Dynamically determine audio, video, and control ports from RTSP DESCRIBE response

src/AudioStream.c

@@ -14,6 +14,7 @@ static uint32_t avRiKeyId;
 
 static unsigned short lastSeq;
 
+static bool pingThreadStarted;
 static bool receivedDataFromPeer;
 static uint64_t firstReceiveTime;
 
@@ -22,8 +23,6 @@ static uint64_t firstReceiveTime;
 static uint8_t opusHeaderByte;
 #endif
 
-#define RTP_PORT 48000
-
 #define MAX_PACKET_SIZE 1400
 
 // This is much larger than we should typically have buffered, but
@@ -47,7 +46,7 @@ static void AudioPingThreadProc(void* context) {
     LC_SOCKADDR saddr;
 
     memcpy(&saddr, &RemoteAddr, sizeof(saddr));
-    SET_PORT(&saddr, RTP_PORT);
+    SET_PORT(&saddr, AudioPortNumber);
 
     // Send PING every 500 milliseconds
     while (!PltIsThreadInterrupted(&udpPingThread)) {
@@ -74,6 +73,7 @@ int initializeAudioStream(void) {
     RtpaInitializeQueue(&rtpAudioQueue);
     lastSeq = 0;
     receivedDataFromPeer = false;
+    pingThreadStarted = false;
     firstReceiveTime = 0;
     audioDecryptionCtx = PltCreateCryptoContext();
 #ifdef LC_DEBUG
@@ -91,15 +91,23 @@ int initializeAudioStream(void) {
         return LastSocketFail();
     }
 
+    return 0;
+}
+
+// This is called when the RTSP DESCRIBE message is parsed and the audio port
+// number is parsed out of it. Alternatively, it's also called if parsing fails
+// and will use the well known audio port instead.
+int notifyAudioPortNegotiationComplete(void) {
+    LC_ASSERT(!pingThreadStarted);
+
     // We may receive audio before our threads are started, but that's okay. We'll
     // drop the first 1 second of audio packets to catch up with the backlog.
     int err = PltCreateThread("AudioPing", AudioPingThreadProc, NULL, &udpPingThread);
     if (err != 0) {
-        closeSocket(rtpSocket);
-        rtpSocket = INVALID_SOCKET;
         return err;
     }
 
+    pingThreadStarted = true;
     return 0;
 }
 
@@ -119,9 +127,11 @@ static void freePacketList(PLINKED_BLOCKING_QUEUE_ENTRY entry) {
 // Tear down the audio stream once we're done with it
 void destroyAudioStream(void) {
     if (rtpSocket != INVALID_SOCKET) {
+        if (pingThreadStarted) {
             PltInterruptThread(&udpPingThread);
             PltJoinThread(&udpPingThread);
             PltCloseThread(&udpPingThread);
+        }
 
         closeSocket(rtpSocket);
         rtpSocket = INVALID_SOCKET;

src/Connection.c

@@ -24,6 +24,10 @@ OPUS_MULTISTREAM_CONFIGURATION HighQualityOpusConfig;
 int OriginalVideoBitrate;
 int AudioPacketDuration;
 bool AudioEncryptionEnabled;
+uint16_t RtspPortNumber;
+uint16_t ControlPortNumber;
+uint16_t AudioPortNumber;
+uint16_t VideoPortNumber;
 
 // Connection stages
 static const char* stageNames[STAGE_MAX] = {
@@ -208,6 +212,12 @@ int LiStartConnection(PSERVER_INFORMATION serverInfo, PSTREAM_CONFIGURATION stre
     OriginalVideoBitrate = streamConfig->bitrate;
     RemoteAddrString = strdup(serverInfo->address);
 
+    // Configure default ports
+    VideoPortNumber = 47998;
+    ControlPortNumber = 47999;
+    AudioPortNumber = 48000;
+    RtspPortNumber = 48010; // TODO: Parse this out of RTSP session URL
+
     alreadyTerminated = false;
     ConnectionInterrupted = false;
     

src/ControlStream.c

@@ -1143,7 +1143,7 @@ int startControlStream(void) {
         ENetEvent event;
         
         enet_address_set_address(&address, (struct sockaddr *)&RemoteAddr, RemoteAddrLen);
-        enet_address_set_port(&address, 47999);
+        enet_address_set_port(&address, ControlPortNumber);
 
         // Create a client that can use 1 outgoing connection and 1 channel
         client = enet_host_create(address.address.ss_family, NULL, 1, 1, 0, 0);
@@ -1166,7 +1166,7 @@ int startControlStream(void) {
         // Wait for the connect to complete
         if (serviceEnetHost(client, &event, CONTROL_STREAM_TIMEOUT_SEC * 1000) <= 0 ||
             event.type != ENET_EVENT_TYPE_CONNECT) {
-            Limelog("Failed to connect to UDP port 47999\n");
+            Limelog("Failed to connect to UDP port %u\n", ControlPortNumber);
             stopping = true;
             enet_peer_reset(peer);
             peer = NULL;
@@ -1182,6 +1182,7 @@ int startControlStream(void) {
         enet_peer_timeout(peer, 2, 10000, 10000);
     }
     else {
+        // NB: Do NOT use ControlPortNumber here. 47995 is correct for these old versions.
         ctlSock = connectTcpSocket(&RemoteAddr, RemoteAddrLen,
             47995, CONTROL_STREAM_TIMEOUT_SEC);
         if (ctlSock == INVALID_SOCKET) {

src/Limelight-internal.h

@@ -32,6 +32,11 @@ extern int OriginalVideoBitrate;
 extern int AudioPacketDuration;
 extern bool AudioEncryptionEnabled;
 
+extern uint16_t RtspPortNumber;
+extern uint16_t ControlPortNumber;
+extern uint16_t AudioPortNumber;
+extern uint16_t VideoPortNumber;
+
 #ifndef UINT24_MAX
 #define UINT24_MAX 0xFFFFFF
 #endif
@@ -100,6 +105,7 @@ int startVideoStream(void* rendererContext, int drFlags);
 void stopVideoStream(void);
 
 int initializeAudioStream(void);
+int notifyAudioPortNegotiationComplete(void);
 void destroyAudioStream(void);
 int startAudioStream(void* audioContext, int arFlags);
 void stopAudioStream(void);

src/RtspConnection.c

@@ -228,7 +228,7 @@ static bool transactRtspMessageTcp(PRTSP_MESSAGE request, PRTSP_MESSAGE response
     // returns HTTP 200 OK for the /launch request before the RTSP handshake port
     // is listening.
     do {
-        sock = connectTcpSocket(&RemoteAddr, RemoteAddrLen, 48010, RTSP_TIMEOUT_SEC);
+        sock = connectTcpSocket(&RemoteAddr, RemoteAddrLen, RtspPortNumber, RTSP_TIMEOUT_SEC);
         if (sock == INVALID_SOCKET) {
             *error = LastSocketError();
             if (*error == ECONNREFUSED) {
@@ -525,6 +525,57 @@ static int parseOpusConfigFromParamString(char* paramStr, int channelCount, POPU
     return 0;
 }
 
+static bool parseMediaEntry(PRTSP_MESSAGE response, const char* mediaType, const char* transport, uint16_t* port) {
+    char paramsPrefix[128];
+    char paramsSuffix[128];
+    char* paramStart;
+
+    sprintf(paramsPrefix, "m=%s ", mediaType);
+    sprintf(paramsSuffix, " %s", transport);
+
+    // Look for the next match
+    paramStart = response->payload;
+    while ((paramStart = strstr(paramStart, paramsPrefix)) != NULL) {
+        // Skip the prefix
+        paramStart += strlen(paramsPrefix);
+
+        // The first part should be the port number
+        char* nextToken;
+        long int rawPort = strtol(paramStart, &nextToken, 10);
+        if (rawPort <= 0 || rawPort >= 65535) {
+            continue;
+        }
+
+        // Skip this entry if the transport isn't what we expect
+        if (strncmp(nextToken, paramsSuffix, strlen(paramsSuffix)) != 0) {
+            continue;
+        }
+
+        // This entry is a match
+        *port = (uint16_t)rawPort;
+        return true;
+    }
+
+    // No match for this media type and transport
+    return false;
+}
+
+static void parsePortConfigurations(PRTSP_MESSAGE response) {
+    // Don't parse ports on very old GFE versions
+    if (!APP_VERSION_AT_LEAST(7, 0, 0)) {
+        return;
+    }
+
+    parseMediaEntry(response, "video", "RTP/AVP", &VideoPortNumber);
+    parseMediaEntry(response, "audio", "RTP/AVP", &AudioPortNumber);
+
+    if (!parseMediaEntry(response, "application", "udp", &ControlPortNumber)) {
+        if (!parseMediaEntry(response, "application", "udp_enc", &ControlPortNumber)) {
+            parseMediaEntry(response, "application", "udp_ag", &ControlPortNumber);
+        }
+    }
+}
+
 // Parses the Opus configuration from an RTSP DESCRIBE response
 static int parseOpusConfigurations(PRTSP_MESSAGE response) {
     HighQualitySurroundSupported = false;
@@ -644,7 +695,7 @@ int performRtspHandshake(void) {
 
     // Initialize global state
     useEnet = (AppVersionQuad[0] >= 5) && (AppVersionQuad[0] <= 7) && (AppVersionQuad[2] < 404);
-    sprintf(rtspTargetUrl, "rtsp%s://%s:48010", useEnet ? "ru" : "", urlAddr);
+    sprintf(rtspTargetUrl, "rtsp%s://%s:%u", useEnet ? "ru" : "", urlAddr, RtspPortNumber);
     currentSeqNumber = 1;
     hasSessionId = false;
     controlStreamId = APP_VERSION_AT_LEAST(7, 1, 431) ? "streamid=control/13/0" : "streamid=control/1/0";
@@ -676,7 +727,7 @@ int performRtspHandshake(void) {
         ENetEvent event;
         
         enet_address_set_address(&address, (struct sockaddr *)&RemoteAddr, RemoteAddrLen);
-        enet_address_set_port(&address, 48010);
+        enet_address_set_port(&address, RtspPortNumber);
         
         // Create a client that can use 1 outgoing connection and 1 channel
         client = enet_host_create(RemoteAddr.ss_family, NULL, 1, 1, 0, 0);
@@ -695,7 +746,7 @@ int performRtspHandshake(void) {
         // Wait for the connect to complete
         if (serviceEnetHost(client, &event, RTSP_TIMEOUT_SEC * 1000) <= 0 ||
             event.type != ENET_EVENT_TYPE_CONNECT) {
-            Limelog("RTSP: Failed to connect to UDP port 48010\n");
+            Limelog("RTSP: Failed to connect to UDP port %u\n", RtspPortNumber);
             enet_peer_reset(peer);
             peer = NULL;
             enet_host_destroy(client);
@@ -773,6 +824,14 @@ int performRtspHandshake(void) {
             }
         }
 
+        // Parse audio, video, and control ports out of the RTSP DESCRIBE response.
+        parsePortConfigurations(&response);
+
+        // Let the audio stream know the port number is now finalized.
+        // NB: This is needed because audio stream init happens before RTSP,
+        // which is not the case for the video stream.
+        notifyAudioPortNegotiationComplete();
+
         // Parse the Opus surround parameters out of the RTSP DESCRIBE response.
         ret = parseOpusConfigurations(&response);
         if (ret != 0) {

src/VideoStream.c

@@ -3,7 +3,6 @@
 #define FIRST_FRAME_MAX 1500
 #define FIRST_FRAME_TIMEOUT_SEC 10
 
-#define RTP_PORT 47998
 #define FIRST_FRAME_PORT 47996
 
 #define RTP_RECV_BUFFER (512 * 1024)
@@ -48,7 +47,7 @@ static void VideoPingThreadProc(void* context) {
     LC_SOCKADDR saddr;
 
     memcpy(&saddr, &RemoteAddr, sizeof(saddr));
-    SET_PORT(&saddr, RTP_PORT);
+    SET_PORT(&saddr, VideoPortNumber);
 
     while (!PltIsThreadInterrupted(&udpPingThread)) {
         // We do not check for errors here. Socket errors will be handled