# SIPdia

> SIPdia is a SIP softphone for Mac, iPhone and iPad. Register SIP accounts from your PBX or provider, call over TLS and SRTP, hold, transfer and record calls.

SIPdia is made by SpaGreen Creative. Official website: https://sipdia.com/

## Facts

- Platforms with an app: macOS (macOS 14 (Sonoma) or later); iPhone and iPad (iOS 17 / iPadOS 17 or later).
- Planned, no app yet: Windows, Linux, Android. No release dates are announced.
- Release status: both apps are in release preparation; no public download or store links exist yet (as of 2026-11-09).
- Features: Multiple SIP accounts (Premium); TLS and SRTP; Call transfer; Call hold and several calls; Call recording (Premium); Call forwarding and Do Not Disturb (Premium); Contacts with photos; STUN and ICE; Opus and G.711 codecs; CallKit integration; Menu bar and floating call window.
- Not implemented: Push notifications for incoming calls.
- Freemium: the free plan includes one SIP account. SIPdia Premium adds: More than one SIP account, all registered at once; Call recording to WAV on your device; Call forwarding and Do Not Disturb through PBX feature codes; Import and export contacts: Contacts app, vCard and CSV. Prices are not published yet.
- Call engine: baresip 4.12 (BSD 3-Clause) with Opus and OpenSSL.

## Key pages

- [Download](https://sipdia.com/download/): release status; download links appear here when published
- [Features](https://sipdia.com/features/): verified feature list with platform support
- [Pricing](https://sipdia.com/pricing/): free plan and SIPdia Premium
- [Security](https://sipdia.com/security/): TLS, SRTP, what stays on the device
- [Privacy policy](https://sipdia.com/privacy/)
- [SIP softphone for macOS](https://sipdia.com/sip-softphone-for-macos/)
- [SIP softphone for iPhone and iPad](https://sipdia.com/sip-softphone-for-iphone/)

## Blog

Articles about SIP calling, softphones and SIPdia. Index: https://sipdia.com/blog/

- [Free vs Paid SIP Softphones: What Should You Consider?](https://sipdia.com/blog/free-vs-paid-sip-softphones/): How freemium softphones split features, what is usually behind the paywall, open source versus proprietary trade-offs, support expectations and update cadence, with SIPdia's own free and Premium split.
- [Introducing SIPdia: A SIP Softphone for Mac, iPhone and iPad](https://sipdia.com/blog/introducing-sipdia/): SIPdia is a SIP softphone for macOS 14+ and iOS/iPadOS 17+. Here is what it does, what is free and what is in Premium, what is not there yet, and where the release stands.
- [What to Look for in a SIP Softphone](https://sipdia.com/blog/what-to-look-for-in-a-sip-softphone/): A practical checklist for choosing a SIP softphone: platforms, transports, SRTP, codecs, multiple accounts, transfer and hold, recording, NAT traversal, push notifications, licensing and privacy.
- [Why Some Softphones Must Stay Open to Receive Calls](https://sipdia.com/blog/why-a-softphone-must-stay-open-to-receive-calls/): How SIP registration and incoming INVITEs work, why iOS and macOS limit background apps, what VoIP push and PushKit do, why a push gateway is needed, and how forward-on-no-answer helps.

## Guides

SIP fundamentals explained in plain language. Index: https://sipdia.com/guides/

- [Common SIP Ports and What They Do](https://sipdia.com/guides/common-sip-ports/): SIP uses port 5060 for UDP and TCP and 5061 for TLS. RTP audio uses a range of UDP ports, and STUN uses 3478. Learn what each port carries, how to find the ports your PBX uses, and how to open them safely.
- [Common SIP Security Risks and How to Reduce Them](https://sipdia.com/guides/common-sip-security-risks/): An overview of the most common SIP threats: scanning, brute force, toll fraud, eavesdropping, spoofed caller ID, registration hijacking, SIP ALG problems and weak PBX defaults, with practical mitigations.
- [How to Secure a SIP Account](https://sipdia.com/guides/how-to-secure-a-sip-account/): Practical steps to protect a SIP account: strong unique secrets, a separate auth username, TLS and SRTP, IP allow-lists, VPN access, credential rotation and toll-fraud monitoring.
- [How to Protect a PBX from SIP Scanning and Brute-Force Attempts](https://sipdia.com/guides/protect-pbx-from-sip-scanning/): Keep SIP port 5060 off the public internet, use VPN or allow-lists, add fail2ban, set strong secrets, restrict international dialing, watch CDRs and keep the PBX updated.
- [Understanding RTP and Audio Streams in VoIP Calls](https://sipdia.com/guides/rtp-and-audio-streams/): RTP carries the voice in a SIP call. Learn the RTP packet structure, how ports are negotiated in SDP, what RTCP reports, how jitter buffers work, how DTMF travels as RTP events, and why one-way audio happens.
- [SIP Client vs VoIP Phone: What Is the Difference?](https://sipdia.com/guides/sip-client-vs-voip-phone/): A SIP client is software or hardware that speaks the SIP protocol. A VoIP phone is any device that makes calls over IP. Learn how the terms overlap, where they differ, and which fits your setup.
- [SIP Codecs Explained: G.711, Opus and Other Common Options](https://sipdia.com/guides/sip-codecs-explained/): A codec turns voice into data and back. Learn how G.711, Opus, G.722, G.729 and GSM compare in bandwidth, quality and licensing, how codecs are negotiated in SDP, and how to choose one for your PBX.
- [SIP over TLS vs SRTP: What Each One Protects](https://sipdia.com/guides/sip-over-tls-vs-srtp/): TLS encrypts SIP signaling hop by hop; SRTP encrypts the audio stream. Learn why you need both, how SDES and DTLS-SRTP exchange keys, and what SIPdia offers.
- [SIP over UDP, TCP and TLS: Key Differences](https://sipdia.com/guides/sip-over-udp-tcp-tls/): SIP can run over UDP, TCP or TLS. Learn how each transport affects reliability, NAT traversal, message size and privacy, which ports they use, and how to pick the right one for your PBX or provider.
- [SIP Registration Explained](https://sipdia.com/guides/sip-registration-explained/): SIP registration is how a softphone tells the server where to reach it. Learn the REGISTER request, the 401 challenge, expiry and refresh, the Contact header, and why registration breaks behind NAT.
- [What Is a SIP Account? Username, Auth ID, Domain and Password](https://sipdia.com/guides/what-is-a-sip-account/): A SIP account is the set of credentials a softphone uses to register with a SIP server. Learn what the username, auth ID, domain, password, server and outbound proxy fields mean and where each one comes from.
- [What Is a SIP Softphone? A Practical Guide](https://sipdia.com/guides/what-is-a-sip-softphone/): A SIP softphone is an app that places and receives phone calls over the internet using the SIP protocol. Learn how it works, what you need to use one, and how it compares to a desk phone.
- [What Is SIP and How Does SIP Calling Work?](https://sipdia.com/guides/what-is-sip/): SIP is the signalling protocol that sets up, changes and ends voice calls over IP networks. Learn the main SIP messages, how a call flows from INVITE to BYE, and how SIP works with SDP and RTP.
- [What Is SRTP and How Does It Protect Voice Traffic?](https://sipdia.com/guides/what-is-srtp/): SRTP is the Secure Real-time Transport Protocol. It encrypts and authenticates the RTP audio packets of a VoIP call. Learn how it works, how keys are exchanged with SDES or DTLS, and why it needs TLS signalling.

## Tutorials

Step-by-step setup for SIPdia, PBXs and providers. Index: https://sipdia.com/tutorials/

- [How to Add Multiple SIP Accounts to a Softphone](https://sipdia.com/tutorials/add-multiple-sip-accounts/): Add a second SIP account in SIPdia on Mac or iPhone, choose which account to call from with the From picker, keep separate feature codes per account, and see which account each call came in on.
- [How to Configure a SIP Extension in FreePBX](https://sipdia.com/tutorials/configure-sip-extension-in-freepbx/): For FreePBX administrators: add a PJSIP extension under Applications > Extensions, set the secret and Advanced options, check the port and codecs under Asterisk SIP Settings, apply the config, and confirm a phone registers.
- [How to Configure a SIP Extension in Issabel](https://sipdia.com/tutorials/configure-sip-extension-in-issabel/): Step-by-step for Issabel administrators: add an extension under PBX > PBX Configuration > Extensions, set the secret, codecs, DTMF and NAT options, apply the configuration, and check that a phone registers.
- [How to Connect a Softphone to an Asterisk PBX](https://sipdia.com/tutorials/connect-softphone-to-asterisk/): Generic steps to register any SIP softphone with Asterisk: collect the extension details, map them to the softphone's fields, fix NAT and codecs, verify registration, then the exact fields for SIPdia on macOS and iPhone.
- [How to Connect a Softphone to an ISP or Provider SIP Account](https://sipdia.com/tutorials/connect-softphone-to-isp-sip-trunk/): Find the SIP settings in your provider's portal, map registrar, port, username, password and auth username to SIPdia's fields, pick TLS and SRTP, add STUN and ICE for NAT, and clear PBX feature codes.
- [How SIP Extensions Work in Asterisk](https://sipdia.com/tutorials/how-sip-extensions-work-in-asterisk/): Learn how an Asterisk PJSIP extension is built from an endpoint, an auth and an AOR, how a context ties it to the dialplan, and what happens on the wire when a phone registers and calls.
- [How SIP Trunks Work](https://sipdia.com/tutorials/how-sip-trunks-work/): A SIP trunk connects a PBX to the telephone network over the internet. Learn how trunks differ from extensions, registration versus IP authentication, DIDs, codecs, channels, and where a softphone fits in.
- [SIP Registration: PBX Extension vs Provider Account](https://sipdia.com/tutorials/pbx-extension-vs-provider-account/): A PBX extension and a provider SIP account register the same way but differ in where credentials come from, feature codes, NAT handling and security exposure. Here is how each maps to a softphone's fields.
- [How to Set Up a SIP Softphone on iPhone and iPad with SIPdia](https://sipdia.com/tutorials/set-up-sip-softphone-on-iphone/): Register a PBX extension or provider SIP account in SIPdia for iPhone and iPad: fill the Account tab, tap Add and Register, allow the permissions and answer calls on the system call screen.
- [How to Set Up a SIP Softphone on macOS with SIPdia](https://sipdia.com/tutorials/set-up-sip-softphone-on-macos/): Register a PBX extension or provider SIP account in SIPdia for Mac: sign in, fill the Account page, click Save and Register, allow the permission prompts and make a first call.

## Troubleshooting

Fix registration, audio, NAT and codec problems. Index: https://sipdia.com/troubleshooting/

- [How NAT Affects SIP and RTP](https://sipdia.com/troubleshooting/nat-and-sip/): SIP writes IP addresses inside its messages, so network address translation breaks it. This article explains rport, symmetric RTP, STUN, ICE, TURN, keep-alives and SIP outbound, and what a softphone can and cannot do about NAT on its own.
- [How to Troubleshoot No Audio in a SIP Call](https://sipdia.com/troubleshooting/no-audio-sip-call/): When a SIP call connects but both sides hear silence, the usual causes are a codec mismatch, RTP blocked in both directions, a missing microphone permission, the wrong audio device, mute, an SRTP mismatch or a Bluetooth route that captured the audio. Work through them in order.
- [How to Fix One-Way Audio in VoIP Calls](https://sipdia.com/troubleshooting/one-way-audio-voip/): One-way audio means the call connected but RTP flows in only one direction, almost always because a private IP address ended up in the SDP. Fix it with STUN and ICE on the phone, NAT settings on the PBX, symmetric RTP and an open RTP port range.
- [Why Can I Register but Not Make Calls?](https://sipdia.com/troubleshooting/register-but-cannot-make-calls/): Registration proves your credentials; calling tests the dialplan, outbound routes, dial format, codecs, trunks and caller ID. Learn to read 404, 403, 488 and 503 replies and find the setting that blocks your outbound call.
- [How to Fix SIP 401 Unauthorized](https://sipdia.com/troubleshooting/sip-401-unauthorized/): A single SIP 401 Unauthorized is normal digest authentication. Repeated 401 replies mean the password, auth username or realm is wrong, or the account is locked. Here is how to tell the two apart and fix it.
- [How to Fix SIP 403 Forbidden](https://sipdia.com/troubleshooting/sip-403-forbidden/): SIP 403 Forbidden means the server understood the request but refuses it. Causes range from wrong credentials and disabled extensions to IP bans, fail2ban, wrong domains and outbound calls the dialplan does not permit.
- [How to Fix SIP 408 Request Timeout](https://sipdia.com/troubleshooting/sip-408-request-timeout/): A SIP 408 or a plain registration timeout means the server never answered. Check the address, port and transport, UDP and TLS port blocks, SIP ALG, DNS SRV records, the server itself and the Local Network permission on Apple devices.
- [What Is SIP ALG and Should You Disable It?](https://sipdia.com/troubleshooting/sip-alg/): SIP ALG is a router feature that rewrites addresses inside SIP packets to help with NAT. It often does more harm than good, causing dropped registrations, one-way audio and missed calls. Here is how to recognise it, why it breaks, and how to turn it off.
- [Why Does SIP Registration Keep Failing?](https://sipdia.com/troubleshooting/sip-registration-keeps-failing/): A systematic checklist for SIP registration failures: credentials, transport and port, DNS, NAT keep-alives, registration expiry, TLS certificate validity, contact limits, IP bans, Wi-Fi versus mobile networks and IPv6.

## Glossary

SIP and VoIP terms, defined. Index: https://sipdia.com/glossary/

- [Attended Transfer](https://sipdia.com/glossary/attended-transfer/): An attended transfer lets you speak to the destination before connecting the caller, then joins the two calls. In SIP it uses REFER with a Replaces header.
- [Auth Username](https://sipdia.com/glossary/auth-username/): The auth username is the identity a SIP client presents during digest authentication. It can differ from the extension number, which improves security on PBXs that support it.
- [Blind Transfer](https://sipdia.com/glossary/blind-transfer/): A blind transfer sends an active call to another number immediately, without the transferring party speaking to the destination first. In SIP it is done with a REFER request.
- [Codec (Audio Codec)](https://sipdia.com/glossary/codec/): A codec encodes audio into a compact digital form for transmission and decodes it at the far end. Both ends of a SIP call must share at least one codec.
- [DID (Direct Inward Dialing)](https://sipdia.com/glossary/did/): A DID is a public phone number that routes directly to a specific extension or destination on a PBX, without going through an operator or auto attendant.
- [DNS SRV Record](https://sipdia.com/glossary/dns-srv/): A DNS SRV record tells a SIP client which host, port and transport to use for a domain. It lets providers change servers without reconfiguring every client.
- [DTMF (Dual-Tone Multi-Frequency)](https://sipdia.com/glossary/dtmf/): DTMF is the tone signaling produced by pressing keys on a phone keypad. In VoIP it can be sent as audio, as RTP events or inside SIP messages, and the methods must match.
- [Extension](https://sipdia.com/glossary/extension/): An extension is an internal number on a PBX assigned to a person or device. It serves as the SIP username when a softphone registers with the PBX.
- [Feature Code](https://sipdia.com/glossary/feature-code/): A feature code is a short dial sequence, often starting with a star, that tells the PBX to perform an action such as enabling call forwarding, Do Not Disturb or checking voicemail.
- [G.711 Codec](https://sipdia.com/glossary/g711/): G.711 is the most widely supported telephone audio codec, available as ulaw (PCMU) and alaw (PCMA). It uses 64 kbit/s and is a common baseline on SIP devices.
- [ICE (Interactive Connectivity Establishment)](https://sipdia.com/glossary/ice/): ICE gathers all possible addresses for a call, tests them in pairs and picks the best working path, combining STUN and TURN to make media work through NAT.
- [Jitter (VoIP)](https://sipdia.com/glossary/jitter/): Jitter is the variation in arrival time between consecutive RTP packets. Too much jitter causes choppy or robotic audio; a jitter buffer trades delay for smoothness.
- [NAT (Network Address Translation)](https://sipdia.com/glossary/nat/): NAT lets many devices share one public IP address by rewriting packet addresses at the router. It breaks SIP and RTP because they carry addresses inside their messages.
- [One-Way Audio](https://sipdia.com/glossary/one-way-audio/): One-way audio is a call in which only one party can hear the other. It is almost always caused by NAT or firewall rules blocking RTP in one direction, not by a codec problem.
- [Opus Codec](https://sipdia.com/glossary/opus/): Opus is a modern, royalty-free audio codec that adapts its bitrate and bandwidth to network conditions. It delivers wideband voice quality and handles packet loss well.
- [Outbound Proxy](https://sipdia.com/glossary/outbound-proxy/): An outbound proxy is the server a SIP client sends all its requests to, regardless of the destination domain. Learn when it is needed and how it is configured.
- [Packet Loss](https://sipdia.com/glossary/packet-loss/): Packet loss is the fraction of RTP packets that never reach the receiver. Even a few percent degrades voice quality; codecs and concealment reduce the effect but cannot remove it.
- [PBX (Private Branch Exchange)](https://sipdia.com/glossary/pbx/): A PBX is the telephone system inside an organisation. An IP PBX uses SIP to connect extensions, route calls and link to the outside world through trunks.
- [Realm (SIP Authentication)](https://sipdia.com/glossary/realm/): The realm is the protection domain named in a SIP authentication challenge. The client includes it when hashing the password, so it must match what the server expects.
- [Registrar](https://sipdia.com/glossary/registrar/): A SIP registrar accepts REGISTER requests and stores where each user can currently be reached, so incoming calls can be routed to the right device.
- [RFC 2833 / RTP Events](https://sipdia.com/glossary/rfc2833/): RFC 2833, now superseded by RFC 4733, defines how DTMF digits are sent as dedicated RTP event packets rather than as audio tones. It is the default DTMF method for most SIP devices.
- [RTCP (RTP Control Protocol)](https://sipdia.com/glossary/rtcp/): RTCP is the companion protocol to RTP. It carries periodic reports on packet loss, jitter and timing so endpoints can monitor and adapt call quality.
- [RTP (Real-time Transport Protocol)](https://sipdia.com/glossary/rtp/): RTP carries the actual audio of a VoIP call in small timestamped UDP packets. Learn what it does, how it relates to SIP and why NAT often breaks it.
- [SDP (Session Description Protocol)](https://sipdia.com/glossary/sdp/): SDP is the format inside SIP messages that describes the media of a call: codecs, IP addresses, ports and encryption parameters. Learn what it contains.
- [SIP (Session Initiation Protocol)](https://sipdia.com/glossary/sip/): SIP is the signaling protocol that sets up, changes and ends real-time sessions such as voice calls. Learn what SIP does and what it leaves to other protocols.
- [SIP ALG (Application Layer Gateway)](https://sipdia.com/glossary/sip-alg/): A SIP ALG is a router feature that rewrites SIP packets to help them cross NAT. In practice it often corrupts messages and breaks calls, so most PBX vendors advise turning it off.
- [SIP Proxy](https://sipdia.com/glossary/sip-proxy/): A SIP proxy receives SIP requests and forwards them toward their destination, making routing decisions and often enforcing authentication along the way.
- [SIP Trunk](https://sipdia.com/glossary/sip-trunk/): A SIP trunk connects a PBX to the public telephone network over the internet, replacing traditional phone lines. Learn how it carries calls and how it is configured.
- [SIP URI (SIP Address)](https://sipdia.com/glossary/sip-uri/): A SIP URI is the address of a SIP user or service, in the form sip:user@domain, optionally with a port and parameters. Learn how it is built and used.
- [Softphone](https://sipdia.com/glossary/softphone/): A softphone is an application that makes and receives phone calls over the internet using a computer, phone or tablet instead of a dedicated handset. Most use SIP.
- [SRTP (Secure Real-time Transport Protocol)](https://sipdia.com/glossary/srtp/): SRTP encrypts and authenticates RTP audio so it cannot be listened to or altered in transit. Learn how keys are exchanged and why it should be paired with TLS.
- [STUN (Session Traversal Utilities for NAT)](https://sipdia.com/glossary/stun/): STUN lets a device behind NAT discover its public IP address and port by asking an external server, so it can advertise a reachable address in SIP and SDP.
- [TLS (Transport Layer Security)](https://sipdia.com/glossary/tls/): TLS encrypts and authenticates the connection between a SIP client and server, protecting signaling from eavesdropping and tampering. It is the secure transport option in SIP.
- [TURN (Traversal Using Relays around NAT)](https://sipdia.com/glossary/turn/): TURN is a relay protocol used when direct media between two endpoints is impossible. A TURN server forwards RTP on behalf of a client behind restrictive NAT or firewalls.
- [User Agent](https://sipdia.com/glossary/user-agent/): In SIP, a user agent is any endpoint that originates or receives requests: a softphone, desk phone or gateway. It acts as client and server at the same time.

## Integrations

Configure SIPdia with popular PBX systems and SIP trunks. Index: https://sipdia.com/integrations/

- [How to Configure SIPdia with Asterisk (PJSIP)](https://sipdia.com/integrations/asterisk/): A minimal pjsip.conf endpoint, auth and AOR for SIPdia on Asterisk 18 and 20, with UDP and TLS transports, RTP port range, SRTP, and the CLI commands to verify registration.
- [How to Configure SIPdia with FreePBX](https://sipdia.com/integrations/freepbx/): Add a PJSIP extension in FreePBX, find the SIP port and codecs under Asterisk SIP Settings, open the firewall, then register SIPdia on macOS or iPhone with the extension number and secret.
- [How to Configure SIPdia with Issabel](https://sipdia.com/integrations/issabel/): Register SIPdia on a Mac or iPhone as an Issabel extension. Create the extension, set codecs and NAT, open the SIP and RTP ports, then enter the account in the app and verify it.
- [How to Use SIPdia with a SIP Trunk or Hosted SIP Provider](https://sipdia.com/integrations/sip-trunk/): Register SIPdia on macOS or iPhone with a hosted SIP provider. Collect the server, username and password, set Domain and Outbound proxy when listed, turn on TLS and SRTP, and fix NAT with STUN and ICE.

## Compare

Fair, sourced comparisons of SIPdia and other softphones. Index: https://sipdia.com/compare/

- [SIPdia vs Bria: A Fair Comparison](https://sipdia.com/compare/sipdia-vs-bria/): A sourced comparison of SIPdia and Bria Solo: platforms, subscription pricing, transports, SRTP, codecs, account limits, transfer, recording and the Bria Push Service, checked on 8 November 2026.
- [SIPdia vs Linphone: A Fair Comparison](https://sipdia.com/compare/sipdia-vs-linphone/): A sourced comparison of SIPdia and Linphone: platforms, open source licence, transports, SRTP and ZRTP, codecs, transfer, recording and push notification limits, checked on 8 November 2026.
- [SIPdia vs MicroSIP: A Fair Comparison](https://sipdia.com/compare/sipdia-vs-microsip/): A sourced comparison of SIPdia and MicroSIP: platforms, GPL licence, transports, SRTP, codecs, account handling, transfer, hold and recording, checked against the official MicroSIP site on 8 November 2026.
- [SIPdia vs Zoiper: A Fair Comparison](https://sipdia.com/compare/sipdia-vs-zoiper/): A sourced, side-by-side comparison of SIPdia and Zoiper: platforms, pricing model, transports, encryption, codecs, transfer, recording and push notifications, checked on 8 November 2026.

## Tools

- [SIP URI generator](https://sipdia.com/tools/sip-uri-generator/)
- [SIP response code lookup](https://sipdia.com/tools/sip-error-code-lookup/)
- [VoIP bandwidth calculator](https://sipdia.com/tools/voip-bandwidth-calculator/)
- [SIP port reference](https://sipdia.com/tools/sip-port-reference/)