CalloraVoipSdk.Audio.Linux 4.15.0

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CalloraVoipSdk

Put a real telephone in a .NET application. SIP signalling, RTP/SRTP media, WebRTC and a self-hostable STUN/TURN server — one facade, no native dependencies.

<p align="center"> <img src="docs/assets/call.svg" width="800" alt="A terminal showing an outbound SIP call placed with CalloraVoipSdk: registration over TLS, dial, SRTP media, RTCP quality metrics, DTMF, attended transfer and hangup — followed by the CI interop checks against a real Asterisk and real browsers."> </p>

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Thirty seconds

dotnet add package CalloraVoipSdk
dotnet add package CalloraVoipSdk.Audio.Linux   # or .Audio.Windows

using CalloraVoipSdk;
using CalloraVoipSdk.Core.Domain.Lines;

using var client = new VoipClient();

var connected = await client.ConnectAsync(new SipAccount
{
    Username  = "1001",
    Password  = "secret",
    SipServer = "pbx.example.org",
});

var dialed = await client.DialAndWaitUntilConnectedAsync(
    connected.Line!, "sip:1002@pbx.example.org");

await client.AttachDefaultAudioAsync(dialed.Call!);

A registered line, a connected call, live audio on the machine's default devices. Both results carry a status you should check rather than the ! used here for brevity — a call that nobody answered is not an exception, it is an outcome.

Everything below is what happens when you need more than that.

Why this one

It is checked against real software, not against itself. Every CI run boots an Asterisk (PJSIP) container and runs the matrix through it — registration, in- and outbound calls with live RTP, codec negotiation, SRTP-SDES, DTMF, hold, blind and attended transfer, session timers, early media, TCP/TLS — plus a two-leg bridged call whose media is verified byte-exact in both directions. The WebRTC side runs against Chromium and Firefox (headless, via Playwright) with the SDK as offerer and as answerer, and the TURN server against a real coturn.

The limits are written down. A protocol library that only advertises what works is a library you find the edges of in production. What is proven, and what is not is a section, not a footnote.

Three depths, mixable per call. Most libraries make you choose once between a black box and a raw protocol stack. Here the levels sit on top of each other, and using the easy one does not close off the others:

What stays internal stays internal: SIP and RTP implementation classes, and arbitrary mutation of the wire. Controlled extensibility rather than an unstable stack with the lid off.

No native dependencies for the protocol stack. Managed C# end to end, so it runs where .NET runs. Audio devices are the exception and live in their own packages.

What it does

<table> <tr><td valign="top" width="50%">

SIP

• Registration with digest auth (MD5 and SHA-256), NAT rport, DNS SRV/NAPTR
• Inbound and outbound calls, hold, re-INVITE, UPDATE
• Blind and attended transfer (REFER, RFC 3515)
• SUBSCRIBE/NOTIFY, PUBLISH, MESSAGE, INFO, PRACK
• Session timers (RFC 4028), early media (RFC 3960)
• UDP, TCP, TLS, WS, WSS

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Media

• G.711 µ-law/A-law, G.722, Opus; RFC 4733 DTMF
• Adaptive jitter buffer with loss concealment
• RTCP sender/receiver reports and RTCP XR VoIP metrics (MOS, burst/gap)
• SRTP via SDES and DTLS-SRTP, with replay windows on RTP and RTCP
• Symmetric RTP (comedia) for NAT
• Per-call media taps: attach frame receivers and senders to any call

</td></tr> <tr><td valign="top">

WebRTC

• Peer connections, offer/answer, trickle ICE, ICE restart
• Audio and VP8/H.264 video, BUNDLE, multi-track
• Receive-side simulcast, recording taps
• Browser-verified in CI against Chromium and Firefox

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Connectivity

• STUN and TURN client
• STUN/TURN server you can host yourself
• UDP relay, plus TCP/TLS relay for networks that only allow 443
• ICE consent freshness (RFC 7675)

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Full API reference: bechsteindigital.github.io/callora-voip-sdk

Status

The line between "we run this in production" and "this compiles and has tests" is where most integration time goes, so it is drawn here.

Known gaps and interop defects are tracked in the issue tracker. Interop reports from a carrier or a device we have not seen are the single most useful contribution.

Packages

Headless services need no audio package at all — media flows through streams and taps.

Examples

Runnable projects in examples/, roughly in order of depth:

Extending it

client.Modules takes feature modules that ship as separate packages. A module implements IVoipClientModule, attaches to the client, and is then resolvable by any interface it implements:

client.Modules.Register(new MyRecordingModule());

var recording = client.Modules.Get<IMyRecordingFeature>();       // throws when absent
if (client.Modules.TryGet<IMyRecordingFeature>(out var feature)) // or probe
    feature.Start();

Modules build on the per-call media tap, and its contract is two sentences worth reading twice: IMediaReceiver.FrameReceived fires synchronously on the media path — a handler must buffer and return immediately, never block, or it stalls the call it is listening to. The negotiated format (payload type, clock rate, samples per packet) is on ICall.MediaParameters.

Commercial modules are in development on top of this seam, distributed through a private feed rather than nuget.org: Callora.Realtime (call audio to realtime AI APIs, with pacing, backpressure and barge-in — the basis for voice agents), Callora.WebSocket (raw call audio over a socket), and Callora.Privacy / Risk / Intelligence (redaction and consent, spam screening, transcription and sentiment). The SDK core stays open and free; the modules are licensed separately — info@bechstein.digital for early access.

Compatibility

net8.0, net9.0, net10.0. Windows, Linux and macOS for the protocol stack; audio devices on Windows and Linux.

Versioning is SemVer, and the public surface is a tracked file. Every public type and member lives in PublicApi.approved.txt, and a CI gate fails the build when a change is not reflected there — so a breaking change is a reviewable diff rather than something a consumer discovers after upgrading.

Per-release detail lives in docs/release-notes/; the machine-readable list is CHANGELOG.md.

Building it yourself

git clone https://github.com/BechsteinDigital/callora-voip-sdk.git
cd callora-voip-sdk
dotnet test tests/CalloraVoipSdk.ArchitectureTests   # the gates CI runs first
dotnet test                                          # the standard set

The Asterisk and browser interop suites need a Docker daemon and self-skip without one. Maintainer workflows, the architecture map and the invariants are in MAINTAINING.md; the rules the tests enforce are in ENGINEERING_RULES.md.

Contributing

Bug reports, interop feedback and pull requests are welcome — see CONTRIBUTING.md and the CODE_OF_CONDUCT.md.

Found a security issue? Do not open an issue — SECURITY.md says where to send it.

License

Apache-2.0. Third-party components are listed in THIRD-PARTY-NOTICES.md.

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