How to Set Up a SIP Softphone on macOS with SIPdia

Setting up SIPdia on a Mac takes one form and one button. Sign in, open the Account page, enter Server, Transport, Username and Password, then click Save and Register. When the dot in the sidebar turns green, the account is registered and you can make and receive calls. The rest of this tutorial walks through each step, the permission prompts macOS shows, the keyboard shortcuts, and two limits you should know before you rely on the app.

SIPdia runs on macOS 14 or later, on Apple silicon and Intel Macs. It works with standards-based SIP servers: a PBX such as Issabel, FreePBX or Asterisk, where you register as an extension, or a SIP provider, where you use the account details from the provider's portal.

What you need before you start

Collect these details from your PBX administrator or your provider's portal:

• The server address. Providers may call it the registrar, proxy or SIP host. Add :port after the host only if the server does not use the standard port for the transport.
• The transport: UDP, TCP or TLS. Most PBX extensions use UDP on port 5060. TLS normally uses TCP port 5061.
• The username. On a PBX this is the extension number.
• The password, called the secret on Asterisk-based systems.
• Optionally an auth username, a separate SIP domain or an outbound proxy, if the provider lists them separately from the server.

If you run the PBX yourself, create the extension first and test it with any softphone. A server-side problem is much easier to spot when you know the extension already works.

Step-by-step setup

1. Sign in. SIPdia opens on a sign-in screen. Sign in with Apple or Google. The phone engine starts, and SIP accounts register, only after you sign in.
2. Open the Account page. Click Account in the sidebar.
3. Enter the server. In Server, type the host name or IP address of the PBX or provider. Add :port if the server listens on a non-standard port, for example pbx.example.com:5080. When you leave the port out, SIPdia performs a DNS SRV lookup as described in RFC 3263 and falls back to the default port for the transport.
4. Choose the transport. Set Transport to UDP, TCP or TLS. Use what the PBX or provider documents. Choose TLS only when the server has a certificate signed by a public certificate authority (see the limits below).
5. Enter the credentials. Type the Username (the extension number on a PBX) and the Password. You can also set a Display name, which is the name shown to people you call when the server passes it on.
6. Fill Advanced fields only if needed. Under Advanced you will find Domain, Outbound proxy, Auth username, STUN server, ICE and Encrypt audio. Leave them empty or at their defaults unless the provider lists a separate domain, proxy or authentication username, or unless you need NAT help (see below).
7. Click Save and Register. SIPdia sends a SIP REGISTER request to the server. The dot in the sidebar turns green when the registration is accepted.
8. Allow the permission prompts. macOS asks for three permissions the first time: the microphone (for your voice), the local network (so the app can reach a PBX on your LAN) and notifications (for missed-call alerts). Allow all three. The local network prompt appears on macOS 15 and later; macOS 14 does not ask for it. Apple explains why the local network prompt appears in TN3179.
9. Add more accounts if you have them. Click Add Account on the Account page and repeat steps 3 to 7. On the free plan only the default account registers; SIPdia Premium removes that limit. See How to Add Multiple SIP Accounts to a Softphone.

Behind a router: STUN and ICE

If your Mac sits behind a home or office router and the server is on the internet, the server sees your router's public address while your SIP messages carry your private address. The usual symptom is one-way or no audio: the call connects, but one side hears silence.

Two settings under Advanced fix most of these cases:

• STUN server. Enter a STUN server such as stun.l.google.com:19302. STUN (RFC 5389) lets SIPdia discover its public address and port so it can advertise them in the call.
• ICE. Turn on ICE (RFC 8445). ICE gathers several candidate addresses and tests them with the other side, so audio takes whichever path actually works.

If you connect to an office PBX through a VPN, you normally do not need either. If the PBX is yours, also set NAT handling on the extension (nat=force_rport,comedia or the older nat=yes on chan_sip; force_rport, rtp_symmetric and rewrite_contact on PJSIP). More detail is in One-way audio on VoIP calls.

Daily use

• Make a call. Type a number on the Phone page and press Return. With more than one registered account, choose which one to call from with the From picker on the dialpad.
• Receive a call. A floating incoming-call window appears on every Space, including over full-screen apps. Click Answer.
• Hold, mute and transfer. During a call you can mute, hold and resume, send keypad tones (RFC 2833), and transfer the call. Blind transfer sends the caller straight to another number; attended transfer connects two calls you already have open.
• Record a call (Premium). Press Record during the call, then press it again to stop. Each call can be recorded once. WAV files are saved to ~/Music/Call Recordings by default and listed on the Recordings page. A notice before recording reminds you that the other party is not told.
• Forwarding and Do Not Disturb (Premium). On the Forwarding page, enter a number and click Turn On. SIPdia dials the PBX feature code, you hear the PBX confirmation, and the short call ends. This needs a PBX with feature codes; the codes are editable under Settings > Feature Codes.
• Close the window, keep the phone. SIPdia shows a menu bar icon and keeps running when you close the window. You can also turn on open at login and choose separate microphone, speaker and ring devices.

Keyboard shortcuts

Action	Shortcut
Answer	⇧⌘A
Hang up	⇧⌘E
Mute	⇧⌘M
Hold	⇧⌘H
Record	⇧⌘R
Send a keypad tone	type the digit while the call screen is focused

Limits to know

TLS certificates must come from a public certificate authority. SIPdia checks the server certificate against the system's trusted roots. Self-signed PBX certificates are rejected. If your PBX only has a self-signed certificate, either install one from a public CA on the PBX, or use UDP or TCP over a VPN instead of TLS.

Forwarding status may be stale. The Forwarding page shows what SIPdia last sent. Changes made from a desk phone or the PBX web panel do not appear in the app.

Codecs. Opus, G.711 ulaw and G.711 alaw are offered. G.722 and G.729 are not.

If registration fails

• "Could not reach the server". Check the address, port and transport. For a PBX on the office network, allow SIPdia under System Settings > Privacy & Security > Local Network. If you chose TLS, remember it uses TCP port 5061 unless the Server field names another port; some networks block 5061, so ask for it to be opened or use the provider's alternative port.
• Wrong password or username. The server answers 401 or 403 and the dot stays red. Re-enter the password and, if the provider lists a separate authentication name, put it in Auth username.
• Detailed logs. Turn on Settings > General > Save detailed logs, open the app again and reproduce the problem. Click Show Logs to find the files. The logs include full SIP messages, so turn logging off again afterwards.

More cases are covered in SIP registration keeps failing.

Next steps

• Read the SIPdia for macOS page for the full feature list.
• Running Issabel? The Issabel integration guide covers the extension, codec, DTMF and firewall settings on the PBX side.
• Using a provider instead of a PBX? See How to Connect a Softphone to an ISP or Provider SIP Account.