Blogs

Voice over Internet Protocol

Hola folks!
I’m Harsh aka G4RUD4, and today I’m sharing a beginner-friendly guide to setting up a VoIP environment. Whether you’re looking to experiment with VoIP tech, dive into packet-level analysis using Wireshark, or build challenges for cybersecurity CTFs, this post has you covered.

What is VoIP?

VoIP (Voice over Internet Protocol) allows you to make voice and video calls over the internet instead of traditional phone lines. It converts analog voice signals into digital data that travels over IP networks.

Here’s a simple diagram to show how it works:

Basic Workflow

Image3

What You’ll Need

You can either use two VMs or, like me, run one instance on Windows and another on a Linux VM (Ubuntu 24.04 on Oracle VirtualBox).

Required Tools:

  • Asterisk (v22.1.0) – Acts as your VoIP server

  • Linphone (v5.0.2) – A softphone for making calls

Setting Up Asterisk Server on Ubuntu

Open your terminal and run:

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wget https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.1.tar.gz | tar -zxvf asterisk-22.4.1.tar.gz

`

I used asterisk-22.1.0.tar.gz, so update the version as needed.

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sudo apt-get install build-essential libssl-dev libncurses5-dev libnewt-dev libxml2-dev libsqlite3-dev uuid-dev libedit-dev libjansson-dev sqlite3

Navigate to the Asterisk folder and enable SIP and RTP modules before building:

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sudo make menuselect

Enable chan_pjsip and chan_rtp modules. Save and exit.

menuselect

Then run the following commands one by one or by script:

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./configure
make -j$(nproc)
sudo make install
sudo make samples
sudo make config

This will take some time. Be patient!

Configuring Asterisk

Edit these files in /etc/asterisk/:

pjsip.conf – SIP user and transport configs

extensions.conf – Dialplan logic

';' is for the comments

pjsip.conf:

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[101]
auth=101 ; we are using username as 101
aors=101
type=endpoint
context=internal
message_context=internal
transport=udp_transport ; Mention the transport protocol you want to use
language=en
deny=0.0.0.0/0.0.0.0
permit=192.168.0.0/24
disallow=all
allow=h264,opus ; The protocols which you want to allow
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes ; If you want to rewrite existing contact then keep 'yes'
direct_media=no ; We are using softphone and we don't want users to share directly
outbound_auth=101
dtmf_mode=rfc4733 ; If you want to use the dial pad which uses DTMF protocol
tos_audio=ef              ; Audio RTP
tos_video=af41            ; Video RTP
cos_audio=5               ; Audio RTP CoS
cos_video=4               ; Video RTP CoS
force_avp=yes
media_encryption=no
callerid="101" <101>

[101]
type=aor
qualify_frequency=30
;contact=sip:101@192.168.0.104:5060
max_contacts=1
remove_existing=yes ;  If you want to rewrite existing contact then keep 'yes'

[101]
type=auth
auth_type=userpass
username=101
password=linphone_password1

[102]
auth=102 ; user as 102
aors=102
type=endpoint
context=internal
message_context=internal
transport=udp_transport
language=en
deny=0.0.0.0/0.0.0.0
permit=192.168.0.0/24
disallow=all
allow=h264,opus
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
outbound_auth=102
dtmf_mode=rfc4733
tos_audio=ef              ; Audio RTP
tos_video=af41            ; Video RTP
cos_audio=5               ; Audio RTP CoS
cos_video=4               ; Video RTP CoS
force_avp=yes
media_encryption=no
callerid="102" <102>

[102]
type=aor
qualify_frequency=30
;contact=sip:102@192.168.0.104:5060
max_contacts=1
remove_existing=yes

[102]
type=auth
auth_type=userpass
username=102
password=linphone_password2

[udp_transport]
type=transport
protocol=udp
bind=0.0.0.0:5060
local_net=192.168.0.0/24
;external_media_address=192.168.0.104 ; Replace with your Asterisk server's local IP
;external_signaling_address=192.168.0.104 ; Replace with your Asterisk server's local IP
tos=cs3                   ; SIP signaling
cos=3                     ; SIP signaling CoS

allow_reload=false

extensions.conf:

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[internal]
; Call handling for endpoints
exten => 101,1,Dial(PJSIP/101,60)
exten => 101,n,Hangup()

exten => 102,1,Dial(PJSIP/102,60)
exten => 102,n,Hangup()

; Handle sending messages (plain text or file)
; Extension 105 sends to 102

; Handle sending messages (plain text)
exten => 105,1,Verbose(1, "Sending a message from 101 to 102")

 same => n,Set(MESSAGE(body)=We will talk further on VoIP)

 ;same => n,Set(MESSAGE(body)=Hey X1E, I hope you're doing well. I'll be sending some data shortly.)

 ;same => n,Set(MESSAGE(body)=Any Message, you can also share the file as base64)

 ;same => n,Set(MESSAGE(body)=${FILE(/home/dr4k3n/Downloads/alwoeenveoivr)})
 ;same => n,Set(MESSAGE(body)=${FILE(/home/dr4k3n/Downloads/delervierveiv)})
 ;same => n,Set(MESSAGE(body)=${FILE(/home/dr4k3n/Downloads/syveiubviesveve)})

 same => n,MessageSend(pjsip:102@192.168.0.104:5060,sip:101@192.168.0.104:5060,sip:102@192.168.0.104:5060)  ; Send the message using MessageSend() through ASTERISK. 

 ;same => n,MessageSend(pjsip:102@192.168.0.101:5060,sip:101@192.168.0.104:5061,sip:102@192.168.0.101:5060)  ; Send the message using MessageSend() Directly through IP. {Check the IP, username and port}

 same => n,GotoIf($["${MESSAGE_SEND_STATUS}" = "SUCCESS"]?sent:failed)
 same => n(sent),Verbose(1, "Message sent successfully from 101 to 102")
 same => n,Hangup()
 same => n(failed),Verbose(1, "Failed to send message: ${MESSAGE_SEND_STATUS}")
 same => n,Hangup()

; Handle sending messages (plain text)
exten => 106,1,Verbose(1, "Sending a message from 102 to 101")
 ;same => n,Set(MESSAGE(body)=I'm good, hope you're doing well too. Don't forget to do VC, GET AWAY ASAP.)
 same => n,Set(MESSAGE(body)=Gotcha)

 same => n,MessageSend(pjsip:101@192.168.0.104:5060,sip:102@192.168.0.104:5060,sip:101@192.168.0.104:5060)  ; Send the message using MessageSend() Through ASTERISK

 ;same => n,MessageSend(pjsip:101@192.168.0.104:5061,sip:102@192.168.0.101:5060,sip:101@192.168.0.104:5061)  ; Send the message using MessageSend() Directly through IP

 same => n,GotoIf($["${MESSAGE_SEND_STATUS}" = "SUCCESS"]?sent:failed)
 same => n(sent),Verbose(1, "Message sent successfully from 101 to 102")
 same => n,Hangup()
 same => n(failed),Verbose(1, "Failed to send message: ${MESSAGE_SEND_STATUS}")
 same => n,Hangup()

; Handle SIP MESSAGE requests
exten => _MESSAGE.,1,NoOp(Handling SIP MESSAGE)
 same => n,Set(ACTUAL_TO=${CUT(MESSAGE(to),@,1)})  ; Extract the recipient
 same => n,MessageSend(${ACTUAL_TO},sip:${MESSAGE(from)})  ; Send the message
 same => n,NoOp(Send status: ${MESSAGE_SEND_STATUS})  ; Log the send status
 same => n,GotoIf($["${MESSAGE_SEND_STATUS}"="SUCCESS"]?success:fail)

; On success
 same => n(success),Verbose(1, Message sent successfully)
 same => n,Hangup()

; On failure
 same => n(fail),Verbose(1, Failed to send message: ${MESSAGE_SEND_STATUS})
 same => n,Hangup()

; Echo test
exten => 200,1,Answer()
exten => 200,n,Echo()
exten => 200,n,Hangup()

; Handle DTMF input
exten => _X.,1,Set(CHANNEL(dtmfmode)=rfc2833)
exten => _X.,n,Playback(digits/${EXTEN})
exten => _X.,n,Hangup()

You configure it for the call, messages, files sharing, DTMF(Dial Pad).

Let’s ensure Asterisk loads the ‘chan_pjsip’ module. We can do this by opening the ‘module.conf’ file and updating a specific line.

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[modules]
autoload=yes

Make sure autoload is yes.

Starting Asterisk Server

Now we are done. Start Asterisk with:

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sudo asterisk -r
sudo asterisk -rvvvvvvvvv

for verbosity -^

Asterisk Server

Now if you update in the files like pjsip.conf and extensions.conf then you have to run these commands to reload the configurations:

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sudo asterisk -rx "dialplan reload"
sudo asterisk -rx "pjsip reload"
OR
sudo asterisk -rx "core reload"

For a complete reload -^

To manage the service:

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sudo systemctl restart asterisk
sudo systemctl enable asterisk
sudo systemctl status asterisk

If you run the pjsip show endpoints in the asterisk terminal

You will see this as the output:

Endpoints

If the softphone is not yet connected, the status will show as “Unavailable.”

Enable debugging by running it in asterisk terminal:

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pjsip set logger on
rtp set debug on

pjsip log:

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dr4k3n*CLI> pjsip set logger on
PJSIP Logging enabled
<--- Transmitting SIP request (421 bytes) to UDP:192.168.0.104:5060 --->
OPTIONS sip:102@192.168.0.104:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:5060;rport;branch=z9hG4bKPj78497896-090d-491a-babc-a1fdf67fa07d
From: <sip:101@192.168.0.104>;tag=1b1058dd-748f-457c-a6b4-2f720ffb4865
To: <sip:102@192.168.0.104>
Contact: <sip:101@192.168.0.104:5060>
Call-ID: b6032719-63e3-4cf7-8285-11064df2db92
CSeq: 25292 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 22.1.0
Content-Length:  0

It continously sends SIP to establish the connection between Asterisk Server and Liphones.

Setting Up Linphone (Softphone)

We’ll use Linphone to connect with our Asterisk server. Other softphone alternatives include MicroSIP, Zoiper, and Mizudroid (Android).

If Linphone doesn’t work on your Asterisk server machine, try Zoiper instead.

Open Required Ports:

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sudo ufw allow 5060/udp
sudo ufw allow 10000:20000/udp

Use the same version of Linphone on both Windows and Linux to avoid compatibility issues.

For Ubuntu OS:

~/.config/linphone/linphonerc file:

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[messaging]
use_cpim=0
default_content_type=text/plain

[misc]
log_collection_upload_server_url=https://www.linphone.org:444/lft.php
empty_chat_room_deletion=1
prefer_basic_chat_room=1
uuid=d6ef36a5-972b-0091-9814-1225cc002681
hide_empty_chat_rooms=1
file_transfer_server_url=https://www.linphone.org:444/lft.php

[sound]
ec_filter=MSWebRTCAEC
playback_dev_id=PulseAudio Unknown: Built-in Audio Analog Stereo
ringer_dev_id=PulseAudio Unknown: Built-in Audio Analog Stereo
capture_dev_id=PulseAudio Unknown: Built-in Audio Analog Stereo
media_dev_id=PulseAudio Unknown: Built-in Audio Analog Stereo
mic_gain_db=0.000000
playback_gain_db=0.000000
remote_ring=/usr/share/sounds/linphone/ringback.wav
disable_record_on_mute=0
ec_delay=-1

[app]
record_aware=1
auto_download_incoming_files_max_size=10000000
auto_download_incoming_voice_recordings=1
auto_download_incoming_icalendars=1
sender_name_hidden_in_forward_message=0

[rtp]
accept_any_encryption=1
audio_rtp_port=10000
video_rtp_port=12000
text_rtp_port=11078
audio_jitt_comp=60
video_jitt_comp=60
nortp_timeout=30
audio_adaptive_jitt_comp_enabled=1
video_adaptive_jitt_comp_enabled=1
audio_dscp=0xb8
video_dscp=0x88

[sip]
chat_messages_aggregation_delay=1000
chat_messages_aggregation=1
zrtp_key_agreements_suites=MS_ZRTP_KEY_AGREEMENT_K255_KYB512
root_ca=/usr/share/linphone/rootca.pem
verify_server_certs=1
verify_server_cn=1
contact=sip:dr4k3n@unknown-host
auto_send_ringing=1
media_encryption=none
supported=replaces, outbound, gruu, path, record-aware
user_agent=Linphone Desktop/5.0.2 (dr4k3n) Ubuntu 24.04.1 LTS, Qt 5.15.13 LinphoneCore/5.2.0
default_proxy=0
guess_hostname=1
inc_timeout=30
in_call_timeout=0
delayed_timeout=4
register_only_when_network_is_up=1
register_only_when_upnp_is_ok=1
sip_transport=udp,tcp
sip_port=5061
sip_tcp_port=5070
sip_tls_port=-1
use_rfc2833=1
use_info=0
dscp=0x18
ipv6_migration_done=1
use_ipv6=0
linphone_specs=ephemeral/1.1,groupchat/1.1,lime
use_cpim=0
default_content_type=text/plain
media_encryption_mandatory=0

[video]
max_conference_size=vga
displaytype=MSQOGL
capture=1
display=1
main_display_mode=Hybrid
other_display_mode=Hybrid
size=svga
show_local=0
preset=default
framerate=24.000000

[ui]
use_qrcode=0
rc_version=5
read_and_agree_terms_and_privacy=1
logs_enabled=1
exit_on_close=1
show_tel_keypad_automatically=0
show_take_screenshot_button_in_call=1

[net]
friendlist_subscription_enabled=1
mtu=1300
adaptive_rate_control=1
download_bw=4000
upload_bw=4000
nat_policy_ref=USbqGgS2ivIxLBv

[lime]
enabled=1
last_update_time=1734544797

[account_creator]
url=http://fs-test-sandbox.linphone.org/flexiapi/api/

[audio_codec_0]
mime=opus
rate=48000
channels=2
enabled=1
recv_fmtp=useinbandfec=1
bitrate=64000

[video_codec_0]
mime=VP8
rate=90000
enabled=0

[video_codec_1]
mime=H264
rate=90000
enabled=1
recv_fmtp=profile-level-id=42801F
bitrate=1500000

[proxy_default_values]
avpf=1
dial_escape_plus=0
publish=1
quality_reporting_enabled=1
quality_reporting_interval=180
reg_expires=600
reg_sendregister=1
quality_reporting_collector=sip:voip-metrics@sip.linphone.org;transport=tls
reg_identity=sip:?@sip.linphone.org
reg_proxy=<sip:sip.linphone.org;transport=tls>
nat_policy_ref=nat_policy_default_values
realm=sip.linphone.org
contact_parameters=message-expires=604800
conference_factory_uri=sip:conference-factory@sip.linphone.org
audio_video_conference_factory_uri=sip:videoconference-factory@sip.linphone.org
cpim_in_basic_chat_rooms_enabled=1
rtp_bundle=1
lime_server_url=https://lime.linphone.org/lime-server/lime-server.php

[assistant]
algorithm=SHA-256
password_max_length=-1
password_min_length=1
username_length=-1
username_max_length=64
username_min_length=1
username_regex=^[a-z0-9+_.\-]*$
domain=sip.linphone.org
xmlrpc_url=https://subscribe.linphone.org:444/wizard.php

[nat_policy_default_values]
protocols=stun,ice
ice_enabled=0

[nat_policy_0]
ref=USbqGgS2ivIxLBv

[auth_info_0]
username=101
ha1=91fae9b1f17bfe1f0aac6af1f16197f0
realm=asterisk
domain=192.168.0.104
algorithm=MD5
available_algorithms=MD5

[proxy_0]
reg_proxy=<sip:101@192.168.0.104:5060;transport=udp>
reg_identity=sip:101@192.168.0.104:5060
realm=asterisk
contact_parameters=message-expires=604800
quality_reporting_collector=sip:voip-metrics@sip.linphone.org;transport=tls
quality_reporting_enabled=1
quality_reporting_interval=180
reg_expires=600
reg_sendregister=1
publish=1
avpf=1
avpf_rr_interval=1
dial_escape_plus=0
use_dial_prefix_for_calls_and_chats=1
privacy=32768
push_notification_allowed=0
remote_push_notification_allowed=0
cpim_in_basic_chat_rooms_enabled=1
idkey=proxy_config_oNbn27hGUzwy2mr
publish_expires=3600
conference_factory_uri=sip:conference-factory@sip.linphone.org
rtp_bundle=1
rtp_bundle_assumption=0
lime_server_url=https://lime.linphone.org/lime-server/lime-server.php

If possible keep the configuration same for both the Linphone.

Configure Linphone settings to match your SIP user and network setup. Here’s how the user and network settings should look:

SIP User

Network

Mostly similar with different port on Windows Linphone, Before starting the call make sure that SIP users and Asterisk server contains the same IP address.

If Ubuntu Linphone doesn’t connect automatically, use a static client first. You can temporarily comment out the contact= line in pjsip.conf to help establish the connection. Once connected, it’ll switch to dynamic mode automatically.

You’ll know it worked when both Linphones show a green “Online” indicator.

avail screen

Making the Call

Establish the Call from user 101 by dialing 102 from dial pad and you will get the call UI like this:

call

After making the call, you can set the rtp debug on which will show you rtp packets which help us to identify whether the call has been connected successfully or not as shown below:

rtp

Success! You’re now sending voice over the network using Opus codec. For video calls, Linphone uses H.264.

You can also send messages by dialing different users like 105 or 106 (as defined in extensions.conf).

Deep Dive with Wireshark

Use Wireshark to inspect:

  • SIP signaling

  • RTP media packets

  • Opus/H.264 streams

  • DTMF signals

This is where it gets exciting for CTF enthusiasts. Capture the VoIP traffic and create puzzles around:

  • Voice recovery

  • DTMF-based flag inputs

  • Hidden messages in SIP headers

Want an example?
Check out my VoIP-based CTF challenge write-up here where you can get to see the SIP, RTP, H.264, Opus, DTMF packets.

Conclusion

Setting up VoIP might sound intimidating at first, but with Asterisk and Linphone, it’s totally doable—even on a basic setup. Whether you’re learning about network protocols, building CTF challenges, or just want to experiment with audio/video over IP—VoIP is a great playground.