PjSip.Net.Interop 1.0.6

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☎️ PjSip.Net

High-level SIP telephony SDK for .NET 10

Based on PJSIP 2.16 with native TLS support (Schannel on Windows, OpenSSL on Android) Compatible with WinForms · WPF · MAUI · Mac Catalyst · Console

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✨ Key Features

	Feature	Description
📞	Call Management	Make, receive, hold, transfer, and record calls
🔒	Native TLS	Schannel (Windows), Secure Transport (macOS/iOS), OpenSSL (Android)
👥	Presence & BLF	Monitor user availability with SUBSCRIBE/NOTIFY
🎙️	Audio Control	Device selection, volume, mute, codec management (G.711, G.729, Opus, Speex, GSM, iLBC)
🔀	Conferencing	Multi-party audio bridge with merge/split
💬	SIP Messaging	Send and receive text messages via SIP MESSAGE (RFC 3428)
📊	Call Quality	Real-time RTP stats, jitter, packet loss, MOS score
🌐	NAT Traversal	STUN, ICE, and TURN support built-in
💉	Dependency Injection	First-class IServiceCollection integration
📱	Multi-Platform	Windows, macOS, Android, iOS from a single API

---

📑 Table of Contents

• Requirements
• Installation
• Quick Start
• Configuration
  • SipPhoneOptions
  • SipAccountOptions
  • Transport and TLS
  • NAT/STUN/ICE/TURN
• Dependency Injection
• API Reference
  • ISipPhone
  • ISipAccount
  • ISipCall
  • ISipAudioManager
  • ISipCodecManager
  • ISipPresenceManager
  • ISipMessaging
  • ISipConferenceBridge
  • ISipCallRecorder
  • ISipToneGenerator
  • ISipCallQualityMonitor
  • ISipCallHistory
  • ISipNetworkMonitor
• Events
• Advanced Features
• Platform Examples
• Error Handling
• Audio
• Low-Level Access (pjsua2)
• Architecture
• Supported Platforms
• Build from Source

---

📋 Requirements

• .NET 10 SDK or higher
• Native package corresponding to your platform (installed automatically via NuGet)

📦 Installation

# Main SDK (required)
dotnet add package PjSip.Net

# Native binaries — install the one for your target platform
dotnet add package PjSip.Net.Native.Win64        # Windows x64
dotnet add package PjSip.Net.Native.MacOS         # macOS x64 / arm64
dotnet add package PjSip.Net.Native.Android       # Android arm64
dotnet add package PjSip.Net.Native.iOS           # iOS arm64

Note: The native package contains the compiled pjsua2 binary and is automatically copied to the output directory.

---

🚀 Quick Start

using Microsoft.Extensions.DependencyInjection;
using PjSip.Net;
using PjSip.Net.Accounts;
using PjSip.Net.DependencyInjection;
using PjSip.Net.Transport;

// 1. Configure services
var services = new ServiceCollection();
services.AddLogging();
services.AddPjSip(options =>
{
    options.Transports.Add(new SipTransportOptions
    {
        Type = SipTransportType.Udp,
        Port = 5060
    });
    options.Accounts.Add(new SipAccountOptions
    {
        Username = "1001",
        Password = "secret",
        Domain = "pbx.mycompany.com",
        Registrar = "sip:pbx.mycompany.com"
    });
});

// 2. Resolve and start
var provider = services.BuildServiceProvider();
var phone = provider.GetRequiredService<ISipPhone>();

phone.IncomingCall += (s, e) =>
{
    Console.WriteLine($"Incoming call from {e.RemoteUri}");
    e.Call.Answer();  // Answer automatically
};

phone.CallStateChanged += (s, e) =>
    Console.WriteLine($"Call {e.Call.Id}: {e.OldState} -> {e.NewState}");

await phone.StartAsync();

// 3. Make a call
var call = phone.MakeCall(phone.Accounts[0], "sip:1002@pbx.mycompany.com");

// 4. Hang up
call.Hangup();

// 5. Shutdown cleanly
await phone.StopAsync();

---

⚙️ Configuration

SipPhoneOptions

Global SIP endpoint options. Configured when registering the service.

services.AddPjSip(options =>
{
    options.UserAgent = "MyApp/2.0";        // User-Agent in SIP headers (default: "PjSip.Net/1.0")
    options.LogLevel = 4;                    // PJSIP log level: 0=fatal, 5=trace (default: 4)
    options.MaxCalls = 8;                    // Maximum simultaneous calls (default: 4)
    options.UseCompactForm = false;          // Compact SIP headers (default: false)
    options.CallHistoryMaxEntries = 1000;    // Maximum history entries (default: 1000)
    options.Transports = [ ... ];            // List of transports to create
    options.Accounts = [ ... ];             // Accounts to register on start
    options.Nat = new NatOptions { ... };   // NAT/STUN/ICE/TURN configuration
});

Property	Type	Default	Description
UserAgent	string	"PjSip.Net/1.0"	User-Agent header value in SIP messages
LogLevel	int	4	PJSIP internal log verbosity (0-5)
MaxCalls	int	4	Maximum number of simultaneous calls
UseCompactForm	bool	false	Use compact form for SIP headers
CallHistoryMaxEntries	int	1000	Maximum entries stored in call history
Transports	List<SipTransportOptions>	[]	SIP transports to create on startup
Accounts	List<SipAccountOptions>	[]	SIP accounts to register automatically
Nat	NatOptions	new()	NAT traversal configuration (STUN/ICE/TURN)

SipAccountOptions

Configuration for an individual SIP account.

new SipAccountOptions
{
    Username = "1001",                       // SIP username (required)
    Password = "secret",                     // Password (required)
    Domain = "pbx.mycompany.com",           // SIP domain (required)
    Registrar = "sip:pbx.mycompany.com",    // Registrar URI (null = uses Domain)
    OutboundProxy = "sip:proxy.mycompany.com", // Outbound proxy (null = none)
    DisplayName = "John Doe",               // Display name for caller ID
    Realm = "*",                             // Authentication realm (null = automatic)
    RegistrationTimeout = 300,               // Registration expiration in seconds (default: 300)
    RegisterOnAdd = true,                    // Register automatically when added (default: true)
    UseTls = false                           // Append ;transport=tls to registrar/proxy URIs (default: false)
}

Property	Type	Default	Description
Username	string	required	SIP username for authentication
Password	string	required	Account password
Domain	string	required	SIP domain/server
Registrar	string?	null	Complete registrar URI. If null, constructed from Domain
OutboundProxy	string?	null	Outbound SIP proxy URI. Scheme (sip:/sips:) added automatically if missing
DisplayName	string?	null	Display name in Caller ID
Realm	string?	null	Realm for digest auth. null = accepts any challenge
RegistrationTimeout	int	300	REGISTER expiration time in seconds
RegisterOnAdd	bool	true	If true, sends REGISTER automatically when adding account
UseTls	bool	false	Appends ;transport=tls to registrar and proxy URIs

🔐 Transport and TLS

// UDP (unencrypted)
options.Transports.Add(new SipTransportOptions
{
    Type = SipTransportType.Udp,
    Port = 5060
});

// TCP
options.Transports.Add(new SipTransportOptions
{
    Type = SipTransportType.Tcp,
    Port = 5060
});

// TLS (encrypted) — Uses Schannel on Windows, no OpenSSL dependency
options.Transports.Add(new SipTransportOptions
{
    Type = SipTransportType.Tls,
    Port = 5061,
    Tls = new TlsOptions
    {
        VerifyServer = true,               // Validate server certificate (default: true)
        VerifyClient = false,              // Require client certificate (default: false)
        CertificateFile = null,            // Path to client certificate (.pem)
        PrivateKeyFile = null,             // Path to client private key (.pem)
        CaListFile = null                  // Path to additional trusted CAs (.pem)
    }
});

// IPv6
options.Transports.Add(new SipTransportOptions
{
    Type = SipTransportType.Tls6,          // TLS over IPv6
    Port = 5061
});

Available transport types:

Enum	Protocol	Default Port
SipTransportType.Udp	UDP/IPv4	5060
SipTransportType.Tcp	TCP/IPv4	5060
SipTransportType.Tls	TLS/IPv4	5061
SipTransportType.Udp6	UDP/IPv6	5060
SipTransportType.Tcp6	TCP/IPv6	5060
SipTransportType.Tls6	TLS/IPv6	5061

TlsOptions:

Property	Type	Default	Description
VerifyServer	bool	true	Validate server TLS certificate
VerifyClient	bool	false	Require client TLS certificate
CertificateFile	string?	null	Path to client certificate (PEM format)
PrivateKeyFile	string?	null	Path to client private key (PEM format)
CaListFile	string?	null	Path to additional trusted CA list

Windows: TLS uses Schannel (OS native). No need to install OpenSSL. macOS/iOS: Uses system Secure Transport. Android: TLS enabled via OpenSSL 3.4.1 (statically linked).

🌐 NAT/STUN/ICE/TURN

NAT traversal configuration for networks behind firewalls or NAT routers.

options.Nat = new NatOptions
{
    EnableStun = true,
    StunServers = ["stun.l.google.com:19302", "stun1.l.google.com:19302"],
    EnableIce = true,                        // ICE for media NAT traversal (default: true)
    EnableTurn = false,                      // TURN relay (for symmetric NAT)
    TurnServer = "turn.mycompany.com:3478",
    TurnUsername = "user",
    TurnPassword = "pass",
    TurnTransport = NatTraversalType.Udp,   // TURN transport: Udp, Tcp, Tls
    IceAggressiveNomination = false          // ICE aggressive nomination
};

Property	Type	Default	Description
EnableStun	bool	false	Enable STUN resolution to discover public IP
StunServers	List<string>	[]	List of STUN servers (host:port)
EnableIce	bool	true	Enable ICE for media NAT traversal
EnableTurn	bool	false	Enable TURN relay for symmetric NAT
TurnServer	string?	null	TURN server address (host:port)
TurnUsername	string?	null	Username for TURN authentication
TurnPassword	string?	null	Password for TURN authentication
TurnTransport	NatTraversalType	Udp	TURN transport: Udp, Tcp, Tls
IceAggressiveNomination	bool	false	ICE aggressive nomination (faster, less reliable)

---

💉 Dependency Injection

Basic registration (Singleton)

services.AddPjSip(options =>
{
    options.Transports.Add(new SipTransportOptions { Type = SipTransportType.Udp });
    options.Accounts.Add(new SipAccountOptions
    {
        Username = "1001",
        Password = "secret",
        Domain = "pbx.mycompany.com"
    });
});

Registration with explicit lifetime

using PjSip.Net.DependencyInjection;

// Singleton (default) — one instance for entire application
services.AddPjSip(options => { ... }, PjSipServiceLifetime.Singleton);

// Scoped — one instance per scope (useful in web applications)
services.AddPjSip(options => { ... }, PjSipServiceLifetime.Scoped);

Registered services

AddPjSip automatically registers:

Service	Description
ISipPhone	Main facade — manages accounts, calls, and transport
ISipAudioManager	Audio device management (microphone, speaker, volume)
ISipCodecManager	Audio codec management (priorities, enable/disable)
ISipPresenceManager	Presence and BLF (Busy Lamp Field)
ISipMessaging	SIP messaging (SIP MESSAGE)
ISipConferenceBridge	Audio conferencing (bridge)
ISipCallRecorder	Call recording
ISipToneGenerator	Tone generator (ringback, busy, dial, DTMF)
ISipCallQualityMonitor	Call quality monitoring (RTP stats, MOS)
ISipCallHistory	Call history
ISipNetworkMonitor	Network change monitoring

Inject into your classes

public class MyTelephonyService
{
    private readonly ISipPhone _phone;

    public MyTelephonyService(ISipPhone phone)
    {
        _phone = phone;
        _phone.IncomingCall += OnIncomingCall;
    }

    public async Task StartAsync()
    {
        await _phone.StartAsync();
    }

    public ISipCall Call(string destination)
    {
        return _phone.MakeCall(_phone.Accounts[0], destination);
    }

    private void OnIncomingCall(object? sender, IncomingCallEventArgs e)
    {
        // Logic for incoming calls
    }
}

You can also inject sub-managers directly:

public class MyPresenceService
{
    private readonly ISipPresenceManager _presence;
    private readonly ISipCallHistory _history;

    public MyPresenceService(ISipPresenceManager presence, ISipCallHistory history)
    {
        _presence = presence;
        _history = history;
    }

    public async Task ShowAvailableAsync()
    {
        await _presence.SetMyPresenceAsync(BuddyState.Online, "Available");
    }

    public int MissedCallsToday()
    {
        return _history.GetMissedCalls().Count;
    }
}

---

📖 API Reference

ISipPhone

Main SDK facade. Manages the SIP endpoint lifecycle, accounts, calls, and all sub-managers.

public interface ISipPhone : IAsyncDisposable, IDisposable

Properties:

Property	Type	Description
State	SipPhoneState	Current phone state
Accounts	IReadOnlyList<ISipAccount>	Registered SIP accounts
Audio	ISipAudioManager	Audio device manager
Codecs	ISipCodecManager	Audio codec manager
Presence	ISipPresenceManager	Presence and BLF manager
Messaging	ISipMessaging	SIP messaging (MESSAGE)
Conference	ISipConferenceBridge	Conference bridge
Recorder	ISipCallRecorder	Call recorder
Tones	ISipToneGenerator	Tone generator
Quality	ISipCallQualityMonitor	Call quality monitor
History	ISipCallHistory	Call history
Network	ISipNetworkMonitor	Network change monitor

Methods:

Method	Return	Description
StartAsync(ct)	Task	Initializes PJSIP, creates transports and registers configured accounts
StopAsync(ct)	Task	Hangs up all calls, unregisters accounts and destroys endpoint
AddAccount(options)	ISipAccount	Adds a new SIP account at runtime
RemoveAccount(account)	void	Removes and unregisters an account
MakeCall(account, uri)	ISipCall	Initiates an outgoing call from an account
MakeCall(account, uri, headers)	ISipCall	Initiates an outgoing call with custom SIP headers

Events:

Event	EventArgs	Description
IncomingCall	IncomingCallEventArgs	Incoming call on any account
CallStateChanged	CallStateChangedEventArgs	State change in any call
RegistrationStateChanged	RegistrationStateChangedEventArgs	Registration change in any account
TransportStateChanged	TransportStateChangedEventArgs	Transport state change
MwiStateChanged	MwiStateChangedEventArgs	New message waiting indicator (voicemail)

States (SipPhoneState):

State	Description
Idle	Newly created, not initialized
Starting	Initializing PJSIP endpoint
Running	Operational — can make and receive calls
Stopping	Shutting down
Stopped	Cleanly stopped
Error	Error during startup or shutdown

---

ISipAccount

Represents a SIP account from which calls can be sent/received.

public interface ISipAccount : IDisposable

Properties:

Property	Type	Description
Id	string	Unique account identifier
Uri	string	SIP URI of the account (e.g., sip:1001@pbx.com)
RegistrationState	SipRegistrationState	Current registration state
Options	SipAccountOptions	Account configuration
ActiveCalls	IReadOnlyList<ISipCall>	Active calls on this account
DndMode	DndMode	Do Not Disturb mode (read/write)
CallForwarding	CallForwardingOptions	Call forwarding configuration
MwiInfo	MwiInfo?	Message waiting information (voicemail), null if no data

Methods:

Method	Return	Description
RegisterAsync(ct)	Task	Sends REGISTER to server
UnregisterAsync(ct)	Task	Sends un-REGISTER to server
MakeCall(destinationUri)	ISipCall	Initiates a call from this account
MakeCall(destinationUri, headers)	ISipCall	Initiates a call with custom SIP headers

Events:

Event	EventArgs	Description
RegistrationStateChanged	RegistrationStateChangedEventArgs	Registration state change
IncomingCall	IncomingCallEventArgs	Incoming call for this account
MwiStateChanged	MwiStateChangedEventArgs	New voicemail state

Registration states (SipRegistrationState):

State	Description
Unregistered	Not registered
Registering	REGISTER sent, waiting for response
Registered	Successfully registered (200 OK)
Unregistering	Un-REGISTER sent
Error	Registration error (401, 403, timeout, etc.)

---

ISipCall

Represents an active SIP call (incoming or outgoing).

public interface ISipCall : IDisposable

Properties:

Property	Type	Description
Id	string	Unique call identifier
State	SipCallState	Current call state
Direction	CallDirection	Incoming or Outgoing
Info	SipCallInfo	Detailed information (URIs, duration, status code)
CustomHeaders	IReadOnlyList<SipHeader>	Custom SIP headers from the call
IsMuted	bool	If microphone is muted for this call
IsOnHold	bool	If call is on hold

Methods:

Method	Description
Answer(statusCode)	Answer the call. Default: 200 (OK)
Answer(statusCode, headers)	Answer with custom SIP headers
Hangup(statusCode)	Hang up the call. Default: 603 (Decline)
Hold()	Put on hold
Unhold()	Remove from hold (re-INVITE)
Transfer(destinationUri)	Transfer call to another destination (REFER)
AttendedTransfer(targetCall)	Attended transfer — connects this call with another active call
SendDtmf(digits)	Send DTMF tones (e.g., "1234#")
SetMute(mute)	Mute/unmute microphone

Common response codes for Answer():

Code	Meaning
180	Ringing (without answering, only signal ring)
200	OK — answer the call
486	Busy Here — reject as busy
603	Decline — reject the call

Call states (SipCallState):

stateDiagram-v2
    [*] --> Null: Created
    Null --> Calling: MakeCall()
    Null --> Incoming: INVITE received
    Calling --> EarlyMedia: 183 + SDP
    Calling --> Connecting: 200 OK
    Incoming --> Connecting: Answer(200)
    Incoming --> Disconnected: Hangup(603)
    EarlyMedia --> Connecting: 200 OK
    Connecting --> Confirmed: Media ready
    Confirmed --> Disconnected: Hangup / BYE
    Disconnected --> [*]

State	Description
Null	Newly created call
Calling	INVITE sent, waiting for response
Incoming	INVITE received, not answered
EarlyMedia	Receiving early media (183 + SDP)
Connecting	2xx response received, establishing media
Confirmed	Active call with bidirectional audio
Disconnected	Call terminated

SipCallInfo (call information):

Property	Type	Description
CallId	string	Call-ID from SIP header
RemoteUri	string	Remote party URI
LocalUri	string	Local URI
State	SipCallState	Current state
Direction	CallDirection	Call direction
Duration	TimeSpan	Call duration
RemoteDisplayName	string?	Remote caller display name
StatusCode	int	Last SIP code received
StatusText	string?	Text of last SIP status

SipHeader (custom header):

public sealed record SipHeader
{
    public required string Name { get; init; }
    public required string Value { get; init; }
}

---

ISipAudioManager

Audio device and volume management.

public interface ISipAudioManager

Properties:

Property	Type	Description
CurrentInputDevice	AudioDeviceInfo?	Current active microphone
CurrentOutputDevice	AudioDeviceInfo?	Current active speaker/headset
InputLevel	float	Input volume level (0.0 — 1.0)
OutputLevel	float	Output volume level (0.0 — 1.0)

Methods:

Method	Return	Description
GetInputDevices()	IReadOnlyList<AudioDeviceInfo>	List of available microphones
GetOutputDevices()	IReadOnlyList<AudioDeviceInfo>	List of available speakers
SetInputDevice(deviceId)	void	Change active microphone
SetOutputDevice(deviceId)	void	Change active speaker

AudioDeviceInfo:

Property	Type	Description
DeviceId	int	Device ID
Name	string	Device name (e.g., "Realtek HD Audio")
InputChannels	int	Number of input channels
OutputChannels	int	Number of output channels
Driver	string?	Audio driver name

---

ISipCodecManager

Audio codec management: list, prioritize, enable and disable.

public interface ISipCodecManager

Methods:

Method	Return	Description
GetCodecs()	IReadOnlyList<CodecInfo>	List of available codecs with their priorities
SetCodecPriority(codecId, priority)	void	Set codec priority (0-255, 0 = disabled)
EnableCodec(codecId, priority)	void	Enable codec with optional priority (default: 128)
DisableCodec(codecId)	void	Disable codec (priority = 0)

CodecInfo:

Property	Type	Description
CodecId	string	Codec identifier (e.g., "PCMU/8000", "opus/48000", "G729/8000")
Description	string	Human-readable codec description
Priority	int	Current priority (0-255, 0 = disabled)
ClockRate	int	Sampling frequency in Hz
ChannelCount	int	Number of audio channels

---

ISipPresenceManager

Presence management (SUBSCRIBE/NOTIFY) and BLF (Busy Lamp Field).

public interface ISipPresenceManager

Properties:

Property	Type	Description
Buddies	IReadOnlyList<ISipBuddy>	List of monitored buddies
MyState	BuddyState	My current presence state

Methods:

Method	Return	Description
AddBuddy(uri)	ISipBuddy	Add a buddy to monitor their presence
RemoveBuddy(buddy)	void	Stop monitoring a buddy
SetMyPresenceAsync(state, statusText, ct)	Task	Publish my presence state

Events:

Event	EventArgs	Description
BuddyStateChanged	BuddyStateChangedEventArgs	Buddy state change

ISipBuddy:

Property/Method	Type	Description
Uri	string	Buddy SIP URI
State	BuddyState	Current state
Info	BuddyInfo	Complete information (name, state, text, timestamp)
StateChanged	event	State change notification
SubscribeAsync(ct)	Task	Subscribe to presence notifications
UnsubscribeAsync(ct)	Task	Cancel subscription

BuddyState:

State	Description
Unknown	Unknown state
Online	Available
Away	Away
Busy	Busy
OnThePhone	On a call
Offline	Offline

---

ISipMessaging

Sending and receiving SIP messages (MESSAGE method, RFC 3428).

public interface ISipMessaging

Methods:

Method	Return	Description
SendMessageAsync(account, destUri, body, contentType, ct)	Task	Send a SIP message. contentType default: "text/plain"

Events:

Event	EventArgs	Description
MessageReceived	SipMessageReceivedEventArgs	Message received (contains SipMessage)
MessageStatus	SipMessageStatusEventArgs	Delivery status of sent message

SipMessage:

Property	Type	Description
From	string	Sender URI
To	string	Recipient URI
Body	string	Message body
ContentType	string	Content type (default: "text/plain")
Timestamp	DateTime	Message time (UTC)

---

ISipConferenceBridge

Conference bridge for mixing audio from multiple calls.

public interface ISipConferenceBridge

Properties:

Property	Type	Description
Participants	IReadOnlyList<ISipCall>	Calls currently in conference

Methods:

Method	Return	Description
AddParticipant(call)	void	Add a call to conference
RemoveParticipant(call)	void	Remove a call from conference
MergeAll(calls)	void	Merge multiple calls into one conference
SplitAll()	void	Split all calls from conference

---

ISipCallRecorder

Call recording to file.

public interface ISipCallRecorder : IDisposable

Properties:

Property	Type	Description
IsRecording	bool	If recording is in progress
CurrentFilePath	string?	Current file path, null if not recording

Methods:

Method	Return	Description
StartRecording(call, filePath, format)	void	Start recording. format default: Wav
StopRecording()	void	Stop current recording

Events:

Event	EventArgs	Description
RecordingStateChanged	RecordingStateChangedEventArgs	Recording state change

RecordingFormat:

Value	Description
Wav	Uncompressed WAV format

---

ISipToneGenerator

Signaling tone generator.

public interface ISipToneGenerator : IDisposable

Properties:

Property	Type	Description
IsPlaying	bool	If a tone is playing

Methods:

Method	Return	Description
PlayTone(tone)	void	Play a predefined tone type
PlayTones(tones)	void	Play a custom tone sequence
PlayRingbackTone()	void	Ringback tone (North American: 440+480 Hz)
PlayBusyTone()	void	Busy tone (480+620 Hz)
PlayDialTone()	void	Dial tone (350+440 Hz)
Stop()	void	Stop current tone

ToneDescriptor (for custom tones):

Property	Type	Description
Frequency1	int	First frequency in Hz
Frequency2	int	Second frequency in Hz (0 = single tone)
OnMs	int	Tone duration in milliseconds
OffMs	int	Silence duration in milliseconds
Volume	int	Volume (default: 16000)

ToneType:

Value	Description
Ringback	Standard ringback tone
Busy	Busy tone
Dial	Dial tone
Custom	Custom tone

---

ISipCallQualityMonitor

Call quality monitoring: RTP statistics, jitter, packet loss and MOS score.

public interface ISipCallQualityMonitor

Methods:

Method	Return	Description
GetQuality(call)	CallQualityInfo?	Get current call quality (synchronous)
GetQualityAsync(call, ct)	Task<CallQualityInfo?>	Get current quality (asynchronous, thread-safe)

Events:

Event	EventArgs	Description
QualityReportAvailable	CallQualityEventArgs	Quality report available

CallQualityInfo:

Property	Type	Description
CallId	string	Call ID
Duration	TimeSpan	Call duration at measurement time
RtpPacketsSent	long	Total RTP packets sent
RtpPacketsReceived	long	Total RTP packets received
RtpPacketsLost	long	Lost RTP packets
RtpLossPercentage	double	Packet loss percentage
RtpJitterMs	int	Jitter in milliseconds
RtpRoundTripTimeMs	int	Round-trip time in milliseconds
CodecName	string?	Active codec in call
CodecClockRate	int	Active codec clock rate
MosScore	double	Estimated Mean Opinion Score (1.0 — 5.0)

---

ISipCallHistory

Call history with filtering by type.

public interface ISipCallHistory

Properties:

Property	Type	Description
Entries	IReadOnlyList<CallHistoryEntry>	All history entries

Methods:

Method	Return	Description
GetMissedCalls()	IReadOnlyList<CallHistoryEntry>	Unanswered incoming calls
GetIncomingCalls()	IReadOnlyList<CallHistoryEntry>	All incoming calls
GetOutgoingCalls()	IReadOnlyList<CallHistoryEntry>	All outgoing calls
Clear()	void	Clear history

Events:

Event	EventArgs	Description
EntryAdded	CallHistoryEntry	New entry added to history

CallHistoryEntry:

Property	Type	Description
CallId	string	Call ID
RemoteUri	string	Remote party URI
RemoteDisplayName	string?	Remote party display name
Direction	CallDirection	Incoming or Outgoing
StartTime	DateTime	Start time
EndTime	DateTime?	End time
Duration	TimeSpan	Call duration
FinalState	SipCallState	Final call state
StatusCode	int	Final SIP code
AccountUri	string?	Local account URI

---

ISipNetworkMonitor

Network change monitoring to automatically re-register accounts.

public interface ISipNetworkMonitor : IDisposable

Properties:

Property	Type	Description
CurrentState	NetworkState	Current network state

Methods:

Method	Return	Description
HandleNetworkChangeAsync(ct)	Task	Manually notify network change (re-registers accounts, restarts transports)

Events:

Event	EventArgs	Description
NetworkStateChanged	NetworkStateChangedEventArgs	Network state change

NetworkState:

State	Description
Connected	Network connected
Disconnected	No network connectivity
Changed	Network changed (new IP, WiFi/data change)

---

🔔 Events

All events are fired on the thread that processed the PJSIP callback. In UI applications (WinForms/WPF/MAUI), use the corresponding dispatcher to update the interface.

In ISipPhone (global level)

// Incoming call on any account
phone.IncomingCall += (sender, e) =>
{
    Console.WriteLine($"Call from {e.RemoteDisplayName} <{e.RemoteUri}>");
    Console.WriteLine($"Target account: {e.Account.Uri}");

    e.Call.Answer();           // Answer
    // or: e.Call.Hangup(486);  // Reject as busy
};

// State change in any call
phone.CallStateChanged += (sender, e) =>
{
    Console.WriteLine($"Call {e.Call.Id}: {e.OldState} -> {e.NewState}");

    if (e.NewState == SipCallState.Disconnected)
        Console.WriteLine("Call ended");
};

// Registration change in any account
phone.RegistrationStateChanged += (sender, e) =>
{
    Console.WriteLine($"Account {e.Account.Uri}: {e.OldState} -> {e.NewState}");

    if (e.NewState == SipRegistrationState.Error)
        Console.WriteLine($"Registration error: {e.StatusCode} {e.Reason}");
};

// Transport state change
phone.TransportStateChanged += (sender, e) =>
{
    Console.WriteLine($"Transport {e.TransportType}: {e.State}");
};

// Message Waiting Indicator (voicemail)
phone.MwiStateChanged += (sender, e) =>
{
    Console.WriteLine($"Account {e.Account.Uri}: {e.MwiInfo.NewMessages} new message(s)");
};

In ISipAccount (account level)

var account = phone.Accounts[0];

account.RegistrationStateChanged += (sender, e) =>
    Console.WriteLine($"My account: {e.NewState}");

account.IncomingCall += (sender, e) =>
    Console.WriteLine($"Incoming call for this account: {e.RemoteUri}");

account.MwiStateChanged += (sender, e) =>
    Console.WriteLine($"Voicemail: {e.MwiInfo.NewMessages} new, {e.MwiInfo.OldMessages} old");

In ISipCall (call level)

var call = phone.MakeCall(account, "sip:1002@pbx.com");

call.StateChanged += (sender, e) =>
{
    Console.WriteLine($"State: {e.OldState} -> {e.NewState}");

    if (e.NewState == SipCallState.Confirmed)
        Console.WriteLine("Audio active!");
};

call.MediaStateChanged += (sender, e) =>
{
    Console.WriteLine($"Media active: {e.IsActive}");
};

---

🧩 Advanced Features

👥 Presence and BLF

Monitoring presence state of other users (Busy Lamp Field).

var presence = phone.Presence;

// Publish my state
await presence.SetMyPresenceAsync(BuddyState.Online, "Available");

// Add a buddy to monitor
var buddy = presence.AddBuddy("sip:1002@pbx.com");
await buddy.SubscribeAsync();

// Listen to state changes
buddy.StateChanged += (s, e) =>
    Console.WriteLine($"{buddy.Uri}: {e.OldState} -> {e.NewState}");

// Also at global level
presence.BuddyStateChanged += (s, e) =>
    Console.WriteLine($"Buddy {e.Buddy.Uri}: {e.NewState}");

// Query current state
Console.WriteLine($"Current state: {buddy.State}");
Console.WriteLine($"Last updated: {buddy.Info.LastUpdated}");

// Stop monitoring
await buddy.UnsubscribeAsync();
presence.RemoveBuddy(buddy);

💬 SIP Messaging (MESSAGE)

Sending and receiving text messages via SIP MESSAGE (RFC 3428).

var messaging = phone.Messaging;

// Send a message
await messaging.SendMessageAsync(
    phone.Accounts[0],
    "sip:1002@pbx.com",
    "Hello, are you available for a call?"
);

// Receive messages
messaging.MessageReceived += (s, e) =>
    Console.WriteLine($"Message from {e.Message.From}: {e.Message.Body}");

// Delivery status
messaging.MessageStatus += (s, e) =>
    Console.WriteLine($"Message to {e.DestinationUri}: code {e.StatusCode}");

🔀 Conference

Mixing audio from multiple calls into a conference.

var conference = phone.Conference;
var account = phone.Accounts[0];

// Create calls
var call1 = phone.MakeCall(account, "sip:1002@pbx.com");
var call2 = phone.MakeCall(account, "sip:1003@pbx.com");

// Wait until connected, then merge
conference.AddParticipant(call1);
conference.AddParticipant(call2);

// View participants
Console.WriteLine($"Participants: {conference.Participants.Count}");

// Merge all at once
conference.MergeAll(new[] { call1, call2 });

// Split all
conference.SplitAll();

// Remove one
conference.RemoveParticipant(call2);

🎙️ Call Recording

Recording call audio to file.

var recorder = phone.Recorder;

// Start recording
recorder.StartRecording(call, @"C:\recordings\call-001.wav");

// Check status
Console.WriteLine($"Recording: {recorder.IsRecording}");
Console.WriteLine($"File: {recorder.CurrentFilePath}");

// Listen to state changes
recorder.RecordingStateChanged += (s, e) =>
    Console.WriteLine($"Recording: {(e.IsRecording ? "started" : "stopped")}");

// Stop recording
recorder.StopRecording();

🎵 Tone Generator

Playing standard or custom signaling tones.

var tones = phone.Tones;

// Standard tones (North American frequencies)
tones.PlayDialTone();      // 350+440 Hz continuous
tones.PlayRingbackTone();  // 440+480 Hz, 2s on / 4s off
tones.PlayBusyTone();      // 480+620 Hz, 0.5s on / 0.5s off

// Generic tone by type
tones.PlayTone(ToneType.Busy);

// Custom tones
tones.PlayTones(new[]
{
    new ToneDescriptor { Frequency1 = 941, Frequency2 = 1336, OnMs = 100, OffMs = 100 },  // '#' key
    new ToneDescriptor { Frequency1 = 697, Frequency2 = 1209, OnMs = 100, OffMs = 100 },  // '1' key
});

// Stop
tones.Stop();
Console.WriteLine($"Playing: {tones.IsPlaying}");

📊 Call Quality

Monitoring RTP statistics and MOS score during an active call.

var quality = phone.Quality;

// Query quality of an active call
var info = quality.GetQuality(call);
if (info != null)
{
    Console.WriteLine($"Codec: {info.CodecName}");
    Console.WriteLine($"Packets sent: {info.RtpPacketsSent}");
    Console.WriteLine($"Loss: {info.RtpLossPercentage:F1}%");
    Console.WriteLine($"Jitter: {info.RtpJitterMs}ms");
    Console.WriteLine($"RTT: {info.RtpRoundTripTimeMs}ms");
    Console.WriteLine($"MOS: {info.MosScore:F1}/5.0");
}

// Or asynchronously (thread-safe)
var asyncInfo = await quality.GetQualityAsync(call);

// Listen to periodic reports
quality.QualityReportAvailable += (s, e) =>
    Console.WriteLine($"Call {e.Call.Id}: MOS={e.Quality.MosScore:F1}");

📋 Call History

Automatic call history with filtering.

var history = phone.History;

// History is filled automatically when a call disconnects

// Query all entries
foreach (var entry in history.Entries)
{
    Console.WriteLine($"[{entry.Direction}] {entry.RemoteUri} - " +
                      $"{entry.Duration:mm\\:ss} - {entry.FinalState}");
}

// Filter by type
var missed = history.GetMissedCalls();
var incoming = history.GetIncomingCalls();
var outgoing = history.GetOutgoingCalls();

Console.WriteLine($"Missed: {missed.Count}");
Console.WriteLine($"Incoming: {incoming.Count}");
Console.WriteLine($"Outgoing: {outgoing.Count}");

// Listen to new entries
history.EntryAdded += (s, entry) =>
    Console.WriteLine($"New entry: {entry.RemoteUri} ({entry.Direction})");

// Clear history
history.Clear();

Maximum history size is configured with SipPhoneOptions.CallHistoryMaxEntries (default: 1000).

🔕 Do Not Disturb (DND)

Control incoming call behavior per account.

var account = phone.Accounts[0];

// Activate DND — reject all calls
account.DndMode = DndMode.RejectAll;

// Reject with busy signal (486 Busy Here)
account.DndMode = DndMode.RejectWithBusy;

// Silent ring (call arrives but without tone)
account.DndMode = DndMode.SilentRing;

// Disable DND
account.DndMode = DndMode.Off;

DND modes (DndMode):

Mode	Description
Off	Disabled — normal behavior
RejectAll	Rejects all incoming calls (603 Decline)
RejectWithBusy	Rejects with busy signal (486 Busy Here)
SilentRing	Call arrives but doesn't ring (silent ring)

↪️ Call Forwarding

Configure call forwarding for an account.

var account = phone.Accounts[0];

// Unconditional forwarding
account.CallForwarding.Enabled = true;
account.CallForwarding.Type = CallForwardingType.Unconditional;
account.CallForwarding.DestinationUri = "sip:1003@pbx.com";

// Forward on no answer (after 20 seconds)
account.CallForwarding.Type = CallForwardingType.OnNoAnswer;
account.CallForwarding.NoAnswerTimeout = TimeSpan.FromSeconds(20);

// Forward on busy
account.CallForwarding.Type = CallForwardingType.OnBusy;

// Disable
account.CallForwarding.Enabled = false;

Forwarding types (CallForwardingType):

Type	Description
Unconditional	Forwards all calls immediately
OnBusy	Forwards if account is busy
OnNoAnswer	Forwards if not answered within configured timeout
OnNotReachable	Forwards if account is not available

📬 Message Waiting Indicator (MWI)

Receiving voicemail notifications.

var account = phone.Accounts[0];

// Listen to MWI changes at account level
account.MwiStateChanged += (s, e) =>
{
    Console.WriteLine($"Voicemail updated:");
    Console.WriteLine($"  Has waiting: {e.MwiInfo.HasWaiting}");
    Console.WriteLine($"  New: {e.MwiInfo.NewMessages}");
    Console.WriteLine($"  Old: {e.MwiInfo.OldMessages}");
    Console.WriteLine($"  New urgent: {e.MwiInfo.NewUrgentMessages}");
    Console.WriteLine($"  Old urgent: {e.MwiInfo.OldUrgentMessages}");
};

// Or at global level
phone.MwiStateChanged += (s, e) =>
    Console.WriteLine($"Account {e.Account.Uri}: {e.MwiInfo.NewMessages} new");

// Query current state (null if no notification received yet)
var mwi = account.MwiInfo;
if (mwi != null && mwi.HasWaiting)
    Console.WriteLine($"You have {mwi.NewMessages} voice message(s)");

MwiInfo:

Property	Type	Description
HasWaiting	bool	If messages are waiting
NewMessages	int	Number of new messages
OldMessages	int	Number of already listened messages
NewUrgentMessages	int	New urgent messages
OldUrgentMessages	int	Already listened urgent messages
AccountUri	string?	Associated account URI

🏷️ Custom Headers

Sending custom SIP headers in calls.

using PjSip.Net.Calls;

var headers = new[]
{
    new SipHeader { Name = "X-Tenant-Id", Value = "acme-corp" },
    new SipHeader { Name = "X-Call-Tag", Value = "support-level2" }
};

// In MakeCall
var call = phone.MakeCall(account, "sip:1002@pbx.com", headers);

// Or from account
var call2 = account.MakeCall("sip:1003@pbx.com", headers);

// Read headers from a call
foreach (var h in call.CustomHeaders)
    Console.WriteLine($"{h.Name}: {h.Value}");

// When answering with headers
call.Answer(200, new[]
{
    new SipHeader { Name = "X-Agent-Id", Value = "42" }
});

🤝 Attended Transfer

Connecting two active calls (transfer with prior consultation).

sequenceDiagram
    participant Agent
    participant Customer
    participant Specialist

    Agent->>Customer: Active call
    Agent->>Agent: Hold(customer)
    Agent->>Specialist: MakeCall()
    Note over Agent,Specialist: Consultation
    Agent->>Customer: AttendedTransfer(specialist)
    Customer-->>Specialist: Connected directly
    Note over Agent: Both calls disconnect

// Active call with customer
var callCustomer = phone.MakeCall(account, "sip:customer@example.com");
// ... customer is on the line ...

// Put customer on hold
callCustomer.Hold();

// Call specialist for consultation
var callSpecialist = phone.MakeCall(account, "sip:specialist@example.com");
// ... talk to specialist ...

// Connect customer with specialist (attended transfer)
callCustomer.AttendedTransfer(callSpecialist);
// Both calls disconnect from agent; customer and specialist remain connected

🌐 Network Monitor

Detecting network changes and automatically re-registering accounts.

var network = phone.Network;

// Current state
Console.WriteLine($"Network: {network.CurrentState}");

// Listen to changes
network.NetworkStateChanged += (s, e) =>
{
    Console.WriteLine($"Network changed: {e.OldState} -> {e.NewState}");

    if (e.NewState == NetworkState.Disconnected)
        Console.WriteLine("No network connectivity");
};

// Manually notify network change (e.g., from OS events)
await network.HandleNetworkChangeAsync();

---

📱 Platform Examples

Console App

using Microsoft.Extensions.DependencyInjection;
using PjSip.Net;
using PjSip.Net.Accounts;
using PjSip.Net.DependencyInjection;
using PjSip.Net.Transport;

var services = new ServiceCollection();
services.AddLogging();
services.AddPjSip(options =>
{
    options.Transports.Add(new SipTransportOptions
    {
        Type = SipTransportType.Tls,
        Port = 5061
    });
    options.Accounts.Add(new SipAccountOptions
    {
        Username = "1001",
        Password = "secret",
        Domain = "pbx.mycompany.com",
        Registrar = "sip:pbx.mycompany.com"
    });
});

var provider = services.BuildServiceProvider();
var phone = provider.GetRequiredService<ISipPhone>();

phone.IncomingCall += (s, e) =>
{
    Console.WriteLine($"Call from {e.RemoteUri}");
    e.Call.Answer();
};

await phone.StartAsync();
Console.WriteLine("Phone active. Press Enter to exit...");
Console.ReadLine();
await phone.StopAsync();

WPF

// In App.xaml.cs or with a HostBuilder
public partial class App : Application
{
    private ServiceProvider? _serviceProvider;

    protected override async void OnStartup(StartupEventArgs e)
    {
        var services = new ServiceCollection();
        services.AddLogging();
        services.AddPjSip(options =>
        {
            options.Transports.Add(new SipTransportOptions
            {
                Type = SipTransportType.Tls,
                Port = 5061
            });
            options.Accounts.Add(new SipAccountOptions
            {
                Username = "1001",
                Password = "secret",
                Domain = "pbx.mycompany.com"
            });
        });

        _serviceProvider = services.BuildServiceProvider();
        var mainWindow = new MainWindow(_serviceProvider.GetRequiredService<ISipPhone>());
        mainWindow.Show();
    }
}

// In MainWindow.xaml.cs
public partial class MainWindow : Window
{
    private readonly ISipPhone _phone;
    private ISipCall? _activeCall;

    public MainWindow(ISipPhone phone)
    {
        InitializeComponent();
        _phone = phone;

        // IMPORTANT: Use Dispatcher to update UI from SIP events
        _phone.IncomingCall += (s, e) =>
            Dispatcher.Invoke(() =>
            {
                StatusText.Text = $"Incoming call from {e.RemoteDisplayName}";
                // Show accept/reject dialog
            });

        _phone.CallStateChanged += (s, e) =>
            Dispatcher.Invoke(() =>
                StatusText.Text = $"Call: {e.NewState}");
    }

    private async void OnStartClick(object sender, RoutedEventArgs e)
    {
        await _phone.StartAsync();
        StatusText.Text = $"Connected ({_phone.Accounts.Count} accounts)";
    }

    private void OnCallClick(object sender, RoutedEventArgs e)
    {
        _activeCall = _phone.MakeCall(_phone.Accounts[0], DestinationBox.Text);
    }

    private void OnHangupClick(object sender, RoutedEventArgs e)
    {
        _activeCall?.Hangup();
        _activeCall = null;
    }
}

WinForms

public partial class MainForm : Form
{
    private ISipPhone? _phone;

    private async Task StartPhoneAsync()
    {
        var services = new ServiceCollection();
        services.AddLogging();
        services.AddPjSip(options =>
        {
            options.Transports.Add(new SipTransportOptions { Type = SipTransportType.Udp });
            options.Accounts.Add(new SipAccountOptions
            {
                Username = "1001",
                Password = "secret",
                Domain = "pbx.mycompany.com"
            });
        });

        var provider = services.BuildServiceProvider();
        _phone = provider.GetRequiredService<ISipPhone>();

        // IMPORTANT: Use BeginInvoke to update UI
        _phone.IncomingCall += (s, e) =>
            BeginInvoke(() =>
                MessageBox.Show($"Call from {e.RemoteUri}", "Incoming Call"));

        _phone.CallStateChanged += (s, e) =>
            BeginInvoke(() =>
                lblStatus.Text = $"Call: {e.NewState}");

        await _phone.StartAsync();
        lblStatus.Text = "Connected";
    }

    private void btnCall_Click(object sender, EventArgs e)
    {
        _phone?.MakeCall(_phone.Accounts[0], txtDestination.Text);
    }
}

MAUI

// In MauiProgram.cs
public static class MauiProgram
{
    public static MauiApp CreateMauiApp()
    {
        var builder = MauiApp.CreateBuilder();
        builder.UseMauiApp<App>();

        builder.Services.AddPjSip(options =>
        {
            options.Transports.Add(new SipTransportOptions
            {
                Type = SipTransportType.Udp
            });
            options.Accounts.Add(new SipAccountOptions
            {
                Username = "1001",
                Password = "secret",
                Domain = "pbx.mycompany.com"
            });
        });

        return builder.Build();
    }
}

// In a page
public partial class PhonePage : ContentPage
{
    private readonly ISipPhone _phone;

    public PhonePage(ISipPhone phone)
    {
        InitializeComponent();
        _phone = phone;

        // MAUI: Use MainThread.BeginInvokeOnMainThread for UI
        _phone.IncomingCall += (s, e) =>
            MainThread.BeginInvokeOnMainThread(() =>
                StatusLabel.Text = $"Call from {e.RemoteUri}");
    }

    private async void OnStartClicked(object? sender, EventArgs e)
    {
        await _phone.StartAsync();
        StatusLabel.Text = "Connected";
    }

    private void OnCallClicked(object? sender, EventArgs e)
    {
        _phone.MakeCall(_phone.Accounts[0], DestinationEntry.Text);
    }
}

---

⚠️ Error Handling

The SDK defines specific exceptions for SIP errors:

using PjSip.Net.Exceptions;

try
{
    await phone.StartAsync();
}
catch (SipTransportException ex)
{
    // Transport creation error (port busy, misconfigured TLS, etc.)
    Console.WriteLine($"Transport error: {ex.Message} (PJSIP code: {ex.PjStatusCode})");
}
catch (PjSipException ex)
{
    // Generic PJSIP error
    Console.WriteLine($"PJSIP error: {ex.Message} (code: {ex.PjStatusCode})");
}

// Registration errors are notified via event
phone.RegistrationStateChanged += (s, e) =>
{
    if (e.NewState == SipRegistrationState.Error)
    {
        // e.StatusCode contains SIP code (401, 403, 408, etc.)
        Console.WriteLine($"Registration error: {e.StatusCode} - {e.Reason}");
    }
};

Exception hierarchy:

PjSipException                    Base — any PJSIP error
├── SipRegistrationException      REGISTER error (4xx, 5xx)
└── SipTransportException         Transport error (bind, TLS, network)

---

🎧 Audio

var audio = phone.Audio;

// List devices
var microphones = audio.GetInputDevices();
var speakers = audio.GetOutputDevices();

foreach (var mic in microphones)
    Console.WriteLine($"[{mic.DeviceId}] {mic.Name} ({mic.InputChannels}ch)");

foreach (var spk in speakers)
    Console.WriteLine($"[{spk.DeviceId}] {spk.Name} ({spk.OutputChannels}ch)");

// Change device
audio.SetInputDevice(microphones[1].DeviceId);
audio.SetOutputDevice(speakers[0].DeviceId);

// Adjust volume (0.0 = silence, 1.0 = maximum)
audio.InputLevel = 0.8f;    // Microphone at 80%
audio.OutputLevel = 1.0f;   // Speaker at 100%

// Mute a specific call
call.SetMute(true);   // Mute microphone for this call
call.SetMute(false);  // Unmute

---

🔧 Low-Level Access (pjsua2)

For advanced scenarios requiring direct access to pjsua2 classes generated by SWIG:

// SWIG classes are in the PjSip.Net.Interop.Generated namespace
using PjSip.Net.Interop.Generated;

// Example: access native endpoint directly
// (available once SWIG wrappers are generated)

Note: Low-level access requires knowledge of the pjsua2 API. See the official PJSIP documentation.

---

🏗️ Architecture

graph TB
    subgraph App["Your Application"]
        WF[WinForms]
        WPF[WPF]
        MAUI[MAUI]
        CON[Console]
    end

    subgraph SDK["PjSip.Net — High-level SDK"]
        Phone[ISipPhone]
        Account[ISipAccount]
        Call[ISipCall]
        Audio[ISipAudioManager]
        Codec[ISipCodecManager]
        Presence[ISipPresenceManager]
        Msg[ISipMessaging]
        Conf[ISipConferenceBridge]
        Rec[ISipCallRecorder]
        Tone[ISipToneGenerator]
        QoS[ISipCallQualityMonitor]
        Hist[ISipCallHistory]
        Net[ISipNetworkMonitor]
        DI["DI (AddPjSip)"]
        Events["Events"]
    end

    subgraph Interop["PjSip.Net.Interop"]
        Loader[NativeLoader]
        SWIG["Generated/ (SWIG C#)"]
    end

    subgraph Native["PjSip.Net.Native.{Platform}"]
        Win["Win64 — pjsua2.dll"]
        Mac["MacOS — libpjsua2.dylib"]
        And["Android — libpjsua2.so"]
        iOS["iOS — libpjsua2.dylib"]
    end

    App --> SDK
    SDK --> Interop
    Interop --> Native

    style App fill:#e1f5fe
    style SDK fill:#f3e5f5
    style Interop fill:#fff3e0
    style Native fill:#e8f5e9

Design Patterns used:

Pattern	Use
Facade	ISipPhone as single entry point with 11 sub-managers
Options	SipPhoneOptions, SipAccountOptions, NatOptions via IOptions<T>
Observer	.NET events (IncomingCall, CallStateChanged, BuddyStateChanged, etc.)
Factory	AddAccount(), MakeCall(), AddBuddy()
Adapter	ManagedAccount/ManagedCall/ManagedBuddy adapt pjsua2 callbacks to .NET events
Dispose	Cascading cleanup of native resources

---

📱 Supported Platforms

Platform	RID	TLS Backend	Native Package
<img src="https://img.shields.io/badge/Windows-0078D6?logo=windows&logoColor=white" alt="Windows" />	win-x64	Schannel	PjSip.Net.Native.Win64
<img src="https://img.shields.io/badge/macOS-000000?logo=apple&logoColor=white" alt="macOS" /> x64	osx-x64	Apple SSL	PjSip.Net.Native.MacOS
<img src="https://img.shields.io/badge/macOS-000000?logo=apple&logoColor=white" alt="macOS" /> ARM64	osx-arm64	Apple SSL	PjSip.Net.Native.MacOS
<img src="https://img.shields.io/badge/Mac_Catalyst-000000?logo=apple&logoColor=white" alt="Mac Catalyst" />	osx-arm64 / osx-x64	Apple SSL	PjSip.Net.Native.MacOS
<img src="https://img.shields.io/badge/Android-3DDC84?logo=android&logoColor=white" alt="Android" /> ARM64	android-arm64	OpenSSL 3.4.1	PjSip.Net.Native.Android
<img src="https://img.shields.io/badge/iOS-000000?logo=ios&logoColor=white" alt="iOS" /> ARM64	ios-arm64	Secure Transport	PjSip.Net.Native.iOS

---

🛠️ Build from Source

Prerequisites

• .NET 10 SDK
• Visual Studio 2022 with C++ workload (to compile pjsua2 on Windows)
• SWIG 4.0+ (to generate C# wrappers)

Compile managed solution

dotnet build PjSip.Net.slnx
dotnet test tests/PjSip.Net.Tests.Unit/PjSip.Net.Tests.Unit.csproj

Compile native binaries

# Windows x64 (PowerShell)
./native/build-win64.ps1

# macOS (bash)
./native/build-macos.sh

# Mobile (bash)
./native/build-android.sh
./native/build-ios.sh

Create NuGet packages

dotnet pack src/PjSip.Net/PjSip.Net.csproj -o ./artifacts
dotnet pack src/PjSip.Net.Interop/PjSip.Net.Interop.csproj -o ./artifacts
dotnet pack src/PjSip.Net.Native.Win64/PjSip.Net.Native.Win64.csproj -o ./artifacts

---

📝 Changelog

v1.0.3

• Feature: Account-specific buddy subscriptions — AddBuddy(uri, account) overload allows binding presence subscriptions to a specific SIP account instead of always using the primary account

v1.0.2

• Fix: Populate RemoteDisplayName on incoming calls — the display name from the SIP From header is now correctly extracted when the call is first created
• Fix: Preserve RemoteDisplayName across call state updates — on onCallState callbacks, the display name is retained even if the native CallInfo no longer includes it

v1.0.1

• Feature: Outbound proxy support (SipAccountOptions.OutboundProxy)
• Feature: TLS transport suffix — automatic ;transport=tls append via SipAccountOptions.UseTls

v1.0.0

• First stable production release

---

📄 License

MIT