///<summary> 创建 IceSocket
/// IceSocket 是可靠的TCP传输
///</summary>
///<param name="rtcConfiguration"> 服务器地址</param>
///<param name="rtcIceParameters"> 可选的来源标识</param>
///<param name="isCaller"> 是否为主叫</param>
///<param name="observer"> 事件侦听器</param>
static public IceSocket Connect(string rtcConfiguration, string rtcIceParameters, bool isCaller, RTCSocketObserver observer)
{
if (observer == null)
{
throw new ArgumentNullException();
}
int index = 1;
while (index < OBSERVERS.Length && OBSERVERS[index] != null)
{
++index;
}
if (index == OBSERVERS.Length)
{
throw new OverflowException();
}
OBSERVERS[index] = observer;
IntPtr iface = JSEP._CreateIceSocket(rtcConfiguration, rtcIceParameters, isCaller, (IntPtr)index, IceSocketCallback);
if (iface == IntPtr.Zero)
{
OBSERVERS[index] = null;
throw new ArgumentException();
}
IceSocket self = new IceSocket();
SOCKETS.Add(iface, self);
self.iface = iface;
return(self);
}
///<summary> 创建P2P通信实例
///</summary>
///<param name="config"> 配置参数,JSON格式. 例如{'iceServers': [{'url': url}]}</param>
///<param name="zmfAudioPump"> 是否使用内置的'全局音频泵',可与外部音频流对接, @see JSEP_AUDIO_PUMP</param>
///<param name="isCaller"> 是否为主叫</param>
///<param name="observer"> 事件侦听器</param>
///<remarks>
/// config 支持的配置参数
/// - STRUN 服务器地址
/// iceServers: [{
/// urls:'url'
/// username:''
/// credential:''
/// }]
/// - bundlePolicy:'balanced'
/// - rtcpMuxPolicy:'require' | 'negotiate'
/// - iceTransportPolicy:'all'
/// - zmfAudioPump: false //全局设置,使用JSEP内置的音频流泵,可与外部音频流对接
///
/// constraints 支持的限制
/// - googIPv6: true
/// - googDscp: true
/// - DtlsSrtpKeyAgreement: true
/// - RtpDataChannels: false
/// - googSuspendBelowMinBitrate: true
/// - googNumUnsignalledRecvStreams: 20
/// - googScreencastMinBitrate:int
/// - googHighStartBitrate:int
/// - googHighBitrate:true
/// - googVeryHighBitrate:true
/// - googCombinedAudioVideoBwe:true
///</remarks>
public RTCPeerConnection(string config, bool zmfAudioPump, bool isCaller, RTCSessionObserver observer)
{
if (observer == null)
{
throw new ArgumentNullException();
}
int index = 1;
while (index < OBSERVERS.Length && OBSERVERS[index] != null)
{
++index;
}
if (index == OBSERVERS.Length)
{
throw new OverflowException();
}
OBSERVERS[index] = observer;
iface = JSEP._CreatePeerConnection(config, zmfAudioPump, isCaller, (IntPtr)index, RTCSessionCallback);
if (iface == IntPtr.Zero)
{
OBSERVERS[index] = null;
throw new ArgumentException();
}
}
///<summary> 创建 WebSocket 服务器并侦听客户的连接
/// WebSocket 是可靠的TCP传输
///</summary>
///<param name="wsURL"> 服务器地址</param>
///<param name="rtcWebSocketInit"> 配置参数</param>
///<param name="observer"> 事件侦听器</param>
static public WebSocket Listen(string wsURL, string rtcWebSocketInit, RTCSocketObserver observer)
{
if (observer == null)
{
throw new ArgumentNullException();
}
int index = 1;
while (index < OBSERVERS.Length && OBSERVERS[index] != null)
{
++index;
}
if (index == OBSERVERS.Length)
{
throw new OverflowException();
}
OBSERVERS[index] = observer;
IntPtr iface = JSEP._CreateWebSocketServer(wsURL, rtcWebSocketInit, (IntPtr)index, WebSocketCallback);
if (iface == IntPtr.Zero)
{
OBSERVERS[index] = null;
throw new ArgumentException();
}
WebSocket self = new WebSocket();
SOCKETS.Add(iface, self);
self.iface = iface;
return(self);
}
///<summary> 销毁所有资源
///</summary>
///<param name="isDisposing"> 是否是主动丢弃</param>
protected virtual void Dispose(bool isDisposing)
{
if (iface != IntPtr.Zero)
{
OBSERVERS[(int)JSEP._ReleasePeerConnection(iface)] = null;
}
iface = IntPtr.Zero;
}
///<summary> 销毁所有资源
///</summary>
///<param name="isDisposing"> 是否是主动丢弃</param>
protected virtual void Dispose(bool isDisposing)
{
if (iface != IntPtr.Zero)
{
OBSERVERS[(int)JSEP._CloseSocket(iface)] = null;
SOCKETS.Remove(iface);
iface = IntPtr.Zero;
}
}
///<summary> 手工设置发送码率
///</summary>
///<param name="current_bitrate_bps"> 当前码率,若不大于0,则不设置</param>
///<param name="max_bitrate_bps"> 允许的最大码率,若不大于0,则不设置</param>
///<param name="min_bitrate_bps"> 允许的最小码率,若不大于0,则不设置</param>
///<returns> 成功返回0</returns>
public int SetBitrate(int current_bitrate_bps, int max_bitrate_bps, int min_bitrate_bps)
{
return(JSEP._SetBitrate(iface, current_bitrate_bps, max_bitrate_bps, min_bitrate_bps));
}
///<summary> 开始/停止记录通信事件
///</summary>
///<param name="filename"> 日志文件名,空或null则停止记录</param>
///<param name="max_size_mb"> 日志文件允许大小MB</param>
///<returns> 成功返回0</returns>
public int LogRtcEvent(string filename, int max_size_mb)
{
return(JSEP._LogRtcEvent(iface, Encoding.UTF8.GetBytes(filename), max_size_mb));
}
///<summary> 获取统计
///</summary>
///<param name="statsType"> 统计类别,""或null表示获取所有统计</param>
///<param name="bDebug"> 是否更详细的调试级别</param>
///<returns> 成功返回0, 将触发 JsepEventStatsReport 事件</returns>
public int GetStats(string statsType, bool bDebug)
{
return(JSEP._GetStats(iface, statsType, bDebug));
}
///<summary> 创建offer SDP
///</summary>
///<param name="constraints"> 可选的SDP的限制参数.JSON格式</param>
///<returns> 无效参数立即返回-1,反之异步过程,结果由JsepEventCreateDescription 事件返回</returns>
///<remarks>
/// constraints 支持如下限制
/// - OfferToReceiveAudio: true
/// - OfferToReceiveVideo: true
/// - VoiceActivityDetection: true
/// - IceRestart: false
/// - googUseRtpMUX: true
///</remarks>
public int CreateOffer(string constraints)
{
return(JSEP._CreateOffer(iface, constraints));
}
///<summary> 配置对端(发送)的SDP
///</summary>
///<param name="desc"> SDP的JSON串,通常对端由 JsepEventCreateDescription 事件的JsepSdp获得</param>
///<returns> 无效参数立即返回-1,反之异步过程,结果由JsepEventSetDescription 事件返回 </returns>
public int SetRemoteDescription(string desc)
{
return(JSEP._SetRemoteDescription(iface, desc));
}
///<summary> 移除本地媒体流
///</summary>
///<param name="streamId"> 媒体流ID,若""或null则移除所有本地流</param>
public void RemoveLocalStream(string streamId)
{
JSEP._RemoveLocalStream(iface, streamId);
}
///<summary> 添加对端的ICE备选地址
///</summary>
///<param name="candidate"> 备选地址,通常对端由 JsepEventIceCandidate 事件的JsepIceCandidate字段获得</param>
///<returns> 成功返回0 </returns>
public int AddIceCandidate(string candidate)
{
return(JSEP._AddIceCandidate(iface, candidate));
}
///<summary> 发送自定义数据
///</summary>
///<param name="channelId"> 通道ID</param>
///<param name="message"> 消息 </param>
///<returns> 成功返回0 </returns>
public int SendMessage(string channelId, string message)
{
byte[] buffer = Encoding.UTF8.GetBytes(message);
return(JSEP._SendMessage(iface, channelId, buffer, buffer.Length));
}
///<summary> 设置对端的ICE参数
///</summary>
///<param name="rtcIceParameters">对端参数, @see RTCIceParameters</param>
///<returns> 成功返回0 </returns>
public int SetRemoteParameters(string rtcIceParameters)
{
return(JSEP._SetSocketIceParameters(iface, rtcIceParameters));
}
///<summary> 添加对端的ICE地址
///</summary>
///<param name="candidate">对端地址</param>
///<returns> 成功返回0 </returns>
public int AddRemoteCandidate(string candidate)
{
return(JSEP._AddSocketIceCandidate(iface, candidate));
}
///<summary> 发送DTMF
///</summary>
///<param name="tones"> DMTF的音符,0-9,A-D或a-d,#,*. 忽略无法识别的字符</param>
///<param name="duration"> 每个音符的持续毫秒MS,不能超过6000或小于70</param>
///<param name="inter_tone_gap"> 音符间隔,必须至少为50ms,但应尽可能短</param>
///<returns> 成功返回0,将触发 JsepEventToneChange 事件 </returns>
///<remarks>
/// 队列化发送DTMF 任务.
/// ','表示延迟2秒处理下一个字符
/// 若调用时,上次仍在运行,则之前的任务将被取消
///</remarks>
public int InsertDtmf(string tones, int duration, int inter_tone_gap)
{
return(JSEP._InsertDtmf(iface, tones, duration, inter_tone_gap));
}
///<summary> 将媒体流发布到ZMF中
///</summary>
///<param name="streamId"> 对端媒体流ID</param>
///<param name="renderOrCapturerBits"> 按位将视频轨道作为0-ZMF渲染,1-ZMF镜头</param>
///<param name="videoTrackMask"> 上个参数renderOrCapturerBits中的有效位掩码</param>
///<returns> 成功返回0 </returns>
public int PublishRemoteStream(string streamId, int renderOrCapturerBits, int videoTrackMask)
{
return(JSEP._PublishRemoteStream(iface, streamId, renderOrCapturerBits, videoTrackMask));
}
///<summary> 添加本地媒体流
///</summary>
///<param name="streamId"> 媒体流ID</param>
///<param name="bAudio"> 是否含音频</param>
///<param name="bVideo"> 是否含视频</param>
///<param name="constraints"> 媒体流的限制参数,JSON格式</param>
///<returns> 成功返回0,成功返回0, bAudio, bVideo分别指示是否含音视频 </returns>
///<remarks>
/// 本地媒体流通常在协商前添加,对端媒体流是自动添加的.
/// streamId 要保证唯一性,通常用账户ID.
///
/// constraints 目前支持
/// - 视频
/// video: {
/// zmfCapture:Id0 //ZMF采集源ID
/// zmfRender:Id0 //ZMF第三方渲染源ID,转接为采集
/// }
/// - 音频
/// audio: {
/// DTMF:false //不使用JSEP_InsertDtmf函数,可节省些内存.
/// }
///</remarks>
public int AddLocalStream(string streamId, ref bool bAudio, ref bool bVideo, string constraints)
{
return(JSEP._AddLocalStream(iface, streamId, ref bAudio, ref bVideo, constraints));
}
///<summary> 配置本地(接收)的SDP
///</summary>
///<param name="desc"> desc SDP的JSON串,通常对端由 JsepEventCreateDescription 事件的JsepSdp获得</param>
///<returns> 无效参数立即返回-1,反之异步过程,结果由JsepEventSetDescription 事件返回 </returns>
public int SetLocalDescription(string desc)
{
return(JSEP._SetLocalDescription(iface, desc));
}
///<summary> 关闭数据通道
///</summary>
///<param name="channelId"> 通道ID</param>
public void CloseDataChannel(string channelId)
{
JSEP._CloseDataChannel(iface, channelId);
}
///<summary> 发送数据
///</summary>
///<param name="message"> 消息 </param>
///<returns> 成功返回0 </returns>
public int Send(string message)
{
byte[] buffer = Encoding.UTF8.GetBytes(message);
return(JSEP._SendSocket(iface, buffer, buffer.Length, 0));
}
///<summary> 动态创建数据通道
///</summary>
///<param name="channelId"> 通道ID</param>
///<param name="constraints"> 配置参数</param>
///<returns> 成功返回0</returns>
///<remarks>
/// 支持的配置参数
/// - ordered:true, //是否保证次序,默认true
/// - maxPacketLifeTime:0,//超过限时,将不重发.默认-1,始终重发
/// - maxRetransmits:0, //超过次数,将不重发,默认-1,,始终重发
/// - negotiated:false, //是否由上层应用负责协商建立过程,
/// 即不触发DataChannelOpen事件.默认false由内部自动完成
/// - protocol:'', //自定义的上层应用协议名,默认空
///</remarks>
public int CreateDataChannel(string channelId, string constraints)
{
return(JSEP._CreateDataChannel(iface, channelId, constraints));
}
