${MESSAGE(from)})
exten => s,n,Set(PEER=${CUT(CUT(CUT(MESSAGE(from),@,1),<,2),:,)})
exten => s,n,Set(FROM=${SHELL(asterisk -rx 'pjsip show endpoint ${PEER}' | grep 'callerid ' | cut -d':' -f2- | sed 's/^\ *//' | tr -d '\n')})
exten => s,n,Set(CALLERID_NUM=${CUT(CUT(FROM,<,1),<2)})
exten => s,n,Set(FROM_SIP=${STRREPLACE(MESSAGE(from),<sip:${PEER}@,<sip:${CALLERID_NUM}@)})
exten => s,n,MessageSend(pjsip:${ARG1},${FROM_SIP})
exten => s,n,Hangup()
    

For more details, please refer to the documentation on the configuration of the dial plan:
https://wiki.asterisk.org/wiki/display/AST/Contexts%2C+Extensions%2C+and+Priorities.

5. Restart the asterisk service.

sudo systemctl restart asterisk

Communication tests with the Web Browser Phone client

After restarting the asterisk service. We will go to the Raspberry Pi and install Mozilla Firefox via the command sudo apt install firefox-esr.

Let's open Mozilla Firefox and enter the following URL in the address bar: https://www.innovateasterisk.com/phone/.

You must configure the fields as follows and then click on Save:

(Figure 19 - Configuring the WebRTC Browser Phone client)

The Registered indication indicates that the client is indeed connected to the Asterisk server.

(Figure 20 - Registering the rpi client on the Asterisk server)

So we can call the Alcatel IP Touch.

(Figure 21 - Raspberry Pi to Alcatel IP Touch 4018 EE call)

(Figure 22 - Receiving the call from the Raspberry Pi)

Or from the Alcatel, call the Raspberry Pi.

(Figure 23 - Alcatel IP Touch 4018 EE to Raspberry Pi call)

(Figure 24 - Receiving the call from the Alcatel IP Touch)

Development of a JavaScript SIP client

Given the project and time constraints, we only had time to test existing solutions (Browser Phone). However, the development of a SIP client can be done via already existing libraries such as: SIP.js, JsSip, sipML5... For your information Browser Phone uses SIP.js.

Conclusion

This project allowed the development of a VoIP communication between a Raspberry Pi and an Alcatel IP Touch 4018 EE phone. We discovered new protocols such as SIP or RTP and the WebRTC API, which allows us to encapsulate SIP to establish a communication between several endpoints, using a web browser or not. It is easy to see the potential of IP for communications, and the need to use this type of technology in companies or public services.

This project could be improved by the implementation of encryption in particular with the SIPS protocol (SIP over SSL/TLS) at the level of the establishment of the communication between two end points (done in part when encapsulated in a WebSocket over TLS), but this could be integral, since the Alcatel supports SIP over TLS very well. Finally, the RTP protocol can also be encrypted, this is the SRTP (Secure Real-time Transport Protocol) when there is a call in progress.

I would like to thank Mr Lounis for having proposed this experimental work, which was particularly enriching, as I was unaware of a large part of the functioning of IP telephony. This subject allowed me to manipulate both the back end and front end of the system and to measure the power of this architecture when it is integrated into the Internet network.

Abbreviations

References

(1): Wikipedia, Business telephone system, 21^st january 2022, available at: https://en.wikipedia.org/wiki/Business_telephone_system.

(2): ITU Telecommunication Standardization Sector, ISDN user-network interface layer 3 specification for basic call control, ITU, May 1998, available at: https://www.itu.int/rec/T-REC-Q.931-199805-I/en.

(3): ITU Telecommunication Standardization Sector, Interface between Data Terminal Equipment (DTE) and Data Circuit-terminating Equipment (DCE) for terminals operating in the packet mode and connected to public data networks by dedicated circuit, ITU, October 1996, available at: https://www.itu.int/rec/T-REC-X.25-199610-I/en.

(4): Matt Fredrickson, PSA: chan_sip status changed to “deprecated” & Asterisk 17.0.0-rc2 Release, Asterisk.org, 25^th september 2019, available at: https://www.asterisk.org/deprecating-chan_sip-asterisk-17-0-0-rc2-release/.

(5): Y. Yeryomin, F. Evers and J. Seitz, II. The NAT and firewall problem In: Solving the firewall and NAT traversal issues for SIP-based VoIP, International Conference on Telecommunications, p.1-2, July 2008, DOI: 10.1109/ICTEL.2008.4652645, available at: https://www.researchgate.net/publication/224341038_Solving_the_firewall_and_NAT_traversal_issues_for_SIP-based_VoIP.

(6): Alcatel-Lucent, Audio characteristics In: Alcatel-Lucent IP Touch 4008/4018 Extended Edition Phones, p.2, 2013, available at: https://assets.bmdstatic.com/assets/Data/brochure/SKU01413265_2.pdf.

(7): Wikipedia, G.711, 20^th august 2021, available at: https://en.wikipedia.org/wiki/G.711.

(8): Malcolm Davenport, Answer, Playback, and Hangup Applications In: Asterisk Documentation, Asterisk.org, 19^th december 2013, available at: https://wiki.asterisk.org/wiki/display/AST/Answer%2C+Playback%2C+and+Hangup+Applications.

(9): bluesman, 64 bit Raspberry Pi OS is here!, Audiophile Style, 4^th june 2020, available at: https://audiophilestyle.com/forums/topic/59499-64-bit-raspberry-pi-os-is-here/.

(10): Alcatel-Lucent, 3.3. Initializing an IP Touch 40x8 EE phone In: IP Touch 4008/4018 Extended Edition - SIP Phone Installation Guide - 8AL90824AAAA ed02, p.7-8, August 2010, available at: https://www.cluster2.hostgator.co.in/files/writeable/uploads/hostgator136107/file/iptouchsipphoneinstallationguide-ed02.pdf.

(11): SECTIGO Store, ECDSA vs RSA: Everything You Need to Know, 9^th june 2020, available at: https://sectigostore.com/blog/ecdsa-vs-rsa-everything-you-need-to-know/.

(12): Agence nationale de la sécurité des systèmes d'information, Logarithme discret dans les courbes elliptiques définies sur GF(p) In: Référentiel Général de Sécurité version 2.0 - Annexe B1, p.19-20, 21^st february 2014, available at: https://www.ssi.gouv.fr/uploads/2014/11/RGS_v-2-0_B1.pdf.

All references were consulted on 1^st February 2022.

Attributions

Post Cover Image: Dooder, Social media and communication concept illustration, available at: https://www.vecteezy.com/vector-art/1105576-social-media-and-communication-concept-illustration.

Figure 5 - PBX Matra MC6500 serie: the original uploader was After310 at French Wikipedia, PABX Matra série MC6500, CC BY-SA 3.0, via Wikimedia Commons, available at: https://commons.wikimedia.org/wiki/File:PABX_Matra6500.JPG.

Figure 17 - Establishing a connection between two clients: adapted from the original work of Feyd-Aran, Etablissement d'une connexion par WebRTC, CC BY-SA 3.0, via Wikimedia Commons, available at: https://commons.wikimedia.org/wiki/File:Etablissement_d'une_connexion_par_WebRTC.svg.