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Roadmap

Where Sipral goes next.

1.0 is complete. Every 1.x release adds to it and keeps every application built against 1.0 working, without a rebuild.

Every release at a glance

  1. October 2026 1.0 Released
  2. October 2026 1.1 Released
  3. October 2026 1.2 Released
  4. November 2026 1.3 Next
  5. December 2026 1.4 Planned
  6. March 2027 1.5 Planned
  7. Mid 2027 2.0 Planned

01

Coming next

  1. November 2026 · 1.3 (planned)

    On real phones, and on WebRTC lines

    The iOS and Android packages proven on real phones and through the test tracks of both stores, and calls on the WebRTC lines PBXs offer: one port for the whole call, in the profile browsers speak.

    • iOS and Android proven on real phones: the system call screen, a cellular call interrupting, earpiece, speaker, wired and Bluetooth headsets, the car
    • An incoming call that wakes a sleeping phone, end to end through a push gateway (RFC 8599) In progress
    • Wi-Fi to mobile data on a phone, the time until audio returns measured and published
    • A registered phone with no call: battery and background behaviour measured on both platforms
    • Test builds accepted through TestFlight and Google Play internal testing In progress
    • A cold start after a push in Swift and Kotlin, as in .NET MAUI: the woken app resumes its account rather than registering from scratch Done
    • Android transactional Telecom API (API 34 and later) beside the self-managed ConnectionService Done
    • Calls on a PBX WebRTC line: BUNDLE (RFC 8843) and the UDP/TLS/RTP/SAVPF profile, proven against Asterisk and FreeSWITCH Done

  2. December 2026 · 1.4 (planned)

    Browsers and the telephone network

    Trickle ICE, an audio call to and from a browser, SIP-T and SIP-I through a carrier gateway, and C, C++ and Rust applications that no longer write their own networking.

    • Trickle ICE (RFC 8838, RFC 8840): candidates sent as they are found, so a call is set up sooner Done
    • An audio call to and from a browser over WebRTC, through a SIP over WebSocket server, proven in the lab
    • SIP-T and SIP-I: the ISUP a carrier gateway carries inside SIP (RFC 3204, RFC 3372) read and written Done
    • Calling number, redirection and release cause taken from the ISUP when a gateway sends them Done
    • SIP-T and SIP-I calls proven against a carrier-style gateway in the lab Done
    • Networking ready to use from Rust and C: sockets, TLS and NAPTR/SRV lookups in one optional component, so an application without a binding no longer writes its own
    • A C++ API over the C ABI: accounts, calls and events as classes, their lifetimes handled for the caller
    • A hook that reads and rewrites any SIP message just before it is sent and just after it arrives, from Rust, C and every binding

02

Further out, and in every release

  1. March 2027 · 1.5 (planned)

    Keys that never leave the device

    A client certificate on SIP over TLS, the private key held and used by the platform security hardware, on phones and on the desktop alike.

    • A client certificate on SIP over TLS, signed by the platform: Secure Enclave on iOS, Android Keystore, the Keychain on macOS, CNG on Windows
    • The same certificate in the C ABI and in the Swift, Kotlin and .NET bindings
    • Proven against PBXs and proxies that require a client certificate, in the lab

  2. Mid 2027 · 2.0 (planned)

    Beyond audio

    Video with VP8, VP9 and AV1, and ZRTP end-to-end key agreement.

    • Video with VP8, VP9 and AV1, with RTCP feedback for pictures
    • ZRTP end-to-end key agreement

  3. Every release · Ongoing (planned)

    Proven, and kept proven

    What no single release finishes: the cryptography reviewed, more PBXs and carriers proven, and every published figure measured again.

    • Every published figure measured again by the gate on each change In progress
    • An adversarial review of the DTLS-SRTP code, with interop and failure tests against other implementations Done
    • More PBXs proven in the lab, and caller identity, diversion, redirection and failover between servers shown working In progress
    • Interop with carriers and a commercial SBC

03

Released

  1. October 2026 · 1.2 (released)

    Voice agents, the enterprise and more platforms

    Connectors for the services voice agents run on, OAuth2 sign-in to the PBX, Node.js and .NET MAUI packages, a network test before the call, answering-machine detection, SIP over WebSocket and LiveCommunicationKit on iOS.

    What it delivered, 18 items
    • A SIP bridge from a PBX line to any voice agent that answers SIP, the outcome handed back to the PBX
    • Call audio at the rate a speech service asks for, in every language
    • A Pipecat transport, reaching every speech service Pipecat supports
    • How to connect more than forty voice-agent services, one by one
    • Direct connectors to the realtime speech APIs of five voice-agent services
    • A ready-to-run bridge from a configuration file, each account to its own agent
    • OAuth2 sign-in to the PBX (RFC 8898)
    • A call as a participant in a LiveKit room, as an example
    • A phone voice agent that runs in five minutes, speech to speech, from one command
    • Time from the INVITE to the first audio the agent hears, and from its reply to the first packet, measured and published
    • Calls held for a day and for a week with no growth in memory or processor time, measured and published
    • Node.js and TypeScript package
    • .NET MAUI package for iOS and Android
    • A network test before the call
    • Answering-machine detection on calls the agent places
    • SIP over WebSocket with the connection made by the stack (RFC 7118)
    • LiveCommunicationKit on iOS
    • The Rust API promised between releases and published on crates.io, checked on every change

  2. October 2026 · 1.1 (released)

    Sharper at scale, released ahead of schedule

    Call set-up with no idle wait, in-band DTMF detection at a fraction of its cost, a forked call ringing on every line it reaches, outgoing audio on a steady clock, lost Opus frames rebuilt.

    What it delivered, 5 items
    • Call set-up in the voice agent with no idle wait between reads
    • In-band DTMF detection at a fraction of its cost per frame
    • A call forked to two lines of one stack rings on both
    • Outgoing audio sent on a steady 20 ms clock
    • Opus rebuilds a lost frame from the copy the next packet carries, sized by the loss the far end reports

  3. October 2026 · 1.0 (released)

    Complete, at C ABI 1.0

    Signalling, media, encryption and NAT traversal behind one C ABI, bindings for seven languages, ten thousand calls in one process. Commercial licences are available now.

    What it delivered, 4 items
    • SIP signalling, media, SRTP and DTLS-SRTP, ICE, STUN and TURN, STIR/SHAKEN behind one C ABI
    • Swift, Kotlin and Java, .NET, Python, Dart and React Native bindings
    • Device mode with the platform echo cancellation, or application mode for voice agents
    • Ten thousand concurrent calls in one process

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