How to Connect a Softphone to an ISP or Provider SIP Account

Connecting a softphone to a provider account comes down to copying five values from the provider's portal into the app: the SIP server (also called registrar, proxy or SIP host), its port if it is not the standard one, the username, the password, and the auth username when the provider lists one separately. In SIPdia those go in Server, Transport, Username, Password and Auth username. Then choose TLS with Encrypt audio set to Required if the provider supports it, add a STUN server and turn on ICE if you are not on a VPN, and clear the PBX feature codes that a provider does not understand. The steps below are the same on macOS and on iPhone and iPad; only the button name differs.

Step 1: Find the SIP settings in the provider's portal

Every provider shows the details somewhere different, but the fields have the same meaning. Look for a page called SIP settings, device settings, softphone setup or credentials. You need:

• Server. May be labelled registrar, SIP server, proxy, SIP host or domain. It is a host name such as sip.example.net or an IP address.
• Port. Most providers use 5060 for UDP and TCP and 5061 for TLS. Some use another port to avoid scanners, for example 5080. Note it if it is not the default.
• Transport. UDP, TCP or TLS. Many providers list all three; some support only UDP.
• Username. Often the phone number, sometimes an account ID.
• Password. Sometimes called the SIP password or secret. This is not your portal login.
• Auth username. Only some providers have one. It appears as authentication ID, auth user or authorization name.
• Outbound proxy and Domain, when listed separately from the server.

Some portals also show a voicemail access number and a codec list. Note both.

Step 2: Enter the account in SIPdia

1. Sign in to SIPdia and open the Account page (Mac) or Account tab (iPhone and iPad). If you already have an account, click Add Account (Mac) or tap Add (iPhone).
2. In Server, enter the host from the portal. Add :port only if the provider uses a non-standard port, for example sip.example.net:5080. Without a port SIPdia looks up DNS SRV records as defined in RFC 3263 and falls back to the default port.
3. Set Transport to what the provider supports. Use TLS when offered (see step 3).
4. Enter Username and Password exactly as the portal shows them.
5. Set Display name to the name you want shown to people you call. Many providers replace it with your registered number, so it may not always appear.
6. Open Advanced. Fill Auth username only if the portal lists one. Fill Domain only if the SIP domain in your address differs from the server host. Fill Outbound proxy only if the provider tells you to send traffic through a proxy that is different from the registrar.
7. Click Save and Register (Mac) or tap Add and Register (iPhone and iPad). The dot turns green when the provider accepts the registration.
8. Allow the microphone, local network and notification prompts.

Step 3: Choose encryption

If the provider supports it, set Transport to TLS and Encrypt audio to Required. TLS encrypts the SIP signalling, including your password exchange; SRTP (RFC 3711) encrypts the audio. With Required, SIPdia refuses to set up a call without SRTP, so you never fall back to clear audio by accident. Choose Optional only when the provider supports SRTP on some routes but not all.

Two conditions apply. The provider's TLS certificate must be signed by a public certificate authority; self-signed certificates are rejected. And TLS uses TCP port 5061 unless the Server field names another port. Some office and guest networks block 5061, so if TLS never connects, ask for the port to be opened or use the alternative TLS port the provider offers. Details are on the TLS and SRTP feature page.

Step 4: Handle NAT with STUN and ICE

Provider servers are on the public internet, while your Mac or iPhone is almost always behind a router or a carrier NAT. SIP messages carry your private address inside them, so the provider may send audio to an address it cannot reach. The symptom is a call that connects but has one-way or no audio.

Under Advanced:

• Set STUN server to a public STUN server, for example stun.l.google.com:19302. STUN (RFC 5389) tells SIPdia what its public address and port look like from outside.
• Turn on ICE (RFC 8445). ICE gathers several candidate addresses and checks connectivity with the provider's media server, then picks the path that works.

Skip both when you connect through a VPN that gives you a reachable address. If audio problems persist, work through One-way audio on VoIP calls.

Step 5: Check the codecs

SIPdia offers Opus, G.711 ulaw (PCMU) and G.711 alaw (PCMA). The codec for each call is negotiated with the provider. Most providers accept ulaw (common in North America) or alaw (common in Europe and elsewhere), so a call normally works without changes. If the provider lists Opus, calls may use it for better quality on poor connections. G.722 and G.729 are not offered, so if the provider only allows G.729, ask them to enable G.711 for your account.

Step 6: Clear feature codes and set voicemail

SIPdia's Settings > Feature Codes holds star codes such as *72 for call forwarding and *78 for Do Not Disturb. These are PBX features. Most providers do not have them, and dialling them would place an ordinary call to a number that does not exist.

1. Open Settings > Feature Codes for the provider account.
2. Clear the forwarding and Do Not Disturb codes.
3. Set the voicemail code to the provider's voicemail access number, so the voicemail button in SIPdia dials the right place.
4. Configure forwarding and Do Not Disturb in the provider's portal instead.

Step 7: Make a test call

Call a number you can answer, such as your mobile. Check that you hear audio in both directions, that keypad tones work on an IVR, and that your caller ID appears as expected. Then call your SIPdia number from the mobile to confirm incoming calls ring. Remember that SIPdia must be running (Mac) or open (iPhone) to receive calls. On iPhone the app unregisters about 20 seconds after leaving the foreground, so set forward-on-no-answer in the provider's portal to a number that is always reachable.

Next steps

• Not sure what a SIP account includes? Read What is a SIP account?.
• Compare this with registering to a PBX in SIP Registration: PBX Extension vs Provider Account.
• Need the standard port numbers? See the SIP port reference.