portsip 0.1.5

A Flutter plugin for integrating the PortSIP VoIP SDK, enabling SIP-based voice and video communications in your applications.

0.1.5

• Feature (Android): raw audio recording. startPcmRecord({sessionId, recordFilePath, recordFileName}) and stopPcmRecord({sessionId}) write the PCM the SDK hands to its audio stream callback, instead of routing it through the SDK's own recorder as [startRecord] does. Both directions land in one stereo <recordFileName>.wav — sent audio on the left channel, received on the right — so neither party can clip the other and the two stay separable; a call that produced only one direction stays mono. recordFilePath must be an existing directory. stopPcmRecord merges the per-direction parts off the platform thread and completes with the finished recording's path, or null if nothing was captured; stopping from an onInviteClosed handler still returns the recording.

0.1.4

• Feature: call recording. startRecord({sessionId, recordFilePath, recordFileName, appendTimeStamp, audioChannels, fileFormat, audioRecordMode, videoRecordMode}) and stopRecord({sessionId}), wired on both iOS and Android. recordFilePath must be an existing directory — the SDK does not create it. New [PortsipRecordFileFormat] (wave | amr | mp3 | mp4) and [PortsipRecordMode] (none | receive | send | both) enums carry the SDK's values.
• Feature: play a file to the remote party. startPlayingFileToRemote({sessionId, fileUrl, loop, playAudio}) and stopPlayingFileToRemote({sessionId}), wired on both iOS and Android. The new [PortsipPlayFileAudioMode] enum (none | fileOnly | mixedWithMicrophone) controls how the file mixes with the microphone;

0.1.3

• Fix (iOS): startKeepAwake no longer outlives the SDK. Both teardown paths now call stopKeepAwake() before unInitialize(), so a keep-awake started on background can't leave a background assertion running on a disposed SDK instance.
• Fix (iOS): background/foreground transitions no longer recreate a torn-down SDK. The appDidEnterBackground / appWillEnterForeground handlers now toggle keep-awake through the backing SDK reference and no-op when the SDK isn't initialized, instead of lazily spinning up a fresh, unconfigured instance.

0.1.2

• Feature (Android): Telecom-owned audio routing. Adds setAudioRoute({kind}) (builtInReceiver | builtInSpeaker | bluetooth | headphones) and the onConnectionServiceAudioRoute ([ConnectionServiceAudioRouteEvent]) event carrying availablePorts and currentPort, so a route picker can be driven entirely from plugin events. The chosen route now survives hold, unhold, swap, conference join and end-and-resume. When routing through Telecom, call enableAudioManager(enable: false) — the SDK's audio manager competes with Telecom and resets the route on every new call leg.
• Fix (Android): Bluetooth audio no longer drops seconds after connecting.
• Fix (Android): endCallAndResume no longer leaves a phantom call in the system UI and preserves the audio route across the resume.
• Fix (Android): system hold requests are no longer forwarded to the SDK while a conference is active (they silenced joined legs).
• Fix (iOS): the loudspeaker choice survives answer, hold, swap and merge instead of resetting to the receiver mid-call.
• Fix (iOS): the system in-call UI's speaker button no longer shows stale state or stops responding.
• Fix (iOS): Bluetooth audio no longer goes silent after ending one leg of a multi-call.
• Fix: setAudioRoute returns -1 on platforms where it isn't implemented (iOS) instead of throwing MissingPluginException.

0.1.1

• Feature: getStatistics({sessionId}) — requests RTP media statistics for a session. Results are delivered asynchronously on the events stream as an onStatistics ([StatisticsEvent]) carrying a JSON stat string with byte/packet/jitter/loss counters. Poll periodically (e.g. once per second) to track the live data usage of an ongoing call. Wired on both iOS and Android.

0.1.0

• BREAKING: Conference merge no longer ends any call leg. The old "end the merged-away leg" model tore down the shared audio session and silenced participants on merge.
• Feature: requestCallGroup({sessionIds}) — group calls into one native conference in the system UI (iOS CXSetGroupCallAction; Android signals the host to build the SDK bridge). No call is ended; the host is notified via onCallKitMerge to build/confirm the audio bridge.
• Feature: endCallAndResume({endSessionId, resumeSessionId}) — ends one leg and resumes another atomically, keeping exactly one active system call so the audio session is never torn down. Prefer over separate hangUp/unHold.
• Deprecated: reportConferenceMerged(...) — use requestCallGroup. Ending the merged-away leg tears down the audio session.

0.0.9

• Feature: Audio conferencing on both platforms. Adds createAudioConference(), joinToConference({sessionId}), removeFromConference({sessionId}), and destroyConference() to build and tear down a mixing bridge so all participants hear each other.
• Feature: Expose swapHold({holdSessionId, unholdSessionId}) — atomically holds one call and resumes another. Required on iOS, where the CallKit mirror must be a single transaction or CallKit's one-active-call enforcement can end a call when swapping with separate hold/unHold calls.
• Feature: Expose reportConferenceMerged({primarySessionId, mergedAwaySessionIds}). Ends the CallKit / ConnectionService calls for merged-away legs (their SIP sessions stay live in the bridge) so the OS presents one call instead of several.
• Feature: Expose updateOsCallDisplay({sessionId, displayName}) to update the name the system call UI shows for a session — e.g. a conference title after a merge.
• Feature: New CallKitMergeEvent (onCallKitMerge, iOS only) emitted when the user taps Merge in the system call UI (CXSetGroupCallAction), carrying both sessionId and otherSessionId so the app can bridge them into a conference.
• Feature: makeCall(...) gains reportToConnectionService (default true) and displayName. Pass reportToConnectionService: false for additional conference legs to keep the whole conference as one system call; set displayName at call time since iOS doesn't reliably honor display updates after an outgoing call starts.
• Feature: hangUp(...) gains reportToConnectionService (default true). Pass false to drop a single conference leg without ending the system call or tearing down the shared audio session.

0.0.8

• Feature: Expose refreshRegistration({int expires = 0}) on both platforms. Call it when the device's network changes (e.g. WiFi to cellular handoff) — the native SDK re-registers over the new network interface. Pass expires: 0 (default) to keep the previously negotiated expiry and force an immediate re-register.
• Feature: Expose updateCall({sessionId, enableAudio, enableVideo}) on both platforms. Sends a re-INVITE that renegotiates the media streams of an established call — use it after a network change to re-anchor audio to the new interface.
• Feature: Expose enableSessionTimer({int timerSeconds = 90, int refreshMode = 0}) on both platforms. Enables SIP session timers (RFC 4028) so a dead call leg is reaped by the server at expiry when the client stops refreshing (e.g. lost network mid-call), instead of lingering for minutes. Minimum 90s; refreshMode 0 = UAC (caller) refreshes, 1 = UAS (callee).

0.0.7

• Fix (Android): Correctly report and track self-managed outgoing calls with the system ConnectionService across their full lifecycle (placement, connect, hang-up), and reliably tear the call down on hang-up.

0.0.6

• iOS: When the app is swipe-killed (or system-terminated) while a CallKit call is active, the SIP dialog is now properly closed natively before the process dies. The controller observes UIApplication.willTerminateNotification and ends each active call and unregisters the device from the SIP server, so the server no longer holds the call open or keeps the device registered after termination.

0.0.5

• Fix: Add unInitialize() for runtime SIP user swap. Tears down the native SDK (unRegisterServer(0) + unInitialize()) so the next initialize() + register() can load a different user cleanly without hitting -60095 (ECoreAllowOnlyOneUser). Unlike dispose(), this keeps the event stream, method-channel handler, iOS observers/CallKit, and Android ConnectionService alive — existing event listeners keep firing after the swap.

0.0.4

• Fix: Change the CXHandle type from .generic to .phoneNumber when reporting outgoing calls to CallKit via CXStartCallAction.

0.0.3

• Fix: unregister no longer uninitializes the SDK, so calling register again afterwards works correctly without requiring a full re-initialization.

0.0.2

• BREAKING CHANGE: PortSIP VoIP SDK binaries are no longer bundled with the plugin
  • iOS: Developers must place PortSIPVoIPSDK.xcframework at ios/Frameworks/ and embed it in Xcode (Embed & Sign)
  • Android: Developers must place portsipvoipsdk.jar at android/app/libs/ and add it as an implementation dependency
  • Plugin package size reduced from ~80MB to under 1MB
  • See README for detailed setup instructions

0.0.1

• Initial release of the PortSIP Flutter plugin
• SIP Registration: Configure and register SIP accounts with server
• Outgoing Calls: Make voice calls with session management
• Call Controls: Hold, unhold, mute, and hang up calls
• DTMF Support: Send DTMF tones via RFC2833, INFO, or INBAND methods
• Audio Management:
  • Loudspeaker toggle
  • Audio codec configuration
  • AEC, AGC, ANS, CNG, and VAD controls
• iOS CallKit Integration: Native call UI support for iOS
• Android ConnectionService: Native call UI support for Android
• Security: SRTP policy configuration and TLS certificate support
• Event System: Broadcast stream for SDK events with typed event classes
• Lifecycle Management: Proper SDK state management with initialize/dispose
• Logging: Configurable debug logging across Dart, iOS, and Android layers
• 3GPP Support: Optional 3GPP tags for carrier compatibility