How to Use SIPdia with a SIP Trunk or Hosted SIP Provider

With a hosted SIP provider, SIPdia registers as a SIP user account, the same way a desk phone would. Open the provider's portal, copy the server, username and password into SIPdia's Account page, and register. Fill in Domain, Outbound proxy and Auth username only when the provider lists them as separate values. If the provider supports TLS, use it and set Encrypt audio to Required. If you hear one-way audio from home or a hotel, add a STUN server and turn on ICE.

A softphone is a user, not a trunk

Providers use the word trunk for the link between their network and a customer's PBX. A trunk often carries many calls at once and may be authenticated by the PBX's IP address rather than a password. SIPdia is a softphone, so it needs a single SIP user account (some portals call it a device, a line, an endpoint or a SIP credential). If your provider only sells trunks, ask whether a trunk can be used with registration from a softphone, or create a user under the trunk if the portal allows it. The SIP standard itself (RFC 3261) makes no difference between the two; it is the provider's account model that does.

What to collect from the provider's portal

Look for a page named SIP settings, device credentials or manual configuration. You need:

Provider value	Typical names in portals	SIPdia field
SIP server	Registrar, proxy, SIP host, server address	Server
Port	Only if it is not 5060 (UDP/TCP) or 5061 (TLS)	Added to Server as host:port
Transport	UDP, TCP or TLS	Transport
Username	SIP username, user ID, device ID, line number	Username
Password	SIP password, secret	Password
Auth username	Authentication ID, auth user	Auth username (Advanced), only if different
SIP domain	Domain, realm	Domain (Advanced), only if different from the server
Outbound proxy	Outbound proxy, edge server, SBC	Outbound proxy (Advanced), only if listed
Codecs	Supported codecs	No field; SIPdia offers Opus, ulaw and alaw
Voicemail number	Voicemail access number	Settings > Feature Codes > Voicemail

Many providers publish a generic setup page for "other SIP clients". Those values are the ones to use.

Enter the account in SIPdia

1. Open SIPdia, sign in, and go to Account.
2. Server: enter the SIP server. If the provider gives a port other than the default, add it as sip.example.com:5080. If you leave the port out, SIPdia performs a DNS SRV lookup (RFC 3263) and falls back to the default port.
3. Transport: pick the transport the provider documents. Prefer TLS where available.
4. Username and Password: copy them exactly. Passwords from portals often end with a space when copied; check for one.
5. Display name: your name, or the name the provider expects. Some providers ignore it and use the account's caller ID instead.
6. Open the Advanced section:
   • Domain: fill in only if the provider lists a SIP domain or realm that differs from the server address. Example: server proxy.example.com, domain example.com.
   • Outbound proxy: fill in only if the provider lists one. SIPdia then sends every request to the proxy while keeping the domain in the SIP addresses.
   • Auth username: fill in only if the provider shows an authentication ID that differs from the username.
7. Click Save and Register (Mac) or Add and Register (iPhone).

Turn on encryption

If the provider offers TLS, set Transport to TLS and Encrypt audio to Required. TLS protects the signalling, including your password exchange, and SRTP (RFC 3711) protects the audio. With Required, SIPdia refuses to set up a call without SRTP, so you know every call is encrypted. If the provider supports SRTP on some routes only, choose Optional instead. The provider's certificate must be signed by a public certificate authority; SIPdia rejects self-signed certificates.

TLS uses TCP port 5061 unless the server address names another port. Some office and guest networks block 5061. If registration over TLS times out but UDP works, ask for the port to be opened or use the provider's alternative TLS port.

Fix NAT with STUN and ICE

At home, in a hotel or on mobile data, the device sits behind a router that rewrites addresses. The provider may then send audio to an address that cannot be reached, and you hear nothing or only one side. Two settings in the Advanced section fix most of this:

1. STUN server: enter a STUN server, for example stun.l.google.com:19302. SIPdia asks it for the public address and port (RFC 5389) and places them in its SIP and SDP messages.
2. ICE: turn it on. SIPdia then offers every candidate address and lets both sides test which path actually works (RFC 8445).

Many providers run their own session border controllers that handle NAT without help. Start without STUN and ICE; add them if you hear one-way audio. On a corporate VPN these settings are usually unnecessary.

Codecs

SIPdia offers Opus, G.711 ulaw (PCMU) and G.711 alaw (PCMA). It does not offer G.722 or G.729. Almost every provider accepts ulaw or alaw, and some accept Opus. If your provider lists G.729 only, it will not work with SIPdia; ask whether G.711 can be enabled on the account.

Feature codes and voicemail

Feature codes such as *72 are a PBX convention. Most providers do not implement them, and dialling them may place a real call. Go to Settings > Feature Codes and clear the forwarding and Do Not Disturb codes for the provider account. Set forwarding in the provider's portal instead. Then set the voicemail code to the provider's voicemail access number so the voicemail button in SIPdia still works. A few providers do support star codes; if yours documents them, enter those instead of clearing the fields.

Verify registration

The dot next to the account turns green when the provider answers the REGISTER request with 200 OK. Most provider portals also show the registration status of each device, often with the IP address and the user agent string. If the dot stays red:

• A 401 or 403 error means the credentials or the domain are wrong. Check Username, Password, Auth username and Domain against the portal. See SIP 401 Unauthorized.
• "Could not reach the server" means no answer came back. Check the server address, port and transport. Try UDP if TLS times out, to find out whether the network blocks 5061.
• Turn on Settings > General > Save detailed logs (Mac) or Settings > Save detailed logs (iPhone) and read the SIP messages. The logs include your SIP traffic, so turn the setting off afterwards.

Place a test call

1. On the Phone page, dial the provider's echo test or voicemail number if it has one, or your own mobile number.
2. Speak and listen. You should hear yourself (echo test) or the call should ring your mobile with the caller ID the provider assigned.
3. Call your SIP number from another phone. On the Mac the floating incoming-call window appears; on iPhone the call shows on the system call screen.
4. During the call, press a few digits. If an IVR on the other end reacts, keypad tones (RFC 4733) work.

Troubleshooting

• Registration fails after a while. Some providers expect registration every few minutes and drop silent clients behind NAT. Setting a STUN server often helps. See SIP registration keeps failing.
• Outgoing calls fail with 403 or 404. Check the number format. Many providers expect E.164 with a leading plus or a country code without the plus. The portal's dialling rules page says which.
• One-way audio. Add STUN server and ICE. If the problem persists, read one-way audio in VoIP.
• Calls stop after 30 seconds. A classic NAT symptom: the provider's ACK or the audio path breaks. STUN, ICE or TLS usually fixes it. Ask the provider whether their SBC handles far-end NAT traversal.

Next steps

• TLS and SRTP explains what the encryption settings protect.
• Multiple SIP accounts shows how to run a provider account and a PBX extension side by side.
• Common SIP ports lists the defaults to check when registration times out.