mediasoup 源码分析（八）分析PlainTransport

mediasoup 源码分析（六）分析PlainTransport

- - - 一、接收裸RTP流
    - 二、mediasoup 中udp建立过程
- tips

一、接收裸RTP流

PlainTransport 可以接收裸RTP流，也可以接收AES加密的RTP流。源码中提供了一个通过ffmpeg发送裸RTP流到mediasoup的脚本，具体地址为：mediasoup-demo/broadcasters/ffmpeg.sh
脚本就是通过HTTP Post发送创建PlainTranport请求，然后通过ffmpeg向指定地址+端口，发送RTP流

res=$(${HTTPIE_COMMAND} \
	POST ${SERVER_URL}/rooms/${ROOM_ID}/broadcasters/${BROADCASTER_ID}/transports \
	type="plain" \
	comedia:=true \
	rtcpMux:=false \
	2> /dev/null)

ffmpeg发送RTP流

#
# NOTES:
# - We can add ?pkt_size=1200 to each rtp:// URI to limit the max packet size
#   to 1200 bytes.
#
ffmpeg \
	-re \
	-v info \
	-stream_loop -1 \
	-i ${MEDIA_FILE} \
	-map 0:a:0 \
	-acodec libopus -ab 128k -ac 2 -ar 48000 \
	-map 0:v:0 \
	-pix_fmt yuv420p -c:v libvpx -b:v 1000k -deadline realtime -cpu-used 4 \
	-f tee \
	"[select=a:f=rtp:ssrc=${AUDIO_SSRC}:payload_type=${AUDIO_PT}]rtp://${audioTransportIp}:${audioTransportPort}?rtcpport=${audioTransportRtcpPort}|[select=v:f=rtp:ssrc=${VIDEO_SSRC}:payload_type=${VIDEO_PT}]rtp://${videoTransportIp}:${videoTransportPort}?rtcpport=${videoTransportRtcpPort}"

二、mediasoup 中udp建立过程

RTP:UdpSocket的构造函数，开启IP绑定

namespace RTC
{
	/* Instance methods. */

	UdpSocket::UdpSocket(Listener* listener, std::string& ip)
	  : // This may throw.
	    ::UdpSocket::UdpSocket(PortManager::BindUdp(ip)), listener(listener)
	{
		MS_TRACE();
	}

真正绑定的地方在PortManager::BindUdp(ip)中，随机从指定Port范围中选出可用Port，然后绑定

// Choose a random port index to start from.
   portIdx = static_cast<size_t>(Utils::Crypto::GetRandomUInt(
   static_cast<uint32_t>(0), static_cast<uint32_t>(ports.size() - 1)));
   .....
   .....
   // Increase current port index.
   portIdx = (portIdx + 1) % ports.size();

   // So the corresponding port is the vector position plus the RTC minimum port.
   port = static_cast<uint16_t>(portIdx + Settings::configuration.rtcMinPort);
   .....
   //udp绑定
   switch (transport)
   {
	  case Transport::UDP:
	  {
		 err = uv_udp_bind(
		 reinterpret_cast<uv_udp_t*>(uvHandle),
		 reinterpret_cast<const struct sockaddr*>(&bindAddr),flags);
      }

三、全局的UdpSocket绑定接收

UdpSocket::UdpSocket(uv_udp_t* uvHandle) : uvHandle(uvHandle)
{
	MS_TRACE();

	int err;

	this->uvHandle->data = static_cast<void*>(this);

	err = uv_udp_recv_start(
	  this->uvHandle, static_cast<uv_alloc_cb>(onAlloc), static_cast<uv_udp_recv_cb>(onRecv)); // 接收回调函数
}


inline static void onRecv(
  uv_udp_t* handle, ssize_t nread, const uv_buf_t* buf, const struct sockaddr* addr, unsigned int flags)
{
	auto* socket = static_cast<UdpSocket*>(handle->data);

	if (socket)
		socket->OnUvRecv(nread, buf, addr, flags);
}
//然后回调到RTC::UdpSocket::UserOnUdpDatagramReceived的函数
	void UdpSocket::UserOnUdpDatagramReceived(const uint8_t* data, size_t len, const struct sockaddr* addr)
	{
		MS_TRACE();

		if (this->listener == nullptr)
		{
			MS_ERROR("no listener set");

			return;
		}

		// Notify the reader.此处的Listener就是PlainTransport
		this->listener->OnUdpSocketPacketReceived(this, data, len, addr);
	}

//PainTransport接收到数据
	inline void PlainTransport::OnUdpSocketPacketReceived(
	  RTC::UdpSocket* socket, const uint8_t* data, size_t len, const struct sockaddr* remoteAddr)
	{
		MS_TRACE();
        //TranportTuple维护socket与客户端地址的关联关系
		RTC::TransportTuple tuple(socket, remoteAddr);

		OnPacketReceived(&tuple, data, len);
	}
//处理接收到的udp包
inline void PlainTransport::OnPacketReceived(RTC::TransportTuple* tuple, const uint8_t* data, size_t len)
	{
		MS_TRACE();

		// Increase receive transmission.
		RTC::Transport::DataReceived(len);

		// Check if it's RTCP.
		if (RTC::RTCP::Packet::IsRtcp(data, len))
		{
			OnRtcpDataReceived(tuple, data, len);
		}
		// Check if it's RTP.
		else if (RTC::RtpPacket::IsRtp(data, len))
		{
			OnRtpDataReceived(tuple, data, len);
		}
		// Check if it's SCTP.
		else if (RTC::SctpAssociation::IsSctp(data, len))
		{
			OnSctpDataReceived(tuple, data, len);
		}
		else
		{
			MS_WARN_DEV("ignoring received packet of unknown type");
		}
	}