Design and Implementation of a VOIP Solution for Overseas Travelers Using Asterisk and Kamailio

The growing demand for overseas travel services such as Wi‑Fi, local guides, and online assistance requires a seamless VOIP solution that lets users call domestic customer service numbers from abroad; a free network‑call feature was launched and achieved over 30% adoption within seven days.

VOIP Overview – VOIP (Voice over Internet Protocol) digitizes, compresses, and transports voice packets over IP networks using protocols like RTP, UDP, and SIP. The basic processing steps include analog‑to‑digital conversion, PCM sampling, packetization, transport, jitter removal, and digital‑to‑analog conversion.

Comparison with Traditional Telephony – While traditional PSTN offers higher stability, modern VOIP services (e.g., Mimicall, VP3000) provide comparable reliability with lower cost, especially when using free data or Wi‑Fi.

Open‑Source Soft‑PBX Selection – The two popular choices are FreeSwitch ( https://freeswitch.org/ ) and Asterisk ( http://www.asterisk.org/ ); the author chose Asterisk because of prior experience, both based on SIP.

SIP Client Library – PJSIP ( http://www.pjsip.org/ ) was selected for its support of SIP, SDP, RTP, STUN, TURN, ICE, clear API, NAT traversal, and cross‑platform compatibility (desktop, embedded, mobile).

Demo Setup

1) Install PJSIP on a Linux machine (installation steps omitted). Verify the shared library:

ldconfig -p | grep pj

2) Install Asterisk 13 (requires PJSIP ≥ 2.4). Configure the SIP endpoint in /etc/asterisk/pjsip.conf :

[transport-udp]
 type=transport
 protocol=udp
 bind=0.0.0.0:5061

[8005]
 type=endpoint
 context=from-external
 disallow=all
 allow=ulaw
 transport=transport-udp
 auth=8005
 aors=8005

[8005]
 type=auth
 auth_type=userpass
 password=8005
 username=8005

[8005]
 type=aor
 max_contacts=1

[8006]
 type=endpoint
 context=from-external
 disallow=all
 allow=ulaw
 transport=transport-udp
 auth=8006
 aors=8006

[8006]
 type=auth
 auth_type=userpass
 password=8006
 username=8006

[8006]
 type=aor
 max_contacts=1

3) Add dialing rules in /etc/asterisk/extensions.conf :

[from-external]
 exten => _80XX,1,NoOp()
 same => n,Dial(PJSIP/${EXTEN})

Start Asterisk:

service asterisk start
sudo asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv

Register the two extensions (8005, 8006) and test calls.

Load Balancing with Kamailio

To avoid a single point of failure, Kamailio is used as a SIP load balancer. The dispatcher list ( /usr/local/etc/kamailio/dispatcher.list ) defines backend Asterisk nodes:

1 sip:192.168.0.1 0 1 weight=50
1 sip:192.168.0.2 0 1 weight=50

Key Kamailio configuration snippets:

listen=udp:192.168.0.3:5060
modparam("dispatcher", "list_file", "/usr/local/etc/kamailio/dispatcher.list")
if(!ds_select_dst("$var(dispatcher_group)", "9")) {
    send_reply("404", "No destination dispatcher");
    exit;
}

After restarting Kamailio, calls are distributed according to the defined weights.

High Availability (HA) with Keepalived

Two servers (192.168.1.5 and 192.168.1.6) run Keepalived to provide a virtual IP (192.168.1.4). The Linux kernel parameter net.ipv4.ip_nonlocal_bind = 1 must be enabled so Kamailio can bind the virtual IP.

Keepalived is started on both nodes:

service keepalived start

Kamailio’s listen address is changed to the virtual IP:

listen=udp:192.168.1.4

After restarting Kamailio, calls can be directed to the virtual IP, providing seamless failover.

HA Health Check

SIPSak is used to probe Kamailio:

sipsak -s sip:[email protected]:5060

The exit code determines whether Kamailio is alive; a script can trigger Keepalived to switch the virtual IP when failures are detected.

Callback Solution

When a user’s mobile device cannot run VOIP, a callback mechanism holds the inbound call, notifies the other party (push or SMS), and releases the call once the second party is online, or uses a conference‑room approach.

NAT and Security

Typical enterprise firewalls and DNS require port forwarding and NAT handling; both Asterisk and Kamailio support NAT traversal. For stronger security, TLS and SRTP are added (compiled with tls.so and srtp.so ), and self‑signed certificates are sufficient.

Troubleshooting Tools

Use netcat for basic connectivity, tcpdump or ssldump for SIP/TLS analysis, and Wireshark for comprehensive packet inspection.

Conclusion

The presented architecture demonstrates a practical VOIP integration for overseas users, covering PBX selection, SIP client libraries, load balancing, HA, NAT, security, and debugging, offering a solid reference for similar projects.