RTSPSession.h

/*
 * Author: Phil Schatzmann
 *
 * Based on Micro-RTSP library:
 * https://github.com/Tomp0801/Micro-RTSP-Audio
 * https://github.com/geeksville/Micro-RTSP
 *
 */
 
#pragma once
 
#include <stdio.h>
 
#include <ctime>
 
#include "AudioTools/CoreAudio/AudioBasic/Collections/Vector.h"
#include "RTSPAudioStreamer.h"
#include "RTSPPlatform.h"
 
#define RTSP_BUFFER_SIZE 10000
#define RTSP_PARAM_STRING_MAX 100
#define MAX_HOSTNAME_LEN 256
#define RTSP_RESPONSE_BUFFER_SIZE 2251
#define RTSP_SDP_BUFFER_SIZE 1024
#define RTSP_URL_BUFFER_SIZE 1024
#define RTSP_SMALL_BUFFER_SIZE 256
 
namespace audio_tools {
 
enum RTSP_CMD_TYPES {
  RTSP_OPTIONS,
  RTSP_DESCRIBE,
  RTSP_SETUP,
  RTSP_PLAY,
  RTSP_PAUSE,
  RTSP_TEARDOWN,
  RTSP_UNKNOWN
};
 
template <typename Platform>
class RtspSession {
 public:
  RtspSession(typename Platform::TcpClientType& aClient,
              RTSPAudioStreamerBase<Platform>& aStreamer)
      : m_Client(aClient), m_Streamer(&aStreamer) {
    m_RtspClient = &m_Client;
    m_RtspSessionID = random(65536);  // create a session ID
    LOGI("RTSP session created");
  }
 
  ~RtspSession() {
    LOGI("RTSP session destructor");
 
    // Ensure streaming is stopped and resources are released
    if (m_streaming && m_Streamer) {
      LOGI("Final cleanup: stopping streamer in destructor");
      m_Streamer->stop();
      m_Streamer->releaseUdpTransport();
      m_streaming = false;
    }
 
    // Close the client socket (check connection using client method directly)
    if (m_RtspClient && m_RtspClient->connected()) {
      Platform::closeSocket(m_RtspClient);
    }
 
    LOGI("RTSP session cleanup completed");
  }
 
  bool handleRequests(uint32_t readTimeoutMs) {
    LOGD("handleRequests");
    // initlaize buffers and state if not already done
    init();
 
    if (!m_sessionOpen) {
      delay(100);    // give some time to close down
      return false;  // Already closed down
    }
 
    memset(mRecvBuf.data(), 0x00, RTSP_BUFFER_SIZE);
    int res = readSocket(m_RtspClient, mRecvBuf.data(), RTSP_BUFFER_SIZE,
                         readTimeoutMs);
    if (res > 0) {
      // we filter away everything which seems not to be an RTSP command:
      // O-ption, D-escribe, S-etup, P-lay, T-eardown
      if ((mRecvBuf[0] == 'O') || (mRecvBuf[0] == 'D') ||
          (mRecvBuf[0] == 'S') || (mRecvBuf[0] == 'P') ||
          (mRecvBuf[0] == 'T')) {
        RTSP_CMD_TYPES C = handleRtspRequest(mRecvBuf.data(), res);
        // TODO this should go in the handling functions
        if (C == RTSP_PLAY) {
          m_streaming = true;
        } else if (C == RTSP_PAUSE) {
          m_streaming = false;
        } else if (C == RTSP_TEARDOWN) {
          m_sessionOpen = false;  // Session ended by TEARDOWN
 
          // Properly cleanup streaming on TEARDOWN command
          if (m_streaming && m_Streamer) {
            LOGI("Stopping streamer due to TEARDOWN");
            m_Streamer->stop();
            m_Streamer->releaseUdpTransport();
            m_streaming = false;
          }
        }
      }
      return true;
    } else if (res == 0) {
      LOGW("client closed socket, exiting");
      m_sessionOpen = false;  // Session ended by client disconnect
 
      // CRITICAL: Properly cleanup streaming when client disconnects
      if (m_streaming && m_Streamer) {
        LOGI("Stopping streamer due to client disconnect");
        m_Streamer->stop();
        m_Streamer->releaseUdpTransport();
        m_streaming = false;
      }
 
      return false;  // Return false to indicate session should end
    } else {
      // Timeout on read
      // printf("RTSP read timeout\n");
      return false;
    }
  }
 
  bool isSessionOpen() { return m_sessionOpen; }
 
  bool isStreaming() { return m_streaming; }
 
 protected:
  const char* STD_URL_PRE_SUFFIX = "trackID";
 
  // global session state parameters
  int m_RtspSessionID;
  typename Platform::TcpClientType m_Client;
  typename Platform::TcpClientType* m_RtspClient =
      nullptr;                // RTSP socket of session
  int m_StreamID = -1;        // number of simulated stream of that session
  uint16_t m_ClientRTPPort;   // client port for UDP based RTP transport
  uint16_t m_ClientRTCPPort;  // client port for UDP based RTCP transport
  RTSPAudioStreamerBase<Platform>* m_Streamer =
      nullptr;  // the UDP streamer of that session
 
  // parameters of the last received RTSP request
  RTSP_CMD_TYPES m_RtspCmdType;  // command type (if any) of the current request
  audio_tools::Vector<char> m_URLPreSuffix;  // stream name pre suffix
  audio_tools::Vector<char> m_URLSuffix;     // stream name suffix
  audio_tools::Vector<char> m_CSeq;          // RTSP command sequence number
  audio_tools::Vector<char> m_URLHostPort;   // host:port part of the URL
  unsigned m_ContentLength;                  // SDP string size
  uint16_t m_RtpClientPort =
      0;  // RTP receiver port on client (in host byte order!)
  uint16_t m_RtcpClientPort =
      0;  // RTCP receiver port on client (in host byte order!)
  // Transport parsing (TCP interleaved)
  bool m_TransportIsTcp = false;
  int m_InterleavedRtp = -1;
  int m_InterleavedRtcp = -1;
  audio_tools::Vector<char>
      m_Response;  // Note: we assume single threaded, this large buf we
                   // keep off of the tiny stack
  audio_tools::Vector<char> m_SDPBuf;
  audio_tools::Vector<char> m_URLBuf;
  audio_tools::Vector<char> m_Buf1;
  audio_tools::Vector<char> m_Buf2;
  audio_tools::Vector<char> mRecvBuf;
  audio_tools::Vector<char> mCurRequest;
  audio_tools::Vector<char> m_CmdName;
  bool m_is_init = false;
  bool m_streaming = false;
  volatile bool m_sessionOpen = true;
 
  void init() {
    if (m_is_init) return;
    LOGD("init");
 
    // Reset session state for clean initialization
    m_streaming = false;
    m_sessionOpen = true;
 
    // initialize buffers if not already done
    if (mRecvBuf.size() == 0) {
      mRecvBuf.resize(RTSP_BUFFER_SIZE);
    }
    if (mCurRequest.size() == 0) {
      mCurRequest.resize(RTSP_BUFFER_SIZE);
    }
    if (m_URLPreSuffix.size() == 0) {
      m_URLPreSuffix.resize(RTSP_PARAM_STRING_MAX);
    }
    if (m_URLSuffix.size() == 0) {
      m_URLSuffix.resize(RTSP_PARAM_STRING_MAX);
    }
    if (m_CSeq.size() == 0) {
      m_CSeq.resize(RTSP_PARAM_STRING_MAX);
    }
    if (m_URLHostPort.size() == 0) {
      m_URLHostPort.resize(MAX_HOSTNAME_LEN);
    }
    if (m_Response.size() == 0) {
      m_Response.resize(RTSP_RESPONSE_BUFFER_SIZE);
    }
    if (m_SDPBuf.size() == 0) {
      m_SDPBuf.resize(RTSP_SDP_BUFFER_SIZE);
    }
    if (m_URLBuf.size() == 0) {
      m_URLBuf.resize(RTSP_URL_BUFFER_SIZE);
    }
    if (m_Buf1.size() == 0) {
      m_Buf1.resize(RTSP_SMALL_BUFFER_SIZE);
    }
    if (m_Buf2.size() == 0) {
      m_Buf2.resize(RTSP_SMALL_BUFFER_SIZE);
    }
    if (m_CmdName.size() == 0) {
      m_CmdName.resize(RTSP_PARAM_STRING_MAX);
    }
 
    m_RtspCmdType = RTSP_UNKNOWN;
    memset(m_URLPreSuffix.data(), 0x00, m_URLPreSuffix.size());
    memset(m_URLSuffix.data(), 0x00, m_URLSuffix.size());
    memset(m_CSeq.data(), 0x00, m_CSeq.size());
    memset(m_URLHostPort.data(), 0x00, m_URLHostPort.size());
    m_ContentLength = 0;
    m_TransportIsTcp = false;
    m_InterleavedRtp = -1;
    m_InterleavedRtcp = -1;
    m_is_init = true;
  }
 
  RTSP_CMD_TYPES handleRtspRequest(char const* aRequest,
                                   unsigned aRequestSize) {
    if (parseRtspRequest(aRequest, aRequestSize)) {
      switch (m_RtspCmdType) {
        case RTSP_OPTIONS: {
          handleRtspOption();
          break;
        }
        case RTSP_DESCRIBE: {
          handleRtspDescribe();
          break;
        }
        case RTSP_SETUP: {
          handleRtspSetup();
          break;
        }
        case RTSP_PLAY: {
          handleRtspPlay();
          break;
        }
        case RTSP_PAUSE: {
          handleRtspPause();
          break;
        }
        case RTSP_TEARDOWN: {
          handleRtspTeardown();
          break;
        }
        default: {
        }
      }
    }
    return m_RtspCmdType;
  }
 
  bool parseRtspRequest(char const* aRequest, unsigned aRequestSize) {
    LOGI("aRequest: ------------------------\n%s\n-------------------------",
         aRequest);
 
    const unsigned CurRequestSize = aRequestSize;
    memcpy(mCurRequest.data(), aRequest, aRequestSize);
 
    // 1) Ports and transport
    parseClientPorts(mCurRequest.data());
    parseTransportHeader(mCurRequest.data());
 
    // 2) Command + URL host/parts
    unsigned idxAfterCmd = 0;
    if (!parseCommandName(mCurRequest.data(), CurRequestSize, idxAfterCmd))
      return false;
    determineCommandType();
    parseUrlHostPortAndSuffix(mCurRequest.data(), CurRequestSize, idxAfterCmd);
 
    // 3) CSeq and Content-Length
    if (!parseCSeq(mCurRequest.data(), CurRequestSize, idxAfterCmd))
      return false;
    parseContentLength(mCurRequest.data(), CurRequestSize, idxAfterCmd);
 
    // 4) Client preference toggle (User-Agent / URL)
    detectClientHeaderPreference(mCurRequest.data());
 
    return true;
  }
 
  // ---- Parsing helpers ----
  void parseClientPorts(char* req) {
    char* ClientPortPtr = strstr(req, "client_port");
    if (!ClientPortPtr) return;
    char* lineEnd = strstr(ClientPortPtr, "\r\n");
    if (!lineEnd) return;
    char* CP = m_Response.data();
    memset(CP, 0, m_Response.size());
    char saved = lineEnd[0];
    lineEnd[0] = '\0';
    strcpy(CP, ClientPortPtr);
    char* eq = strstr(CP, "=");
    if (eq) {
      ++eq;
      strcpy(CP, eq);
      char* dash = strstr(CP, "-");
      if (dash) {
        dash[0] = '\0';
        m_ClientRTPPort = atoi(CP);
        m_ClientRTCPPort = m_ClientRTPPort + 1;
      }
    }
    lineEnd[0] = saved;
  }
 
  void parseTransportHeader(char* req) {
    char* TransportPtr = strstr(req, "Transport:");
    if (!TransportPtr) return;
    char* lineEnd = strstr(TransportPtr, "\r\n");
    if (!lineEnd) return;
    char* CP = m_Response.data();
    memset(CP, 0, m_Response.size());
    char saved = lineEnd[0];
    lineEnd[0] = '\0';
    strncpy(CP, TransportPtr, m_Response.size() - 1);
    CP[m_Response.size() - 1] = '\0';
    if (strstr(CP, "RTP/AVP/TCP") || strstr(CP, "/TCP")) m_TransportIsTcp = true;
    char* inter = strstr(CP, "interleaved=");
    if (inter) {
      inter += strlen("interleaved=");
      int a = -1, b = -1;
      if (sscanf(inter, "%d-%d", &a, &b) == 2) {
        m_InterleavedRtp = a;
        m_InterleavedRtcp = b;
      } else if (sscanf(inter, "%d,%d", &a, &b) == 2) {
        m_InterleavedRtp = a;
        m_InterleavedRtcp = b;
      } else if (sscanf(inter, "%d", &a) == 1) {
        m_InterleavedRtp = a;
        m_InterleavedRtcp = a + 1;
      }
    }
    lineEnd[0] = saved;
  }
 
  bool parseCommandName(char* req, unsigned reqSize, unsigned& outIdx) {
    bool ok = false;
    unsigned i;
    for (i = 0; i < m_CmdName.size() - 1 && i < reqSize; ++i) {
      char c = req[i];
      if (c == ' ' || c == '\t') {
        ok = true;
        break;
      }
      m_CmdName[i] = c;
    }
    m_CmdName[i] = '\0';
    if (!ok) {
      LOGE("failed to parse RTSP");
      return false;
    }
    LOGI("RTSP received %s", m_CmdName.data());
    outIdx = i;
    return true;
  }
 
  void determineCommandType() {
    if (strstr(m_CmdName.data(), "OPTIONS"))
      m_RtspCmdType = RTSP_OPTIONS;
    else if (strstr(m_CmdName.data(), "DESCRIBE"))
      m_RtspCmdType = RTSP_DESCRIBE;
    else if (strstr(m_CmdName.data(), "SETUP"))
      m_RtspCmdType = RTSP_SETUP;
    else if (strstr(m_CmdName.data(), "PLAY"))
      m_RtspCmdType = RTSP_PLAY;
    else if (strstr(m_CmdName.data(), "PAUSE"))
      m_RtspCmdType = RTSP_PAUSE;
    else if (strstr(m_CmdName.data(), "TEARDOWN"))
      m_RtspCmdType = RTSP_TEARDOWN;
    else
      LOGE("Error: Unsupported Command received (%s)!", m_CmdName.data());
  }
 
  void parseUrlHostPortAndSuffix(char* req, unsigned reqSize, unsigned& i) {
    unsigned j = i + 1;
    while (j < reqSize && (req[j] == ' ' || req[j] == '\t')) ++j;
    for (; (int)j < (int)(reqSize - 8); ++j) {
      if ((req[j] == 'r' || req[j] == 'R') && (req[j + 1] == 't' || req[j + 1] == 'T') &&
          (req[j + 2] == 's' || req[j + 2] == 'S') && (req[j + 3] == 'p' || req[j + 3] == 'P') &&
          req[j + 4] == ':' && req[j + 5] == '/') {
        j += 6;
        if (req[j] == '/') {
          ++j;
          unsigned uidx = 0;
          while (j < reqSize && req[j] != '/' && req[j] != ' ' && uidx < m_URLHostPort.size() - 1) {
            m_URLHostPort[uidx++] = req[j++];
          }
        } else {
          --j;
        }
        i = j;
        break;
      }
    }
    LOGD("m_URLHostPort: %s", m_URLHostPort.data());
 
    bool ok = false;
    for (unsigned k = i + 1; (int)k < (int)(reqSize - 5); ++k) {
      if (req[k] == 'R' && req[k + 1] == 'T' && req[k + 2] == 'S' && req[k + 3] == 'P' && req[k + 4] == '/') {
        while (--k >= i && req[k] == ' ') {}
        unsigned k1 = k;
        while (k1 > i && req[k1] != '=') --k1;
        if (k - k1 + 1 <= m_URLSuffix.size()) {
          unsigned n = 0, k2 = k1 + 1;
          while (k2 <= k) m_URLSuffix[n++] = req[k2++];
          m_URLSuffix[n] = '\0';
          if (k1 - i <= m_URLPreSuffix.size()) ok = true;
          n = 0;
          k2 = i + 1;
          while (k2 <= k1 - 1) m_URLPreSuffix[n++] = req[k2++];
          m_URLPreSuffix[n] = '\0';
          i = k + 7;
        }
        break;
      }
    }
    LOGD("m_URLSuffix: %s", m_URLSuffix.data());
    LOGD("m_URLPreSuffix: %s", m_URLPreSuffix.data());
    LOGD("URL Suffix parse succeeded: %i", ok);
  }
 
  bool parseCSeq(char* req, unsigned reqSize, unsigned startIdx) {
    bool ok = false;
    for (unsigned j = startIdx; (int)j < (int)(reqSize - 5); ++j) {
      if (req[j] == 'C' && req[j + 1] == 'S' && req[j + 2] == 'e' && req[j + 3] == 'q' && req[j + 4] == ':') {
        j += 5;
        while (j < reqSize && (req[j] == ' ' || req[j] == '\t')) ++j;
        unsigned n;
        for (n = 0; n < m_CSeq.size() - 1 && j < reqSize; ++n, ++j) {
          char c = req[j];
          if (c == '\r' || c == '\n') {
            ok = true;
            break;
          }
          m_CSeq[n] = c;
        }
        m_CSeq[n] = '\0';
        break;
      }
    }
    LOGD("Look for CSeq success: %i", ok);
    return ok;
  }
 
  void parseContentLength(char* req, unsigned reqSize, unsigned startIdx) {
    for (unsigned j = startIdx; (int)j < (int)(reqSize - 15); ++j) {
      if (req[j] == 'C' && req[j + 1] == 'o' && req[j + 2] == 'n' && req[j + 3] == 't' &&
          req[j + 4] == 'e' && req[j + 5] == 'n' && req[j + 6] == 't' && req[j + 7] == '-' &&
          (req[j + 8] == 'L' || req[j + 8] == 'l') && req[j + 9] == 'e' && req[j + 10] == 'n' &&
          req[j + 11] == 'g' && req[j + 12] == 't' && req[j + 13] == 'h' && req[j + 14] == ':') {
        j += 15;
        while (j < reqSize && (req[j] == ' ' || req[j] == '\t')) ++j;
        unsigned num;
        if (sscanf(&req[j], "%u", &num) == 1) m_ContentLength = num;
      }
    }
  }
 
  void detectClientHeaderPreference(char* req) {
    char* ua = strstr(req, "User-Agent:");
    bool want_rfc2250 = false;
    if (ua) {
      if (strcasestr(ua, "ffmpeg") || strcasestr(ua, "ffplay") || strcasestr(ua, "libavformat") ||
          strcasestr(ua, "Lavf")) {
        want_rfc2250 = true;
      }
      if (strcasestr(ua, "vlc")) want_rfc2250 = false;
    }
    char* qm = strchr(req, '?');
    if (qm) {
      if (strstr(qm, "mpa_hdr=1")) want_rfc2250 = true;
      if (strstr(qm, "mpa_hdr=0")) want_rfc2250 = false;
    }
    if (m_Streamer && m_Streamer->getAudioSource()) {
      RTSPFormat& fmt = m_Streamer->getAudioSource()->getFormat();
      fmt.setUseRfc2250Header(want_rfc2250);
    }
  }
 
  void handleRtspOption() {
    snprintf(m_Response.data(), m_Response.size(),
             "RTSP/1.0 200 OK\r\nCSeq: %s\r\n"
             "Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE\r\n\r\n",
             m_CSeq.data());
 
    sendSocket(m_RtspClient, m_Response.data(), strlen(m_Response.data()));
  }
 
  void handleRtspDescribe() {
    // check whether we know a stream with the URL which is requested
    m_StreamID = -1;  // invalid URL
    // find Stream ID
    if (strcmp(m_URLPreSuffix.data(), STD_URL_PRE_SUFFIX) == 0) {
      char* end;
      m_StreamID = strtol(m_URLSuffix.data(), &end, 10);
      if (*end != '\0') m_StreamID = -1;
    }
 
    // simulate DESCRIBE server response
    char* ColonPtr;
    strncpy(m_Buf1.data(), m_URLHostPort.data(), 256);
    ColonPtr = strstr(m_Buf1.data(), ":");
    if (ColonPtr != nullptr) ColonPtr[0] = 0x00;
 
    snprintf(
        m_SDPBuf.data(), m_SDPBuf.size(),
        "v=0\r\n"  // SDP Version
        "o=- %d 0 IN IP4 %s\r\n"
        "%s"
        "a=control:%s=0",
        rand() & 0xFF, m_Buf1.data(),
        m_Streamer->getAudioSource()->getFormat().format(m_Buf2.data(), 256),
        STD_URL_PRE_SUFFIX);
 
    snprintf(m_URLBuf.data(), m_URLBuf.size(), "rtsp://%s",
             m_URLHostPort.data());
 
    snprintf(m_Response.data(), m_Response.size(),
             "RTSP/1.0 200 OK\r\nCSeq: %s\r\n"
             "%s\r\n"
             "Content-Base: %s/\r\n"
             "Content-Type: application/sdp\r\n"
             "Content-Length: %d\r\n\r\n"
             "%s",
             m_CSeq.data(), dateHeader(), m_URLBuf.data(),
             (int)strlen(m_SDPBuf.data()), m_SDPBuf.data());
 
    // LOGI("handleRtspDescribe: %s", (const char*)m_Response.data());
    Serial.println("------------------------------");
    Serial.println((const char*)m_Response.data());
    Serial.println("------------------------------");
    sendSocket(m_RtspClient, m_Response.data(), strlen(m_Response.data()));
  }
 
  void handleRtspSetup() {
    // Build response depending on requested transport
    if (m_TransportIsTcp) {
      // Initialize TCP interleaved transport on same RTSP TCP socket
      int ch0 = (m_InterleavedRtp >= 0) ? m_InterleavedRtp : 0;
      int ch1 = (m_InterleavedRtcp >= 0) ? m_InterleavedRtcp : (ch0 + 1);
      m_Streamer->initTcpInterleavedTransport(m_RtspClient, ch0, ch1);
 
      // Reply with interleaved channels
      snprintf(m_Buf1.data(), m_Buf1.size(),
               "RTP/AVP/TCP;unicast;interleaved=%d-%d", ch0, ch1);
    } else {
      // init RTSP Session transport type (UDP) and ports for UDP transport
      initTransport(m_ClientRTPPort, m_ClientRTCPPort);
      // UDP Transport response with client/server ports and ssrc
      snprintf(m_Buf1.data(), m_Buf1.size(),
               "RTP/AVP;unicast;client_port=%i-%i;server_port=%i-%i;ssrc=%08X",
               m_ClientRTPPort, m_ClientRTCPPort,
               m_Streamer->getRtpServerPort(), m_Streamer->getRtcpServerPort(),
               (unsigned)m_Streamer->currentSsrc());
    }
    snprintf(m_Response.data(), m_Response.size(),
             "RTSP/1.0 200 OK\r\n"
             "CSeq: %s\r\n"
             "%s\r\n"
             "Session: %i\r\n"
             "Transport: %s\r\n"
             "\r\n",
             m_CSeq.data(), dateHeader(), m_RtspSessionID, m_Buf1.data());
 
    Serial.println("------------------------------");
    Serial.println((char*)m_Response.data());
    Serial.println("------------------------------");
 
    sendSocket(m_RtspClient, m_Response.data(), strlen(m_Response.data()));
  }
 
  void handleRtspPlay() {
    // Build RTP-Info to help clients (e.g., VLC) synchronize
    // URL base
    snprintf(m_URLBuf.data(), m_URLBuf.size(), "rtsp://%s/%s=0",
             m_URLHostPort.data(), STD_URL_PRE_SUFFIX);
 
    // Current seq/timestamp from streamer (values used for next packet)
    uint16_t seq = m_Streamer ? m_Streamer->currentSeq() : 0;
    uint32_t rtptime = m_Streamer ? m_Streamer->currentRtpTimestamp() : 0;
 
    // PLAY response with Range, Session and RTP-Info
    snprintf(m_Response.data(), m_Response.size(),
             "RTSP/1.0 200 OK\r\n"
             "CSeq: %s\r\n"
             "Range: npt=0.000-\r\n"
             "Session: %i\r\n"
             "RTP-Info: url=%s;seq=%u;rtptime=%u\r\n\r\n",
             m_CSeq.data(), m_RtspSessionID, m_URLBuf.data(), (unsigned)seq,
             (unsigned)rtptime);
 
    Serial.println("------------------------------");
    Serial.println((char*)m_Response.data());
    Serial.println("------------------------------");
 
    sendSocket(m_RtspClient, m_Response.data(), strlen(m_Response.data()));
 
    m_Streamer->start();
  }
 
  void handleRtspPause() {
    if (m_streaming && m_Streamer) {
      m_Streamer->stop();
    }
    snprintf(m_Response.data(), m_Response.size(),
             "RTSP/1.0 200 OK\r\n"
             "CSeq: %s\r\n"
             "Session: %i\r\n\r\n",
             m_CSeq.data(), m_RtspSessionID);
    sendSocket(m_RtspClient, m_Response.data(), strlen(m_Response.data()));
  }
 
  void handleRtspTeardown() {
    m_Streamer->stop();
 
    // simulate SETUP server response
    snprintf(m_Response.data(), m_Response.size(),
             "RTSP/1.0 200 OK\r\n"
             "CSeq: %s\r\n\r\n",
             m_CSeq.data());
 
    sendSocket(m_RtspClient, m_Response.data(), strlen(m_Response.data()));
 
    m_sessionOpen = false;
  }
 
  int getStreamID() { return m_StreamID; }
 
  void initTransport(u_short aRtpPort, u_short aRtcpPort) {
    m_RtpClientPort = aRtpPort;
    m_RtcpClientPort = aRtcpPort;
 
    IPAddress clientIP;
    uint16_t clientPort;
    Platform::getSocketPeerAddr(m_RtspClient, &clientIP, &clientPort);
 
    m_Streamer->initUdpTransport(clientIP, m_RtpClientPort);
  }
 
  typename Platform::TcpClientType*& getClient() { return m_RtspClient; }
 
  uint16_t getRtpClientPort() { return m_RtpClientPort; }
 
  inline int readSocket(typename Platform::TcpClientType* sock, char* buf,
                        size_t buflen, int timeoutmsec) {
    return Platform::readSocket(sock, buf, buflen, timeoutmsec);
  }
 
  inline ssize_t sendSocket(typename Platform::TcpClientType* sock,
                            const void* buf, size_t len) {
    return Platform::sendSocket(sock, buf, len);
  }
 
  char const* dateHeader() {
    static char buf[200];
    time_t tt = time(NULL);
    strftime(buf, sizeof(buf), "Date: %a, %b %d %Y %H:%M:%S GMT", gmtime(&tt));
    return buf;
  }
};
 
}  // namespace audio_tools