/* GStreamer RTMP Library * Copyright (C) 2013 David Schleef * Copyright (C) 2017 Make.TV, Inc. * Contact: Jan Alexander Steffens (heftig) * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin Street, Suite 500, * Boston, MA 02110-1335, USA. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include "rtmpclient.h" #include "rtmphandshake.h" #include "rtmpmessage.h" #include "rtmputils.h" GST_DEBUG_CATEGORY_STATIC (gst_rtmp_client_debug_category); #define GST_CAT_DEFAULT gst_rtmp_client_debug_category static void send_connect_done (const gchar * command_name, GPtrArray * args, gpointer user_data); static void create_stream_done (const gchar * command_name, GPtrArray * args, gpointer user_data); static void on_publish_or_play_status (const gchar * command_name, GPtrArray * args, gpointer user_data); static void init_debug (void) { static gsize done = 0; if (g_once_init_enter (&done)) { GST_DEBUG_CATEGORY_INIT (gst_rtmp_client_debug_category, "rtmpclient", 0, "debug category for the rtmp client"); GST_DEBUG_REGISTER_FUNCPTR (send_connect_done); GST_DEBUG_REGISTER_FUNCPTR (create_stream_done); GST_DEBUG_REGISTER_FUNCPTR (on_publish_or_play_status); g_once_init_leave (&done, 1); } } static const gchar *scheme_strings[] = { "rtmp", "rtmps", NULL }; #define NUM_SCHEMES (G_N_ELEMENTS (scheme_strings) - 1) GType gst_rtmp_scheme_get_type (void) { static gsize scheme_type = 0; static const GEnumValue scheme[] = { {GST_RTMP_SCHEME_RTMP, "GST_RTMP_SCHEME_RTMP", "rtmp"}, {GST_RTMP_SCHEME_RTMPS, "GST_RTMP_SCHEME_RTMPS", "rtmps"}, {0, NULL, NULL}, }; if (g_once_init_enter (&scheme_type)) { GType tmp = g_enum_register_static ("GstRtmpScheme", scheme); g_once_init_leave (&scheme_type, tmp); } return (GType) scheme_type; } GstRtmpScheme gst_rtmp_scheme_from_string (const gchar * string) { if (string) { gint value; for (value = 0; value < NUM_SCHEMES; value++) { if (strcmp (scheme_strings[value], string) == 0) { return value; } } } return -1; } GstRtmpScheme gst_rtmp_scheme_from_uri (const GstUri * uri) { const gchar *scheme = gst_uri_get_scheme (uri); if (!scheme) { return GST_RTMP_SCHEME_RTMP; } return gst_rtmp_scheme_from_string (scheme); } const gchar * gst_rtmp_scheme_to_string (GstRtmpScheme scheme) { if (scheme >= 0 && scheme < NUM_SCHEMES) { return scheme_strings[scheme]; } return "invalid"; } const gchar *const * gst_rtmp_scheme_get_strings (void) { return scheme_strings; } guint gst_rtmp_scheme_get_default_port (GstRtmpScheme scheme) { switch (scheme) { case GST_RTMP_SCHEME_RTMP: return 1935; case GST_RTMP_SCHEME_RTMPS: return 443; default: g_return_val_if_reached (0); } } GType gst_rtmp_authmod_get_type (void) { static gsize authmod_type = 0; static const GEnumValue authmod[] = { {GST_RTMP_AUTHMOD_NONE, "GST_RTMP_AUTHMOD_NONE", "none"}, {GST_RTMP_AUTHMOD_AUTO, "GST_RTMP_AUTHMOD_AUTO", "auto"}, {GST_RTMP_AUTHMOD_ADOBE, "GST_RTMP_AUTHMOD_ADOBE", "adobe"}, {0, NULL, NULL}, }; if (g_once_init_enter (&authmod_type)) { GType tmp = g_enum_register_static ("GstRtmpAuthmod", authmod); g_once_init_leave (&authmod_type, tmp); } return (GType) authmod_type; } static const gchar * gst_rtmp_authmod_get_nick (GstRtmpAuthmod value) { GEnumClass *klass = g_type_class_peek (GST_TYPE_RTMP_AUTHMOD); GEnumValue *ev = klass ? g_enum_get_value (klass, value) : NULL; return ev ? ev->value_nick : "(unknown)"; } GType gst_rtmp_stop_commands_get_type (void) { static gsize stop_commands_type = 0; static const GFlagsValue stop_commands[] = { {GST_RTMP_STOP_COMMANDS_NONE, "No command", "none"}, {GST_RTMP_STOP_COMMANDS_FCUNPUBLISH, "FCUnpublish", "fcunpublish"}, {GST_RTMP_STOP_COMMANDS_CLOSE_STREAM, "closeStream", "closestream"}, {GST_RTMP_STOP_COMMANDS_DELETE_STREAM, "deleteStream", "deletestream"}, {0, NULL, NULL}, }; if (g_once_init_enter (&stop_commands_type)) { GType tmp = g_flags_register_static ("GstRtmpStopCommands", stop_commands); g_once_init_leave (&stop_commands_type, tmp); } return (GType) stop_commands_type; } void gst_rtmp_location_copy (GstRtmpLocation * dest, const GstRtmpLocation * src) { g_return_if_fail (dest); g_return_if_fail (src); dest->scheme = src->scheme; dest->host = g_strdup (src->host); dest->port = src->port; dest->application = g_strdup (src->application); dest->stream = g_strdup (src->stream); dest->username = g_strdup (src->username); dest->password = g_strdup (src->password); dest->secure_token = g_strdup (src->secure_token); dest->authmod = src->authmod; dest->timeout = src->timeout; dest->tls_flags = src->tls_flags; dest->flash_ver = g_strdup (src->flash_ver); dest->publish = src->publish; } void gst_rtmp_location_clear (GstRtmpLocation * location) { g_return_if_fail (location); g_clear_pointer (&location->host, g_free); location->port = 0; g_clear_pointer (&location->application, g_free); g_clear_pointer (&location->stream, g_free); g_clear_pointer (&location->username, g_free); g_clear_pointer (&location->password, g_free); g_clear_pointer (&location->secure_token, g_free); g_clear_pointer (&location->flash_ver, g_free); location->publish = FALSE; } gchar * gst_rtmp_location_get_string (const GstRtmpLocation * location, gboolean with_stream) { GstUri *uri; gchar *base, *string; const gchar *scheme_string; guint default_port; g_return_val_if_fail (location, NULL); scheme_string = gst_rtmp_scheme_to_string (location->scheme); default_port = gst_rtmp_scheme_get_default_port (location->scheme); uri = gst_uri_new (scheme_string, NULL, location->host, location->port == default_port ? GST_URI_NO_PORT : location->port, "/", NULL, NULL); base = gst_uri_to_string (uri); string = g_strconcat (base, location->application, with_stream ? "/" : NULL, location->stream, NULL); g_free (base); gst_uri_unref (uri); return string; } /* Flag values for the audioCodecs property, * rtmp_specification_1.0.pdf page 32 */ enum { SUPPORT_SND_NONE = 0x001, /* Raw sound, no compression */ SUPPORT_SND_ADPCM = 0x002, /* ADPCM compression */ SUPPORT_SND_MP3 = 0x004, /* mp3 compression */ SUPPORT_SND_INTEL = 0x008, /* Not used */ SUPPORT_SND_UNUSED = 0x010, /* Not used */ SUPPORT_SND_NELLY8 = 0x020, /* NellyMoser at 8-kHz compression */ SUPPORT_SND_NELLY = 0x040, /* NellyMoser compression * (5, 11, 22, and 44 kHz) */ SUPPORT_SND_G711A = 0x080, /* G711A sound compression * (Flash Media Server only) */ SUPPORT_SND_G711U = 0x100, /* G711U sound compression * (Flash Media Server only) */ SUPPORT_SND_NELLY16 = 0x200, /* NellyMoser at 16-kHz compression */ SUPPORT_SND_AAC = 0x400, /* Advanced audio coding (AAC) codec */ SUPPORT_SND_SPEEX = 0x800, /* Speex Audio */ SUPPORT_SND_ALL = 0xFFF, /* All RTMP-supported audio codecs */ }; /* audioCodecs value sent by libavformat. All "used" codecs. */ #define GST_RTMP_AUDIOCODECS \ (SUPPORT_SND_ALL & ~SUPPORT_SND_INTEL & ~SUPPORT_SND_UNUSED) G_STATIC_ASSERT (GST_RTMP_AUDIOCODECS == 4071); /* libavformat's magic number */ /* Flag values for the videoCodecs property, * rtmp_specification_1.0.pdf page 32 */ enum { SUPPORT_VID_UNUSED = 0x01, /* Obsolete value */ SUPPORT_VID_JPEG = 0x02, /* Obsolete value */ SUPPORT_VID_SORENSON = 0x04, /* Sorenson Flash video */ SUPPORT_VID_HOMEBREW = 0x08, /* V1 screen sharing */ SUPPORT_VID_VP6 = 0x10, /* On2 video (Flash 8+) */ SUPPORT_VID_VP6ALPHA = 0x20, /* On2 video with alpha channel */ SUPPORT_VID_HOMEBREWV = 0x40, /* Screen sharing version 2 (Flash 8+) */ SUPPORT_VID_H264 = 0x80, /* H264 video */ SUPPORT_VID_ALL = 0xFF, /* All RTMP-supported video codecs */ }; /* videoCodecs value sent by libavformat. All non-obsolete codecs. */ #define GST_RTMP_VIDEOCODECS \ (SUPPORT_VID_ALL & ~SUPPORT_VID_UNUSED & ~SUPPORT_VID_JPEG) G_STATIC_ASSERT (GST_RTMP_VIDEOCODECS == 252); /* libavformat's magic number */ /* Flag values for the videoFunction property, * rtmp_specification_1.0.pdf page 32 */ enum { /* Indicates that the client can perform frame-accurate seeks. */ SUPPORT_VID_CLIENT_SEEK = 1, }; /* videoFunction value sent by libavformat */ #define GST_RTMP_VIDEOFUNCTION (SUPPORT_VID_CLIENT_SEEK) G_STATIC_ASSERT (GST_RTMP_VIDEOFUNCTION == 1); /* libavformat's magic number */ static void socket_connect (GTask * task); static void socket_connect_done (GObject * source, GAsyncResult * result, gpointer user_data); static void handshake_done (GObject * source, GAsyncResult * result, gpointer user_data); static void send_connect (GTask * task); static void send_stop (GstRtmpConnection * connection, const gchar * stream, const GstRtmpStopCommands stop_commands); static void send_secure_token_response (GTask * task, GstRtmpConnection * connection, const gchar * challenge); static void connection_error (GstRtmpConnection * connection, gpointer user_data); #define DEFAULT_TIMEOUT 5 typedef struct { GstRtmpLocation location; gchar *auth_query; GstRtmpConnection *connection; gulong error_handler_id; } ConnectTaskData; static ConnectTaskData * connect_task_data_new (const GstRtmpLocation * location) { ConnectTaskData *data = g_slice_new0 (ConnectTaskData); gst_rtmp_location_copy (&data->location, location); return data; } static void connect_task_data_free (gpointer ptr) { ConnectTaskData *data = ptr; gst_rtmp_location_clear (&data->location); g_clear_pointer (&data->auth_query, g_free); if (data->error_handler_id) { g_signal_handler_disconnect (data->connection, data->error_handler_id); } g_clear_object (&data->connection); g_slice_free (ConnectTaskData, data); } static GRegex *auth_regex = NULL; void gst_rtmp_client_connect_async (const GstRtmpLocation * location, GCancellable * cancellable, GAsyncReadyCallback callback, gpointer user_data) { GTask *task; init_debug (); if (g_once_init_enter (&auth_regex)) { GRegex *re = g_regex_new ("\\[ *AccessManager.Reject *\\] *: *" "\\[ *authmod=(?.*?) *\\] *: *" "(?\\?.*)\\Z", G_REGEX_DOTALL, 0, NULL); g_once_init_leave (&auth_regex, re); } task = g_task_new (NULL, cancellable, callback, user_data); g_task_set_task_data (task, connect_task_data_new (location), connect_task_data_free); socket_connect (task); } static void socket_connect (GTask * task) { ConnectTaskData *data = g_task_get_task_data (task); GSocketConnectable *addr; GSocketClient *socket_client; if (data->location.timeout < 0) { data->location.timeout = DEFAULT_TIMEOUT; } if (data->error_handler_id) { g_signal_handler_disconnect (data->connection, data->error_handler_id); data->error_handler_id = 0; } if (data->connection) { gst_rtmp_connection_close (data->connection); g_clear_object (&data->connection); } if (!data->location.host) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_INITIALIZED, "Host is not set"); g_object_unref (task); return; } if (!data->location.port) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_INITIALIZED, "Port is not set"); g_object_unref (task); return; } socket_client = g_socket_client_new (); g_socket_client_set_timeout (socket_client, data->location.timeout); switch (data->location.scheme) { case GST_RTMP_SCHEME_RTMP: break; case GST_RTMP_SCHEME_RTMPS: GST_DEBUG ("Configuring TLS, validation flags 0x%02x", data->location.tls_flags); g_socket_client_set_tls (socket_client, TRUE); G_GNUC_BEGIN_IGNORE_DEPRECATIONS; g_socket_client_set_tls_validation_flags (socket_client, data->location.tls_flags); G_GNUC_END_IGNORE_DEPRECATIONS; break; default: g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_SUPPORTED, "Invalid scheme ID %d", data->location.scheme); g_object_unref (socket_client); g_object_unref (task); return; } addr = g_network_address_new (data->location.host, data->location.port); GST_DEBUG ("Starting socket connection"); g_socket_client_connect_async (socket_client, addr, g_task_get_cancellable (task), socket_connect_done, task); g_object_unref (addr); g_object_unref (socket_client); } static void socket_connect_done (GObject * source, GAsyncResult * result, gpointer user_data) { GSocketClient *socket_client = G_SOCKET_CLIENT (source); GSocketConnection *socket_connection; GTask *task = user_data; GError *error = NULL; socket_connection = g_socket_client_connect_finish (socket_client, result, &error); if (g_task_return_error_if_cancelled (task)) { GST_DEBUG ("Socket connection was cancelled"); g_object_unref (task); return; } if (socket_connection == NULL) { GST_ERROR ("Socket connection error"); g_task_return_error (task, error); g_object_unref (task); return; } GST_DEBUG ("Socket connection established"); gst_rtmp_client_handshake (G_IO_STREAM (socket_connection), FALSE, g_task_get_cancellable (task), handshake_done, task); g_object_unref (socket_connection); } static void handshake_done (GObject * source, GAsyncResult * result, gpointer user_data) { GIOStream *stream = G_IO_STREAM (source); GSocketConnection *socket_connection = G_SOCKET_CONNECTION (stream); GTask *task = user_data; ConnectTaskData *data = g_task_get_task_data (task); GError *error = NULL; gboolean res; res = gst_rtmp_client_handshake_finish (stream, result, &error); if (!res) { g_io_stream_close_async (stream, G_PRIORITY_DEFAULT, NULL, NULL, NULL); g_task_return_error (task, error); g_object_unref (task); return; } data->connection = gst_rtmp_connection_new (socket_connection, g_task_get_cancellable (task)); data->error_handler_id = g_signal_connect (data->connection, "error", G_CALLBACK (connection_error), task); send_connect (task); } static void connection_error (GstRtmpConnection * connection, gpointer user_data) { GTask *task = user_data; if (!g_task_had_error (task)) g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED, "error during connection attempt"); } static gchar * do_adobe_auth (const gchar * username, const gchar * password, const gchar * salt, const gchar * opaque, const gchar * challenge) { guint8 hash[16]; /* MD5 digest */ gsize hashlen = sizeof hash; gchar *challenge2, *auth_query; GChecksum *md5; g_return_val_if_fail (username, NULL); g_return_val_if_fail (password, NULL); g_return_val_if_fail (salt, NULL); md5 = g_checksum_new (G_CHECKSUM_MD5); g_checksum_update (md5, (guchar *) username, -1); g_checksum_update (md5, (guchar *) salt, -1); g_checksum_update (md5, (guchar *) password, -1); g_checksum_get_digest (md5, hash, &hashlen); g_warn_if_fail (hashlen == sizeof hash); { gchar *hashstr = g_base64_encode ((guchar *) hash, sizeof hash); g_checksum_reset (md5); g_checksum_update (md5, (guchar *) hashstr, -1); g_free (hashstr); } if (opaque) g_checksum_update (md5, (guchar *) opaque, -1); else if (challenge) g_checksum_update (md5, (guchar *) challenge, -1); challenge2 = g_strdup_printf ("%08x", g_random_int ()); g_checksum_update (md5, (guchar *) challenge2, -1); g_checksum_get_digest (md5, hash, &hashlen); g_warn_if_fail (hashlen == sizeof hash); { gchar *hashstr = g_base64_encode ((guchar *) hash, sizeof hash); if (opaque) { auth_query = g_strdup_printf ("authmod=%s&user=%s&challenge=%s&response=%s&opaque=%s", "adobe", username, challenge2, hashstr, opaque); } else { auth_query = g_strdup_printf ("authmod=%s&user=%s&challenge=%s&response=%s", "adobe", username, challenge2, hashstr); } g_free (hashstr); } g_checksum_free (md5); g_free (challenge2); return auth_query; } static void send_connect (GTask * task) { ConnectTaskData *data = g_task_get_task_data (task); GstAmfNode *node; const gchar *app, *flash_ver; gchar *uri, *appstr = NULL, *uristr = NULL; gboolean publish; node = gst_amf_node_new_object (); app = data->location.application; flash_ver = data->location.flash_ver; publish = data->location.publish; uri = gst_rtmp_location_get_string (&data->location, FALSE); if (!app) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_INITIALIZED, "Application is not set"); g_object_unref (task); goto out; } if (!flash_ver) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_INITIALIZED, "Flash version is not set"); g_object_unref (task); goto out; } if (data->auth_query) { const gchar *query = data->auth_query; appstr = g_strdup_printf ("%s?%s", app, query); uristr = g_strdup_printf ("%s?%s", uri, query); } else if (data->location.authmod == GST_RTMP_AUTHMOD_ADOBE) { const gchar *user = data->location.username; const gchar *authmod = "adobe"; if (!user) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED, "no username for adobe authentication"); g_object_unref (task); goto out; } if (!data->location.password) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED, "no password for adobe authentication"); g_object_unref (task); goto out; } appstr = g_strdup_printf ("%s?authmod=%s&user=%s", app, authmod, user); uristr = g_strdup_printf ("%s?authmod=%s&user=%s", uri, authmod, user); } else { appstr = g_strdup (app); uristr = g_strdup (uri); } /* Arguments for the connect command. * Most of these are described in rtmp_specification_1.0.pdf page 30 */ /* "The server application name the client is connected to." */ gst_amf_node_append_field_take_string (node, "app", appstr, -1); if (publish) { /* Undocumented. Sent by both libavformat and librtmp. */ gst_amf_node_append_field_string (node, "type", "nonprivate", -1); } /* "Flash Player version. It is the same string as returned by the * ApplicationScript getversion () function." */ gst_amf_node_append_field_string (node, "flashVer", flash_ver, -1); /* "URL of the source SWF file making the connection." * XXX: libavformat sends "swfUrl" here, if provided. */ /* "URL of the Server. It has the following format. * protocol://servername:port/appName/appInstance" */ gst_amf_node_append_field_take_string (node, "tcUrl", uristr, -1); if (!publish) { /* "True if proxy is being used." */ gst_amf_node_append_field_boolean (node, "fpad", FALSE); /* Undocumented. Sent by libavformat. */ gst_amf_node_append_field_number (node, "capabilities", 15 /* libavformat's magic number */ ); /* "Indicates what audio codecs the client supports." */ gst_amf_node_append_field_number (node, "audioCodecs", GST_RTMP_AUDIOCODECS); /* "Indicates what video codecs are supported." */ gst_amf_node_append_field_number (node, "videoCodecs", GST_RTMP_VIDEOCODECS); /* "Indicates what special video functions are supported." */ gst_amf_node_append_field_number (node, "videoFunction", GST_RTMP_VIDEOFUNCTION); /* "URL of the web page from where the SWF file was loaded." * XXX: libavformat sends "pageUrl" here, if provided. */ } gst_rtmp_connection_send_command (data->connection, send_connect_done, task, 0, "connect", node, NULL); out: gst_amf_node_free (node); g_free (uri); } static void send_connect_done (const gchar * command_name, GPtrArray * args, gpointer user_data) { GTask *task = G_TASK (user_data); ConnectTaskData *data = g_task_get_task_data (task); const GstAmfNode *node, *optional_args; const gchar *code; if (g_task_return_error_if_cancelled (task)) { g_object_unref (task); return; } if (!args) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED, "connect failed: %s", command_name); g_object_unref (task); return; } if (args->len < 2) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED, "connect failed; not enough return arguments"); g_object_unref (task); return; } optional_args = g_ptr_array_index (args, 1); node = gst_amf_node_get_field (optional_args, "code"); if (!node) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED, "result code missing from connect cmd result"); g_object_unref (task); return; } code = gst_amf_node_peek_string (node, NULL); GST_INFO ("connect result: %s", GST_STR_NULL (code)); if (g_str_equal (code, "NetConnection.Connect.Success")) { node = gst_amf_node_get_field (optional_args, "secureToken"); send_secure_token_response (task, data->connection, node ? gst_amf_node_peek_string (node, NULL) : NULL); return; } if (g_str_equal (code, "NetConnection.Connect.Rejected")) { GstRtmpAuthmod authmod = data->location.authmod; GMatchInfo *match_info; const gchar *desc; GstUri *query; node = gst_amf_node_get_field (optional_args, "description"); if (!node) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED, "Connect rejected; no description"); g_object_unref (task); return; } desc = gst_amf_node_peek_string (node, NULL); GST_DEBUG ("connect result desc: %s", GST_STR_NULL (desc)); if (authmod == GST_RTMP_AUTHMOD_AUTO && strstr (desc, "code=403 need auth")) { if (strstr (desc, "authmod=adobe")) { GST_INFO ("Reconnecting with authmod=adobe"); data->location.authmod = GST_RTMP_AUTHMOD_ADOBE; socket_connect (task); return; } g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED, "unhandled authentication mode: %s", desc); g_object_unref (task); return; } if (!g_regex_match (auth_regex, desc, 0, &match_info)) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED, "failed to parse auth rejection: %s", desc); g_object_unref (task); return; } { gchar *authmod_str = g_match_info_fetch_named (match_info, "authmod"); gchar *query_str = g_match_info_fetch_named (match_info, "query"); gboolean matches; GST_INFO ("regex parsed auth: authmod=%s, query=%s", GST_STR_NULL (authmod_str), GST_STR_NULL (query_str)); g_match_info_free (match_info); switch (authmod) { case GST_RTMP_AUTHMOD_ADOBE: matches = g_str_equal (authmod_str, "adobe"); break; default: matches = FALSE; break; } if (!matches) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED, "server uses wrong authentication mode '%s'; expected %s", GST_STR_NULL (authmod_str), gst_rtmp_authmod_get_nick (authmod)); g_object_unref (task); g_free (authmod_str); g_free (query_str); return; } g_free (authmod_str); query = gst_uri_from_string (query_str); if (!query) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED, "failed to parse authentication query '%s'", GST_STR_NULL (query_str)); g_object_unref (task); g_free (query_str); return; } g_free (query_str); } { const gchar *reason = gst_uri_get_query_value (query, "reason"); if (g_str_equal (reason, "authfailed")) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED, "authentication failed! wrong credentials?"); g_object_unref (task); gst_uri_unref (query); return; } if (!g_str_equal (reason, "needauth")) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED, "unhandled rejection reason '%s'", reason ? reason : ""); g_object_unref (task); gst_uri_unref (query); return; } } g_warn_if_fail (!data->auth_query); data->auth_query = do_adobe_auth (data->location.username, data->location.password, gst_uri_get_query_value (query, "salt"), gst_uri_get_query_value (query, "opaque"), gst_uri_get_query_value (query, "challenge")); gst_uri_unref (query); if (!data->auth_query) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED, "couldn't generate adobe style authentication query"); g_object_unref (task); return; } socket_connect (task); return; } g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED, "unhandled connect result code: %s", GST_STR_NULL (code)); g_object_unref (task); } /* prep key: pack 1st 16 chars into 4 LittleEndian ints */ static void rtmp_tea_decode_prep_key (const gchar * key, guint32 out[4]) { gchar copy[17]; g_return_if_fail (key); g_return_if_fail (out); /* ensure we can read 16 bytes */ strncpy (copy, key, 16); /* placate GCC 8 -Wstringop-truncation */ copy[16] = 0; out[0] = GST_READ_UINT32_LE (copy); out[1] = GST_READ_UINT32_LE (copy + 4); out[2] = GST_READ_UINT32_LE (copy + 8); out[3] = GST_READ_UINT32_LE (copy + 12); } /* prep text: hex2bin, each 8 digits -> 4 chars -> 1 uint32 */ static GArray * rtmp_tea_decode_prep_text (const gchar * text) { GArray *arr; gsize len, i; g_return_val_if_fail (text, NULL); len = strlen (text); arr = g_array_sized_new (TRUE, TRUE, 4, (len + 7) / 8); for (i = 0; i < len; i += 8) { gchar copy[9]; guchar chars[4]; gsize j; guint32 val; /* ensure we can read 8 bytes */ strncpy (copy, text + i, 8); /* placate GCC 8 -Wstringop-truncation */ copy[8] = 0; for (j = 0; j < 4; j++) { gint hi, lo; hi = g_ascii_xdigit_value (copy[2 * j]); lo = g_ascii_xdigit_value (copy[2 * j + 1]); chars[j] = (hi > 0 ? hi << 4 : 0) + (lo > 0 ? lo : 0); } val = GST_READ_UINT32_LE (chars); g_array_append_val (arr, val); } return arr; } /* return text from uint32s to chars */ static gchar * rtmp_tea_decode_return_text (GArray * arr) { #if G_BYTE_ORDER != G_LITTLE_ENDIAN gsize i; g_return_val_if_fail (arr, NULL); for (i = 0; i < arr->len; i++) { guint32 *val = &g_array_index (arr, guint32, i); *val = GUINT32_TO_LE (*val); } #endif /* array is alredy zero-terminated */ return g_array_free (arr, FALSE); } /* http://www.movable-type.co.uk/scripts/tea-block.html */ static void rtmp_tea_decode_btea (GArray * text, guint32 key[4]) { guint32 *v, n, *k; guint32 z, y, sum = 0, e, DELTA = 0x9e3779b9; guint32 p, q; g_return_if_fail (text); g_return_if_fail (text->len > 0); g_return_if_fail (key); v = (guint32 *) text->data; n = text->len; k = key; z = v[n - 1]; y = v[0]; q = 6 + 52 / n; sum = q * DELTA; #define MX ((z>>5^y<<2) + (y>>3^z<<4)) ^ ((sum^y) + (k[(p&3)^e]^z)); while (sum != 0) { e = sum >> 2 & 3; for (p = n - 1; p > 0; p--) z = v[p - 1], y = v[p] -= MX; z = v[n - 1]; y = v[0] -= MX; sum -= DELTA; } #undef MX } /* taken from librtmp */ static gchar * rtmp_tea_decode (const gchar * bin_key, const gchar * hex_text) { guint32 key[4]; GArray *text; rtmp_tea_decode_prep_key (bin_key, key); text = rtmp_tea_decode_prep_text (hex_text); rtmp_tea_decode_btea (text, key); return rtmp_tea_decode_return_text (text); } static void send_secure_token_response (GTask * task, GstRtmpConnection * connection, const gchar * challenge) { ConnectTaskData *data = g_task_get_task_data (task); if (challenge) { GstAmfNode *node1; GstAmfNode *node2; gchar *response; if (!data->location.secure_token || !data->location.secure_token[0]) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED, "server requires secure token authentication"); g_object_unref (task); return; } response = rtmp_tea_decode (data->location.secure_token, challenge); GST_DEBUG ("response: %s", response); node1 = gst_amf_node_new_null (); node2 = gst_amf_node_new_take_string (response, -1); gst_rtmp_connection_send_command (connection, NULL, NULL, 0, "secureTokenResponse", node1, node2, NULL); gst_amf_node_free (node1); gst_amf_node_free (node2); } g_signal_handler_disconnect (connection, data->error_handler_id); data->error_handler_id = 0; g_task_return_pointer (task, g_object_ref (connection), gst_rtmp_connection_close_and_unref); g_object_unref (task); } GstRtmpConnection * gst_rtmp_client_connect_finish (GAsyncResult * result, GError ** error) { GTask *task = G_TASK (result); return g_task_propagate_pointer (task, error); } static void send_create_stream (GTask * task); static void send_publish_or_play (GTask * task); typedef struct { GstRtmpConnection *connection; gulong error_handler_id; gchar *stream; gboolean publish; guint32 id; } StreamTaskData; static StreamTaskData * stream_task_data_new (GstRtmpConnection * connection, const gchar * stream, gboolean publish) { StreamTaskData *data = g_slice_new0 (StreamTaskData); data->connection = g_object_ref (connection); data->stream = g_strdup (stream); data->publish = publish; return data; } static void stream_task_data_free (gpointer ptr) { StreamTaskData *data = ptr; g_clear_pointer (&data->stream, g_free); if (data->error_handler_id) { g_signal_handler_disconnect (data->connection, data->error_handler_id); } g_clear_object (&data->connection); g_slice_free (StreamTaskData, data); } static void start_stream (GstRtmpConnection * connection, const gchar * stream, gboolean publish, GCancellable * cancellable, GAsyncReadyCallback callback, gpointer user_data) { GTask *task; StreamTaskData *data; init_debug (); task = g_task_new (connection, cancellable, callback, user_data); if (!stream) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_INITIALIZED, "Stream is not set"); g_object_unref (task); return; } data = stream_task_data_new (connection, stream, publish); g_task_set_task_data (task, data, stream_task_data_free); data->error_handler_id = g_signal_connect (connection, "error", G_CALLBACK (connection_error), task); send_create_stream (task); } void gst_rtmp_client_start_publish_async (GstRtmpConnection * connection, const gchar * stream, GCancellable * cancellable, GAsyncReadyCallback callback, gpointer user_data) { start_stream (connection, stream, TRUE, cancellable, callback, user_data); } void gst_rtmp_client_start_play_async (GstRtmpConnection * connection, const gchar * stream, GCancellable * cancellable, GAsyncReadyCallback callback, gpointer user_data) { start_stream (connection, stream, FALSE, cancellable, callback, user_data); } static void send_set_buffer_length (GstRtmpConnection * connection, guint32 stream, guint32 ms) { GstRtmpUserControl uc = { .type = GST_RTMP_USER_CONTROL_TYPE_SET_BUFFER_LENGTH, .param = stream, .param2 = ms, }; gst_rtmp_connection_queue_message (connection, gst_rtmp_message_new_user_control (&uc)); } static void send_create_stream (GTask * task) { GstRtmpConnection *connection = g_task_get_source_object (task); StreamTaskData *data = g_task_get_task_data (task); GstAmfNode *command_object, *stream_name; command_object = gst_amf_node_new_null (); stream_name = gst_amf_node_new_string (data->stream, -1); if (data->publish) { /* Not part of RTMP documentation */ GST_DEBUG ("Releasing stream '%s'", data->stream); gst_rtmp_connection_send_command (connection, NULL, NULL, 0, "releaseStream", command_object, stream_name, NULL); gst_rtmp_connection_send_command (connection, NULL, NULL, 0, "FCPublish", command_object, stream_name, NULL); } else { /* Matches librtmp */ gst_rtmp_connection_request_window_size (connection, GST_RTMP_DEFAULT_WINDOW_ACK_SIZE); send_set_buffer_length (connection, 0, 300); } GST_INFO ("Creating stream '%s'", data->stream); gst_rtmp_connection_send_command (connection, create_stream_done, task, 0, "createStream", command_object, NULL); gst_amf_node_free (stream_name); gst_amf_node_free (command_object); } static void create_stream_done (const gchar * command_name, GPtrArray * args, gpointer user_data) { GTask *task = G_TASK (user_data); StreamTaskData *data = g_task_get_task_data (task); GstAmfNode *result; if (g_task_return_error_if_cancelled (task)) { g_object_unref (task); return; } if (!args) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED, "createStream failed: %s", command_name); g_object_unref (task); return; } if (args->len < 2) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED, "createStream failed; not enough return arguments"); g_object_unref (task); return; } result = g_ptr_array_index (args, 1); if (gst_amf_node_get_type (result) != GST_AMF_TYPE_NUMBER) { GString *error_dump = g_string_new (""); gst_amf_node_dump (result, -1, error_dump); g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED, "createStream failed: %s", error_dump->str); g_object_unref (task); g_string_free (error_dump, TRUE); return; } data->id = gst_amf_node_get_number (result); GST_INFO ("createStream success, stream_id=%" G_GUINT32_FORMAT, data->id); if (data->id == 0) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_INVALID_DATA, "createStream returned ID 0"); g_object_unref (task); return; } send_publish_or_play (task); } static void send_publish_or_play (GTask * task) { GstRtmpConnection *connection = g_task_get_source_object (task); StreamTaskData *data = g_task_get_task_data (task); const gchar *command = data->publish ? "publish" : "play"; GstAmfNode *command_object, *stream_name, *argument; command_object = gst_amf_node_new_null (); stream_name = gst_amf_node_new_string (data->stream, -1); if (data->publish) { /* publishing type (live, record, append) */ argument = gst_amf_node_new_string ("live", -1); } else { /* "Start" argument: -2 = live or recording, -1 = only live 0 or positive = only recording, seek to X seconds */ argument = gst_amf_node_new_number (-2); } GST_INFO ("Sending %s for '%s' on stream %" G_GUINT32_FORMAT, command, data->stream, data->id); gst_rtmp_connection_expect_command (connection, on_publish_or_play_status, task, data->id, "onStatus"); gst_rtmp_connection_send_command (connection, NULL, NULL, data->id, command, command_object, stream_name, argument, NULL); if (!data->publish) { /* Matches librtmp */ send_set_buffer_length (connection, data->id, 30000); } gst_amf_node_free (command_object); gst_amf_node_free (stream_name); gst_amf_node_free (argument); } static void on_publish_or_play_status (const gchar * command_name, GPtrArray * args, gpointer user_data) { GTask *task = G_TASK (user_data); GstRtmpConnection *connection = g_task_get_source_object (task); StreamTaskData *data = g_task_get_task_data (task); const gchar *command = data->publish ? "publish" : "play", *code = NULL; GString *info_dump; if (g_task_return_error_if_cancelled (task)) { g_object_unref (task); return; } if (!args) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED, "%s failed: %s", command, command_name); g_object_unref (task); return; } if (args->len < 2) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED, "%s failed; not enough return arguments", command); g_object_unref (task); return; } { const GstAmfNode *info_object, *code_object; info_object = g_ptr_array_index (args, 1); code_object = gst_amf_node_get_field (info_object, "code"); if (code_object) { code = gst_amf_node_peek_string (code_object, NULL); } info_dump = g_string_new (""); gst_amf_node_dump (info_object, -1, info_dump); } if (data->publish) { if (g_strcmp0 (code, "NetStream.Publish.Start") == 0) { GST_INFO ("publish success: %s", info_dump->str); g_task_return_boolean (task, TRUE); goto out; } if (g_strcmp0 (code, "NetStream.Publish.BadName") == 0) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_EXISTS, "publish denied: stream already exists: %s", info_dump->str); goto out; } if (g_strcmp0 (code, "NetStream.Publish.Denied") == 0) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED, "publish denied: %s", info_dump->str); goto out; } } else { if (g_strcmp0 (code, "NetStream.Play.Start") == 0 || g_strcmp0 (code, "NetStream.Play.PublishNotify") == 0 || g_strcmp0 (code, "NetStream.Play.Reset") == 0) { GST_INFO ("play success: %s", info_dump->str); g_task_return_boolean (task, TRUE); goto out; } if (g_strcmp0 (code, "NetStream.Play.StreamNotFound") == 0) { g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_FOUND, "play denied: stream not found: %s", info_dump->str); goto out; } } g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED, "unhandled %s result: %s", command, info_dump->str); out: g_string_free (info_dump, TRUE); g_signal_handler_disconnect (connection, data->error_handler_id); data->error_handler_id = 0; g_object_unref (task); } static gboolean start_stream_finish (GstRtmpConnection * connection, GAsyncResult * result, guint32 * stream_id, GError ** error) { GTask *task; StreamTaskData *data; g_return_val_if_fail (g_task_is_valid (result, connection), FALSE); task = G_TASK (result); if (!g_task_propagate_boolean (G_TASK (result), error)) { return FALSE; } data = g_task_get_task_data (task); if (stream_id) { *stream_id = data->id; } return TRUE; } gboolean gst_rtmp_client_start_publish_finish (GstRtmpConnection * connection, GAsyncResult * result, guint32 * stream_id, GError ** error) { return start_stream_finish (connection, result, stream_id, error); } gboolean gst_rtmp_client_start_play_finish (GstRtmpConnection * connection, GAsyncResult * result, guint32 * stream_id, GError ** error) { return start_stream_finish (connection, result, stream_id, error); } void gst_rtmp_client_stop_publish (GstRtmpConnection * connection, const gchar * stream, const GstRtmpStopCommands stop_commands) { send_stop (connection, stream, stop_commands); } static void send_stop (GstRtmpConnection * connection, const gchar * stream, const GstRtmpStopCommands stop_commands) { GstAmfNode *command_object, *stream_name; command_object = gst_amf_node_new_null (); stream_name = gst_amf_node_new_string (stream, -1); if (stop_commands & GST_RTMP_STOP_COMMANDS_FCUNPUBLISH) { GST_DEBUG ("Sending stop command 'FCUnpublish' for stream '%s'", stream); gst_rtmp_connection_send_command (connection, NULL, NULL, 0, "FCUnpublish", command_object, stream_name, NULL); } if (stop_commands & GST_RTMP_STOP_COMMANDS_CLOSE_STREAM) { GST_DEBUG ("Sending stop command 'closeStream' for stream '%s'", stream); gst_rtmp_connection_send_command (connection, NULL, NULL, 0, "closeStream", command_object, stream_name, NULL); } if (stop_commands & GST_RTMP_STOP_COMMANDS_DELETE_STREAM) { GST_DEBUG ("Sending stop command 'deleteStream' for stream '%s'", stream); gst_rtmp_connection_send_command (connection, NULL, NULL, 0, "deleteStream", command_object, stream_name, NULL); } gst_amf_node_free (stream_name); gst_amf_node_free (command_object); }