Bandwidth RTC Swift SDK

An iOS SDK for building real-time audio communication apps on the Bandwidth platform. Wraps WebRTC and connects to the Bandwidth BRTC gateway over a JSON-RPC 2.0 WebSocket signaling channel. Distributed as a signed XCFramework via Swift Package Manager.

For product documentation, see the Bandwidth RTC developer docs.

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Requirements

• iOS 17+
• Swift 5.9+
• Xcode 16+
• Swift Package Manager

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Installation

Swift Package Manager

In Xcode, go to File → Add Package Dependencies and enter the repository URL. Or add it directly to your Package.swift:

dependencies: [
    .package(url: "https://github.com/Bandwidth/swift-brtc-sdk", from: "1.0.0"),
],
targets: [
    .target(
        name: "YourApp",
        dependencies: [
            .product(name: "BandwidthRTC", package: "swift-brtc-sdk")
        ]
    ),
]

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Quick Start

import BandwidthRTC

class CallService {
    let client = BandwidthRTCClient()

    func startCall(token: String) async throws {
        // Called when a remote participant starts streaming
        client.onStreamAvailable = { stream in
            print("Remote stream available: \(stream.streamId)")
        }

        // Called when a remote participant stops streaming
        client.onStreamUnavailable = { streamId in
            print("Remote stream removed: \(streamId)")
        }

        // Called once the gateway signals readiness
        client.onReady = { metadata in
            print("Connected — endpointId: \(metadata.endpointId ?? "unknown")")
        }

        // Called if the WebSocket drops unexpectedly
        client.onDisconnected = {
            print("Disconnected from gateway")
        }

        // Connect and publish local microphone audio
        try await client.connect(authParams: RtcAuthParams(endpointToken: token))
        let localStream = try await client.publish(audio: true)
        print("Publishing local audio: \(localStream.streamId)")
    }

    func endCall() {
        client.disconnect()
    }
}

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Callbacks

Property	When it fires
onReady	Gateway signals the endpoint is ready to receive calls
onStreamAvailable	A remote participant begins streaming audio
onStreamUnavailable	A remote participant stops streaming
onDisconnected	WebSocket connection dropped unexpectedly
onLocalAudioLevel	Per-chunk Float32 mic samples (for visualization)
onRemoteAudioLevel	Per-chunk Float32 remote playout samples (for visualization)

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Configuration

BandwidthRTCClient init

Parameter	Type	Default	Description
logLevel	LogLevel	.warn	SDK log verbosity (.off, .error, .warn, .info, .debug, .trace)

RtcAuthParams

Passed to connect(authParams:options:).

Field	Type	Required	Description
endpointToken	String	Yes	JWT endpoint token obtained from the Bandwidth Endpoints API

RtcOptions

Passed as options to connect(authParams:options:). All fields are optional.

Field	Type	Default	Description
websocketUrl	String?	nil	Override the default BRTC gateway WebSocket URL
iceServers	[RTCIceServer]?	nil	Custom STUN/TURN servers. Uses WebRTC defaults when nil
iceTransportPolicy	RTCIceTransportPolicy?	.all	Restrict ICE candidate types (e.g. .relay to force TURN)
audioProcessing	AudioProcessingOptions	See below	Audio session, sample rate, and buffer configuration

AudioProcessingOptions

Nested inside RtcOptions.audioProcessing.

Field	Type	Default	Description
audioSessionMode	AVAudioSession.Mode	.voiceChat	AVAudioSession mode. .voiceChat enables Apple hardware AEC and noise suppression; use .default or .measurement to disable
audioSessionCategoryOptions	AVAudioSession.CategoryOptions	[.allowBluetoothHFP]	AVAudioSession category options (e.g. add .defaultToSpeaker to route audio to the loudspeaker)
inputSampleRate	Double	48000	Recording sample rate in Hz. 48 kHz matches WebRTC's Opus rate and avoids a resampling step
outputSampleRate	Double	48000	Playout sample rate in Hz
inputChannels	Int	1	Number of input (microphone) channels
outputChannels	Int	1	Number of output (speaker) channels
useLowLatency	Bool	false	Request a 5 ms I/O buffer from AVAudioSession. Reduces latency at the cost of higher CPU usage
preferredIOBufferDuration	TimeInterval?	nil	Explicit I/O buffer duration in seconds. Overrides useLowLatency when set; the OS rounds to the nearest supported value

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Samples

Sample apps can be found in Bandwidth-Samples.

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Compatibility

This SDK follows SemVer 2.0.0. Each release publishes three Git tags — v{MAJOR}, v{MAJOR}.{MINOR}, and v{MAJOR}.{MINOR}.{PATCH} — so you can pin at any granularity in your Package.swift:

// Exact version
.package(url: "https://github.com/Bandwidth/swift-brtc-sdk", exact: "1.0.0"),

// Latest patch in a minor series
.package(url: "https://github.com/Bandwidth/swift-brtc-sdk", from: "1.0.0"),

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Contributing

Every PR must bump the VERSION file. CI will fail if you don't.

1. Make your changes.
2. Open VERSION at the repo root and increment the version (e.g. 1.0.1 → 1.0.2). Use patch for bug fixes, minor for new features, major for breaking changes.
3. Open your PR — build.yml enforces that the new version is strictly greater than main.

Running Unit Tests

In Xcode

1. Open the package:

   open Package.swift
   

2. Select the BandwidthRTC scheme.
3. Choose an iOS Simulator destination (e.g. iPhone 17 Pro).
4. Press ⌘U.

From the command line

xcodebuild test \
  -scheme BandwidthRTC \
  -destination 'platform=iOS Simulator,name=iPhone 17 Pro'