selfphoneng/docs/admin-guide/network.md

Network, firewall and attacks

Most call problems come from the network between the server and the internet, and any PBX on the internet is attacked within hours of going up. This page covers both.

NAT: a server behind a router

Calls carry the server's address inside the SIP messages. Behind a router (NAT), the server has to know its public IP and which networks are local, or the other side sends the audio to the wrong place.

1. Turn off SIP ALG (also called "SIP helper" or "SIP fixup") on every router between the server and the internet. It rewrites SIP messages and is the classic cause of one-way audio and calls dropping after 30 seconds. Where to find it depends on the router; search for " disable SIP ALG".
2. Forward ports to the server's LAN IP:
   • UDP 5060 (SIP);
   • UDP 10000–20000 (audio; see Call audio ports to use another range);
   • TCP 443 (and 80 with the bundled Caddy);
   • TCP 5060 only for phones or trunks set to TCP (Settings → Security → Accept SIP over TCP; phones this server sets up use UDP);
   • TCP 5061 only if you use SIP over TLS.
   • If nothing reaches the server on UDP 5060 while other UDP ports do (System → Diagnostics, a Complete network test), the router or the ISP keeps that port for itself: often an ISP box with phone jacks, or a SIP ALG you can't turn off. Set another SIP port in Settings → Security, for example 5080, forward that instead, restart telephony, and set the phones up again. Trunks keep working: they're reached from the port the server registers from.
3. Double NAT (the ISP's box in front of your own router): forward the ports on both, or put your router in the ISP box's DMZ / bridge mode.
4. Public IP and local networks.
   • The wizard's Network step sets both (go back to it at /setup?step=3).
   • The public IP is also on Settings → Network: Detect fills it in.
   • Local networks are the networks that reach the server without NAT. The default (192.168.0.0/16, 10.0.0.0/8, 172.16.0.0/12) fits most offices.
5. Dynamic IP? Tick "Re-check the public IP every 5 minutes" on Settings → Network.
   • When the IP changes, SelfPhoneNG updates its settings and emails the admins ("Public IP changed to …").
   • Then restart telephony so Asterisk uses the new address: on the server, run docker compose restart asterisk (calls in progress drop). The dashboard shows "SIP port or transport changes need a telephony restart" until you do, and System shows the command.

Call audio ports

Call audio (RTP) uses UDP 10000–20000 unless you set another range in .env (or, in a Portainer stack, in the api's environment):

RTP_PORT_START=40000
RTP_PORT_END=40199

Then docker compose up -d (the api writes the range into Asterisk's config), and open the same range on the firewall and the router instead of 10000–20000. Each call uses two ports, so 200 ports is plenty for a small office. Pick a range nothing else on the server uses: System → Diagnostics counts how many ports in the range are already taken.

Diagnostics: is the server reachable?

System → Diagnostics checks what usually stops calls from working outside the office, and says how to fix each problem it finds.

This server runs every time you open the page:

• the public IP is set, and is the address the server really reaches the internet from;
• the host name's DNS record points at that address (with Cloudflare, it must be "DNS only": its proxy carries web pages, not SIP);
• the server's LAN address is in the local networks;
• whether phones may sign in from anywhere (the phone network limit refuses phone apps on mobile data), and whether phone setup is limited to the local networks;
• Asterisk is listening on the SIP ports;
• how many call audio ports are free.

From the internet needs a network test probe: a small program, run on a machine outside your network, that connects back to the server the way a phone on mobile data or your SIP provider does. A server can't test this itself: most routers either don't route a connection to their own public address back in, or route it without going through the port forwards.

• Quick network test: TCP 80 and 443, and SIP on 5060 over UDP (and TCP if it's on).
• Complete network test: also SIP over TLS (if on), the browser phone's /ws, and a sample of the call audio ports (the server listens on them while the probe sends to them).

To run a probe, on any machine outside your network (a small VPS is plenty). It needs about 50 MB of memory and one open TCP port (8080).

Without Docker: each release has the probe as a single executable for Linux, with everything it needs inside (selfphoneng-probe-linux-x64, or -arm64 for ARM machines), next to SHA256SUMS. With Node 22 or newer installed you can take the small selfphoneng-probe.cjs instead.

v=v0.0.5   # the release
curl -fLO https://git.dou.bet/spng/selfphoneng/releases/download/$v/selfphoneng-probe-linux-x64
curl -fLO https://git.dou.bet/spng/selfphoneng/releases/download/$v/SHA256SUMS
sha256sum --check --ignore-missing SHA256SUMS
sudo install -m 755 selfphoneng-probe-linux-x64 /usr/local/bin/selfphoneng-probe
curl -fLO https://git.dou.bet/spng/selfphoneng/raw/tag/$v/deploy/probe/selfphoneng-probe.service
sudo cp selfphoneng-probe.service /etc/systemd/system/
sudo systemctl enable --now selfphoneng-probe
sudo ufw allow 8080/tcp

(On GitHub, the downloads are at https://github.com/iamdoubz/selfphoneng/releases/download/$v/….) To try it first without the service, run ./selfphoneng-probe-linux-x64 and open http://<machine>:8080/healthz. PORT changes the port; behind your own HTTPS reverse proxy that sets X-Forwarded-For, also set TRUST_PROXY=true.

With Docker:

docker compose -f deploy/probe/compose.yml up -d

Then put its address, for example http://203.0.113.50:8080, in Settings → Network → Network test probe. The probe only ever tests the address a request comes from, so nobody can use it against someone else, and each address gets a few tests per 10 minutes. Its SIP test is one OPTIONS request; if you run fail2ban or CrowdSec, add the probe's address to their allow list so a test never counts toward a ban.

Firewall

Allow only the ports in the port table. All SelfPhoneNG containers use the host's network, so the host firewall (ufw, firewalld, nftables) applies to them directly.

With ufw:

ufw allow OpenSSH
ufw allow 443/tcp
ufw allow 80/tcp                 # only with the bundled Caddy
ufw allow 5060/udp
ufw allow 10000:20000/udp          # or your RTP_PORT_START:RTP_PORT_END
ufw enable

Only voip.ms may reach SIP (no remote phones)

If every desk phone is in the office, nothing else on the internet needs port 5060. Limit it to voip.ms and your local networks.

1. Look up your POP's address (the server you put on the trunk):

   dig +short chicago1.voip.ms
   

2. Save this as /etc/nftables.d/selfphoneng.nft, with that address in sip_ok. Add any other office that has phones (the same networks as Settings → Security → "Other networks allowed").

   table inet selfphoneng {
     set sip_ok {
       type ipv4_addr
       flags interval
       elements = { 10.0.0.0/8, 172.16.0.0/12, 192.168.0.0/16, 208.100.60.8 }
     }
     chain input {
       type filter hook input priority 0; policy accept;
       meta l4proto { tcp, udp } th dport { 5060, 5061 } ip saddr != @sip_ok drop
       meta nfproto ipv6 meta l4proto { tcp, udp } th dport { 5060, 5061 } drop
     }
   }
   

3. Load it: nft -f /etc/nftables.d/selfphoneng.nft. To keep it after a reboot, include it from /etc/nftables.conf.

This table only drops SIP from elsewhere and leaves your other rules alone. voip.ms can change its addresses: if the trunk stops registering one day, run dig again.

Desk phones only from the office

Settings → Security → "Only accept desk phones from the networks below" makes Asterisk refuse desk phones signing in from anywhere but your local networks and the ones you list. A stolen phone password is then useless from outside.

• Browser phones aren't affected: they come in through HTTPS.
• "Only to phones on the local networks" does the same for the phone setup files.

Blocking password guessers: fail2ban or CrowdSec

Asterisk writes every failed sign-in to data/asterisk-logs/security.log. Use one of the two tools below; both read that file on the host.

fail2ban

Install it (apt install fail2ban) and create /etc/fail2ban/jail.d/selfphoneng.conf. Change the path to where you cloned SelfPhoneNG:

[asterisk]
enabled  = true
logpath  = /opt/selfphoneng/data/asterisk-logs/security.log
maxretry = 5
findtime = 10m
bantime  = 1d
# never ban your own offices
ignoreip = 127.0.0.1/8 ::1 10.0.0.0/8 172.16.0.0/12 192.168.0.0/16

Then:

systemctl restart fail2ban
fail2ban-client status asterisk

The stock asterisk filter already understands the log. We test it against real failed sign-ins (scripts/e2e/m11-t7.mjs).

CrowdSec

Install CrowdSec and a bouncer (https://docs.crowdsec.net/), then:

cscli collections install crowdsecurity/asterisk

Add /etc/crowdsec/acquis.d/selfphoneng.yaml:

filenames:
  - /opt/selfphoneng/data/asterisk-logs/security.log
labels:
  type: asterisk

and run systemctl reload crowdsec.

Web sign-in attacks

The web sign-in is rate limited. Failed attempts are written to the api's log as "event":"auth.failed" with the ip, so you can feed them to fail2ban or CrowdSec too (docker compose logs api). Turn on two-step sign-in for every admin.

Toll fraud

Someone with a stolen SIP password calling expensive international numbers is the attack that costs real money. Keep:

• international calling off at voip.ms and in Outbound rules unless you need it;
• the voip.ms balance alerts and spending limits on;
• the desk phone network limit above on if your phones are all in the office;
• the outside calls at once limits (Settings → Security) as low as your office needs: per extension (3 by default) and for the whole system (10). A stolen password then runs up at most that many calls at a time. Forwarded calls count for the person who set the forward; 911 is never limited. Callers over a limit hear "all circuits are busy".

Never reuse a SIP password. SelfPhoneNG generates long random ones; phones get them from the setup files.