ARI: Add recording controls

This patch implements the controls from ARI recordings. The controls
are:

 * DELETE /recordings/live/{recordingName} - stop recording and
   discard it
 * POST /recordings/live/{recordingName}/stop - stop recording
 * POST /recordings/live/{recordingName}/pause - pause recording
 * POST /recordings/live/{recordingName}/unpause - resume recording
 * POST /recordings/live/{recordingName}/mute - mute recording (record
   silence to the file)
 * POST /recordings/live/{recordingName}/unmute - unmute recording.

Since this underlying functionality did not already exist, is was
added to app.c by a set of control frames, similar to how playback
control works. The pause/mute control frames are toggles, even though
the ARI controls are idempotent, to be consistent with the playback
control frames.

(closes issue ASTERISK-22181)
Review: https://reviewboard.asterisk.org/r/2697/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396331 65c4cc65-6c06-0410-ace0-fbb531ad65f3

13 files changed, 466 insertions, 38 deletions

diff --git a/apps/app_minivm.c b/apps/app_minivm.c
index 06ac02a5e..cdcf8b933 100644
--- a/apps/app_minivm.c
+++ b/apps/app_minivm.c
@@ -1674,7 +1674,7 @@ static int play_record_review(struct ast_channel *chan, char *playfile, char *re
 				ast_channel_setoption(chan, AST_OPTION_RXGAIN, &record_gain, sizeof(record_gain), 0);
 			if (ast_test_flag(vmu, MVM_OPERATOR))
 				canceldtmf = "0";
-			cmd = ast_play_and_record_full(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, global_silencethreshold, global_maxsilence, unlockdir, acceptdtmf, canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
+			cmd = ast_play_and_record_full(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, 0, global_silencethreshold, global_maxsilence, unlockdir, acceptdtmf, canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
 			if (record_gain)
 				ast_channel_setoption(chan, AST_OPTION_RXGAIN, &zero_gain, sizeof(zero_gain), 0);
 			if (cmd == -1) /* User has hung up, no options to give */
diff --git a/apps/app_voicemail.c b/apps/app_voicemail.c
index 3696c740c..a258e6a0a 100644
--- a/apps/app_voicemail.c
+++ b/apps/app_voicemail.c
@@ -14682,7 +14682,7 @@ static int play_record_review(struct ast_channel *chan, char *playfile, char *re
 				ast_channel_setoption(chan, AST_OPTION_RXGAIN, &record_gain, sizeof(record_gain), 0);
 			if (ast_test_flag(vmu, VM_OPERATOR))
 				canceldtmf = "0";
-			cmd = ast_play_and_record_full(chan, playfile, tempfile, maxtime, fmt, duration, sound_duration, silencethreshold, maxsilence, unlockdir, acceptdtmf, canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
+			cmd = ast_play_and_record_full(chan, playfile, tempfile, maxtime, fmt, duration, sound_duration, 0, silencethreshold, maxsilence, unlockdir, acceptdtmf, canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
 			if (strchr(canceldtmf, cmd)) {
 			/* need this flag here to distinguish between pressing '0' during message recording or after */
 				canceleddtmf = 1;
diff --git a/funcs/func_frame_trace.c b/funcs/func_frame_trace.c
index 94a2c13d0..17f7c8965 100644
--- a/funcs/func_frame_trace.c
+++ b/funcs/func_frame_trace.c
@@ -343,6 +343,18 @@ static void print_frame(struct ast_frame *frame)
 		case AST_CONTROL_STREAM_FORWARD:
 			ast_verbose("SubClass: STREAM_FORWARD\n");
 			break;
+		case AST_CONTROL_RECORD_CANCEL:
+			ast_verbose("SubClass: RECORD_CANCEL\n");
+			break;
+		case AST_CONTROL_RECORD_STOP:
+			ast_verbose("SubClass: RECORD_STOP\n");
+			break;
+		case AST_CONTROL_RECORD_SUSPEND:
+			ast_verbose("SubClass: RECORD_SUSPEND\n");
+			break;
+		case AST_CONTROL_RECORD_MUTE:
+			ast_verbose("SubClass: RECORD_MUTE\n");
+			break;
 		}
 
 		if (frame->subclass.integer == -1) {
diff --git a/include/asterisk/app.h b/include/asterisk/app.h
index 251288546..06b903e2f 100644
--- a/include/asterisk/app.h
+++ b/include/asterisk/app.h
@@ -709,11 +709,12 @@ enum ast_record_if_exists {
  *        skip_confirmation_sound is false.
  *
  * \param chan the channel being recorded
- * \param playfile Filename of sound to play before recording begins
+ * \param playfile Filename of sound to play before recording begins. A beep is also played when playfile completes, before the recording begins.
  * \param recordfile Filename to save the recording
  * \param maxtime_sec Longest possible message length in seconds
  * \param fmt string containing all formats to be recorded delimited by '|'
  * \param duration pointer to integer for storing length of the recording
+ * \param beep If true, play a beep before recording begins (and doesn't play \a playfile)
  * \param sound_duration pointer to integer for storing length of the recording minus all silence
  * \param silencethreshold tolerance of noise levels that can be considered silence for the purpose of silence timeout, -1 for default
  * \param maxsilence_ms Length of time in milliseconds which will trigger a timeout from silence, -1 for default
@@ -728,7 +729,7 @@ enum ast_record_if_exists {
  * \retval 't' Recording ended from the message exceeding the maximum duration
  * \retval dtmfchar Recording ended via the return value's DTMF character for either cancel or accept.
  */
-int ast_play_and_record_full(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime_sec, const char *fmt, int *duration, int *sound_duration, int silencethreshold, int maxsilence_ms, const char *path, const char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound, enum ast_record_if_exists if_exists);
+int ast_play_and_record_full(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime_sec, const char *fmt, int *duration, int *sound_duration, int beep, int silencethreshold, int maxsilence_ms, const char *path, const char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound, enum ast_record_if_exists if_exists);
 
 /*!
  * \brief Record a file based on input from a channel. Use default accept and cancel DTMF.
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index bedc3a25d..1cb7d591f 100644
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -278,7 +278,11 @@ enum ast_control_frame_type {
 	AST_CONTROL_STREAM_RESTART = 1002,	/*!< Indicate to a channel in playback to restart the stream */
 	AST_CONTROL_STREAM_REVERSE = 1003,	/*!< Indicate to a channel in playback to rewind */
 	AST_CONTROL_STREAM_FORWARD = 1004,	/*!< Indicate to a channel in playback to fast forward */
-
+	/* Control frames to manipulate recording on a channel. */
+	AST_CONTROL_RECORD_CANCEL = 1100,	/*!< Indicated to a channel in record to stop recording and discard the file */
+	AST_CONTROL_RECORD_STOP = 1101,	/*!< Indicated to a channel in record to stop recording */
+	AST_CONTROL_RECORD_SUSPEND = 1102,	/*!< Indicated to a channel in record to suspend/unsuspend recording */
+	AST_CONTROL_RECORD_MUTE = 1103,	/*!< Indicated to a channel in record to mute/unmute (i.e. write silence) recording */
 };
 
 enum ast_frame_read_action {
diff --git a/include/asterisk/stasis_app_recording.h b/include/asterisk/stasis_app_recording.h
index 9c9930406..e8b4558ab 100644
--- a/include/asterisk/stasis_app_recording.h
+++ b/include/asterisk/stasis_app_recording.h
@@ -44,14 +44,30 @@ enum stasis_app_recording_state {
 	STASIS_APP_RECORDING_STATE_PAUSED,
 	/*! The media has stopped recording */
 	STASIS_APP_RECORDING_STATE_COMPLETE,
-	/*! The media has stopped playing */
+	/*! The media has stopped recording, with error */
 	STASIS_APP_RECORDING_STATE_FAILED,
+	/*! The media has stopped recording, discard the recording file */
+	STASIS_APP_RECORDING_STATE_CANCELED,
+	/*! Sentinel */
+	STASIS_APP_RECORDING_STATE_MAX,
 };
 
 /*! Valid operation for controlling a recording. */
 enum stasis_app_recording_media_operation {
-	/*! Stop the recording operation. */
+	/*! Stop the recording, deleting the media file(s) */
+	STASIS_APP_RECORDING_CANCEL,
+	/*! Stop the recording. */
 	STASIS_APP_RECORDING_STOP,
+	/*! Pause the recording */
+	STASIS_APP_RECORDING_PAUSE,
+	/*! Unpause the recording */
+	STASIS_APP_RECORDING_UNPAUSE,
+	/*! Mute the recording (record silence) */
+	STASIS_APP_RECORDING_MUTE,
+	/*! Unmute the recording */
+	STASIS_APP_RECORDING_UNMUTE,
+	/*! Sentinel */
+	STASIS_APP_RECORDING_OPER_MAX,
 };
 
 #define STASIS_APP_RECORDING_TERMINATE_INVALID 0
diff --git a/main/app.c b/main/app.c
index 8d081fe8c..ee2bbf467 100644
--- a/main/app.c
+++ b/main/app.c
@@ -1145,6 +1145,78 @@ int ast_play_and_wait(struct ast_channel *chan, const char *fn)
 	return d;
 }
 
+/*!
+ * \brief Construct a silence frame of the same duration as \a orig.
+ *
+ * The \a orig frame must be \ref AST_FORMAT_SLINEAR.
+ *
+ * \param orig Frame as basis for silence to generate.
+ * \return New frame of silence; free with ast_frfree().
+ * \return \c NULL on error.
+ */
+static struct ast_frame *make_silence(const struct ast_frame *orig)
+{
+	struct ast_frame *silence;
+	size_t size;
+	size_t datalen;
+	size_t samples = 0;
+	struct ast_frame *next;
+
+	if (!orig) {
+		return NULL;
+	}
+
+	if (orig->subclass.format.id != AST_FORMAT_SLINEAR) {
+		ast_log(LOG_WARNING, "Attempting to silence non-slin frame\n");
+		return NULL;
+	}
+
+	for (next = AST_LIST_NEXT(orig, frame_list);
+		 orig;
+		 orig = next, next = orig ? AST_LIST_NEXT(orig, frame_list) : NULL) {
+		samples += orig->samples;
+	}
+
+	ast_verb(4, "Silencing %zd samples\n", samples);
+
+
+	datalen = sizeof(short) * samples;
+	size = sizeof(*silence) + datalen;
+	silence = ast_calloc(1, size);
+	if (!silence) {
+		return NULL;
+	}
+
+	silence->mallocd = AST_MALLOCD_HDR;
+	silence->frametype = AST_FRAME_VOICE;
+	silence->data.ptr = (void *)(silence + 1);
+	silence->samples = samples;
+	silence->datalen = datalen;
+
+	ast_format_set(&silence->subclass.format, AST_FORMAT_SLINEAR, 0);
+
+	return silence;
+}
+
+/*!
+ * \brief Sets a channel's read format to \ref AST_FORMAT_SLINEAR, recording
+ * its original format.
+ *
+ * \param chan Channel to modify.
+ * \param[out] orig_format Output variable to store channel's original read
+ *                         format.
+ * \return 0 on success.
+ * \return -1 on error.
+ */
+static int set_read_to_slin(struct ast_channel *chan, struct ast_format *orig_format)
+{
+	if (!chan || !orig_format) {
+		return -1;
+	}
+	ast_format_copy(orig_format, ast_channel_readformat(chan));
+	return ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR);
+}
+
 static int global_silence_threshold = 128;
 static int global_maxsilence = 0;
 
@@ -1274,8 +1346,7 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
 			return -1;
 		}
 		ast_dsp_set_threshold(sildet, silencethreshold);
-		ast_format_copy(&rfmt, ast_channel_readformat(chan));
-		res = ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR);
+		res = set_read_to_slin(chan, &rfmt);
 		if (res < 0) {
 			ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
 			ast_dsp_free(sildet);
@@ -1293,9 +1364,15 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
 	}
 
 	if (x == fmtcnt) {
-		/* Loop forever, writing the packets we read to the writer(s), until
-		   we read a digit or get a hangup */
+		/* Loop, writing the packets we read to the writer(s), until
+		 * we have reason to stop. */
 		struct ast_frame *f;
+		int paused = 0;
+		int muted = 0;
+		time_t pause_start = 0;
+		int paused_secs = 0;
+		int pausedsilence = 0;
+
 		for (;;) {
 			if (!(res = ast_waitfor(chan, 2000))) {
 				ast_debug(1, "One waitfor failed, trying another\n");
@@ -1315,11 +1392,29 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
 			}
 			if (f->frametype == AST_FRAME_VOICE) {
 				/* write each format */
-				for (x = 0; x < fmtcnt; x++) {
-					if (prepend && !others[x]) {
-						break;
+				if (paused) {
+					/* It's all good */
+					res = 0;
+				} else {
+					RAII_VAR(struct ast_frame *, silence, NULL, ast_frame_dtor);
+					struct ast_frame *orig = f;
+
+					if (muted) {
+						silence = make_silence(orig);
+						if (!silence) {
+							ast_log(LOG_WARNING,
+								"Error creating silence\n");
+							break;
+						}
+						f = silence;
 					}
-					res = ast_writestream(others[x], f);
+					for (x = 0; x < fmtcnt; x++) {
+						if (prepend && !others[x]) {
+							break;
+						}
+						res = ast_writestream(others[x], f);
+					}
+					f = orig;
 				}
 
 				/* Silence Detection */
@@ -1331,6 +1426,17 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
 					}
 					olddspsilence = dspsilence;
 
+					if (paused) {
+						/* record how much silence there was while we are paused */
+						pausedsilence = dspsilence;
+					} else if (dspsilence > pausedsilence) {
+						/* ignore the paused silence */
+						dspsilence -= pausedsilence;
+					} else {
+						/* dspsilence has reset, reset pausedsilence */
+						pausedsilence = 0;
+					}
+
 					if (dspsilence > maxsilence) {
 						/* Ended happily with silence */
 						ast_verb(3, "Recording automatically stopped after a silence of %d seconds\n", dspsilence/1000);
@@ -1362,15 +1468,51 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
 					break;
 				}
 				if (strchr(canceldtmf, f->subclass.integer)) {
-					ast_verb(3, "User cancelled message by pressing %c\n", f->subclass.integer);
+					ast_verb(3, "User canceled message by pressing %c\n", f->subclass.integer);
 					res = f->subclass.integer;
 					outmsg = 0;
 					break;
 				}
+			} else if (f->frametype == AST_FRAME_CONTROL) {
+				if (f->subclass.integer == AST_CONTROL_RECORD_CANCEL) {
+					ast_verb(3, "Message canceled by control\n");
+					outmsg = 0; /* cancels the recording */
+					res = 0;
+					break;
+				} else if (f->subclass.integer == AST_CONTROL_RECORD_STOP) {
+					ast_verb(3, "Message ended by control\n");
+					res = 0;
+					break;
+				} else if (f->subclass.integer == AST_CONTROL_RECORD_SUSPEND) {
+					paused = !paused;
+					ast_verb(3, "Message %spaused by control\n",
+						paused ? "" : "un");
+					if (paused) {
+						pause_start = time(NULL);
+					} else {
+						paused_secs += time(NULL) - pause_start;
+					}
+				} else if (f->subclass.integer == AST_CONTROL_RECORD_MUTE) {
+					muted = !muted;
+					ast_verb(3, "Message %smuted by control\n",
+						muted ? "" : "un");
+					/* We can only silence slin frames, so
+					 * set the mode, if we haven't already
+					 * for sildet
+					 */
+					if (muted && !rfmt.id) {
+						ast_verb(3, "Setting read format to linear mode\n");
+						res = set_read_to_slin(chan, &rfmt);
+						if (res < 0) {
+							ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
+							break;
+						}
+					}
+				}
 			}
-			if (maxtime) {
+			if (maxtime && !paused) {
 				end = time(NULL);
-				if (maxtime < (end - start)) {
+				if (maxtime < (end - start - paused_secs)) {
 					ast_verb(3, "Took too long, cutting it short...\n");
 					res = 't';
 					outmsg = 2;
@@ -1493,9 +1635,9 @@ static int __ast_play_and_record(struct ast_channel *chan, const char *playfile,
 static const char default_acceptdtmf[] = "#";
 static const char default_canceldtmf[] = "";
 
-int ast_play_and_record_full(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int *sound_duration, int silencethreshold, int maxsilence, const char *path, const char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound, enum ast_record_if_exists if_exists)
+int ast_play_and_record_full(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int *sound_duration, int beep, int silencethreshold, int maxsilence, const char *path, const char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound, enum ast_record_if_exists if_exists)
 {
-	return __ast_play_and_record(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, 0, silencethreshold, maxsilence, path, 0, S_OR(acceptdtmf, default_acceptdtmf), S_OR(canceldtmf, default_canceldtmf), skip_confirmation_sound, if_exists);
+	return __ast_play_and_record(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, beep, silencethreshold, maxsilence, path, 0, S_OR(acceptdtmf, default_acceptdtmf), S_OR(canceldtmf, default_canceldtmf), skip_confirmation_sound, if_exists);
 }
 
 int ast_play_and_record(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int *sound_duration, int silencethreshold, int maxsilence, const char *path)
diff --git a/main/channel.c b/main/channel.c
index f1d98de12..8e6a4c945 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -4303,6 +4303,10 @@ static int attribute_const is_visible_indication(enum ast_control_frame_type con
 	case AST_CONTROL_STREAM_REVERSE:
 	case AST_CONTROL_STREAM_FORWARD:
 	case AST_CONTROL_STREAM_RESTART:
+	case AST_CONTROL_RECORD_CANCEL:
+	case AST_CONTROL_RECORD_STOP:
+	case AST_CONTROL_RECORD_SUSPEND:
+	case AST_CONTROL_RECORD_MUTE:
 		break;
 
 	case AST_CONTROL_INCOMPLETE:
@@ -4561,6 +4565,10 @@ int ast_indicate_data(struct ast_channel *chan, int _condition,
 	case AST_CONTROL_STREAM_REVERSE:
 	case AST_CONTROL_STREAM_FORWARD:
 	case AST_CONTROL_STREAM_RESTART:
+	case AST_CONTROL_RECORD_CANCEL:
+	case AST_CONTROL_RECORD_STOP:
+	case AST_CONTROL_RECORD_SUSPEND:
+	case AST_CONTROL_RECORD_MUTE:
 		/* Nothing left to do for these. */
 		res = 0;
 		break;
diff --git a/res/ari/resource_recordings.c b/res/ari/resource_recordings.c
index 46439ff0b..f87ff0a0a 100644
--- a/res/ari/resource_recordings.c
+++ b/res/ari/resource_recordings.c
@@ -71,27 +71,81 @@ void ast_ari_get_live_recording(struct ast_variable *headers,
 	ast_ari_response_ok(response, ast_json_ref(json));
 }
 
-void ast_ari_cancel_recording(struct ast_variable *headers, struct ast_cancel_recording_args *args, struct ast_ari_response *response)
+static void control_recording(const char *name,
+	enum stasis_app_recording_media_operation operation,
+	struct ast_ari_response *response)
 {
-	ast_log(LOG_ERROR, "TODO: ast_ari_cancel_recording\n");
+	RAII_VAR(struct stasis_app_recording *, recording, NULL, ao2_cleanup);
+	RAII_VAR(struct ast_json *, json, NULL, ast_json_unref);
+	enum stasis_app_recording_oper_results res;
+
+	recording = stasis_app_recording_find_by_name(name);
+	if (recording == NULL) {
+		ast_ari_response_error(response, 404, "Not Found",
+			"Recording not found");
+		return;
+	}
+
+	res = stasis_app_recording_operation(recording, operation);
+
+	switch (res) {
+	case STASIS_APP_RECORDING_OPER_OK:
+		ast_ari_response_no_content(response);
+		return;
+	case STASIS_APP_RECORDING_OPER_FAILED:
+		ast_ari_response_error(response, 500,
+			"Internal Server Error", "Recording operation failed");
+		return;
+	case STASIS_APP_RECORDING_OPER_NOT_RECORDING:
+		ast_ari_response_error(response, 409,
+			"Conflict", "Recording not in session");
+	}
 }
-void ast_ari_stop_recording(struct ast_variable *headers, struct ast_stop_recording_args *args, struct ast_ari_response *response)
+
+void ast_ari_cancel_recording(struct ast_variable *headers,
+	struct ast_cancel_recording_args *args,
+	struct ast_ari_response *response)
 {
-	ast_log(LOG_ERROR, "TODO: ast_ari_stop_recording\n");
+	control_recording(args->recording_name, STASIS_APP_RECORDING_CANCEL,
+		response);
 }
-void ast_ari_pause_recording(struct ast_variable *headers, struct ast_pause_recording_args *args, struct ast_ari_response *response)
+
+void ast_ari_stop_recording(struct ast_variable *headers,
+	struct ast_stop_recording_args *args,
+	struct ast_ari_response *response)
+{
+	control_recording(args->recording_name, STASIS_APP_RECORDING_STOP,
+		response);
+}
+
+void ast_ari_pause_recording(struct ast_variable *headers,
+	struct ast_pause_recording_args *args,
+	struct ast_ari_response *response)
 {
-	ast_log(LOG_ERROR, "TODO: ast_ari_pause_recording\n");
+	control_recording(args->recording_name, STASIS_APP_RECORDING_PAUSE,
+		response);
 }
-void ast_ari_unpause_recording(struct ast_variable *headers, struct ast_unpause_recording_args *args, struct ast_ari_response *response)
+
+void ast_ari_unpause_recording(struct ast_variable *headers,
+	struct ast_unpause_recording_args *args,
+	struct ast_ari_response *response)
 {
-	ast_log(LOG_ERROR, "TODO: ast_ari_unpause_recording\n");
+	control_recording(args->recording_name, STASIS_APP_RECORDING_UNPAUSE,
+		response);
 }
-void ast_ari_mute_recording(struct ast_variable *headers, struct ast_mute_recording_args *args, struct ast_ari_response *response)
+
+void ast_ari_mute_recording(struct ast_variable *headers,
+	struct ast_mute_recording_args *args,
+	struct ast_ari_response *response)
 {
-	ast_log(LOG_ERROR, "TODO: ast_ari_mute_recording\n");
+	control_recording(args->recording_name, STASIS_APP_RECORDING_MUTE,
+		response);
 }
-void ast_ari_unmute_recording(struct ast_variable *headers, struct ast_unmute_recording_args *args, struct ast_ari_response *response)
+
+void ast_ari_unmute_recording(struct ast_variable *headers,
+	struct ast_unmute_recording_args *args,
+	struct ast_ari_response *response)
 {
-	ast_log(LOG_ERROR, "TODO: ast_ari_unmute_recording\n");
+	control_recording(args->recording_name, STASIS_APP_RECORDING_UNMUTE,
+		response);
 }
diff --git a/res/ari/resource_recordings.h b/res/ari/resource_recordings.h
index e3ee88be8..2529766e7 100644
--- a/res/ari/resource_recordings.h
+++ b/res/ari/resource_recordings.h
@@ -134,7 +134,7 @@ struct ast_pause_recording_args {
 /*!
  * \brief Pause a live recording.
  *
- * Pausing a recording suspends silence detection, which will be restarted when the recording is unpaused.
+ * Pausing a recording suspends silence detection, which will be restarted when the recording is unpaused. Paused time is not included in the accounting for maxDurationSeconds.
  *
  * \param headers HTTP headers
  * \param args Swagger parameters
diff --git a/res/res_ari_recordings.c b/res/res_ari_recordings.c
index 01ad49a65..77fc830cf 100644
--- a/res/res_ari_recordings.c
+++ b/res/res_ari_recordings.c
@@ -295,6 +295,7 @@ static void ast_ari_get_live_recording_cb(
 		break;
 	case 500: /* Internal Server Error */
 	case 501: /* Not Implemented */
+	case 404: /* Recording not found */
 		is_valid = 1;
 		break;
 	default:
@@ -351,6 +352,7 @@ static void ast_ari_cancel_recording_cb(
 		break;
 	case 500: /* Internal Server Error */
 	case 501: /* Not Implemented */
+	case 404: /* Recording not found */
 		is_valid = 1;
 		break;
 	default:
@@ -407,6 +409,7 @@ static void ast_ari_stop_recording_cb(
 		break;
 	case 500: /* Internal Server Error */
 	case 501: /* Not Implemented */
+	case 404: /* Recording not found */
 		is_valid = 1;
 		break;
 	default:
@@ -463,6 +466,8 @@ static void ast_ari_pause_recording_cb(
 		break;
 	case 500: /* Internal Server Error */
 	case 501: /* Not Implemented */
+	case 404: /* Recording not found */
+	case 409: /* Recording not in session */
 		is_valid = 1;
 		break;
 	default:
@@ -519,6 +524,8 @@ static void ast_ari_unpause_recording_cb(
 		break;
 	case 500: /* Internal Server Error */
 	case 501: /* Not Implemented */
+	case 404: /* Recording not found */
+	case 409: /* Recording not in session */
 		is_valid = 1;
 		break;
 	default:
@@ -575,6 +582,8 @@ static void ast_ari_mute_recording_cb(
 		break;
 	case 500: /* Internal Server Error */
 	case 501: /* Not Implemented */
+	case 404: /* Recording not found */
+	case 409: /* Recording not in session */
 		is_valid = 1;
 		break;
 	default:
@@ -631,6 +640,8 @@ static void ast_ari_unmute_recording_cb(
 		break;
 	case 500: /* Internal Server Error */
 	case 501: /* Not Implemented */
+	case 404: /* Recording not found */
+	case 409: /* Recording not in session */
 		is_valid = 1;
 		break;
 	default:
diff --git a/res/res_stasis_recording.c b/res/res_stasis_recording.c
index f94f42895..575ccae1e 100644
--- a/res/res_stasis_recording.c
+++ b/res/res_stasis_recording.c
@@ -59,11 +59,13 @@ struct stasis_app_recording {
 	struct stasis_app_recording_options *options;
 	/*! Absolute path (minus extension) of the recording */
 	char *absolute_name;
-	/*! Control object for the channel we're playing back to */
+	/*! Control object for the channel we're recording */
 	struct stasis_app_control *control;
 
 	/*! Current state of the recording. */
 	enum stasis_app_recording_state state;
+	/*! Indicates whether the recording is currently muted */
+	int muted:1;
 };
 
 static int recording_hash(const void *obj, int flags)
@@ -99,6 +101,10 @@ static const char *state_to_string(enum stasis_app_recording_state state)
 		return "done";
 	case STASIS_APP_RECORDING_STATE_FAILED:
 		return "failed";
+	case STASIS_APP_RECORDING_STATE_CANCELED:
+		return "canceled";
+	case STASIS_APP_RECORDING_STATE_MAX:
+		return "?";
 	}
 
 	return "?";
@@ -253,12 +259,13 @@ static void *record_file(struct stasis_app_control *control,
 	}
 
 	ast_play_and_record_full(chan,
-		recording->options->beep ? "beep" : NULL,
+		NULL, /* playfile */
 		recording->absolute_name,
 		recording->options->max_duration_seconds,
 		recording->options->format,
 		&duration,
 		NULL, /* sound_duration */
+		recording->options->beep,
 		-1, /* silencethreshold */
 		recording->options->max_silence_seconds * 1000,
 		NULL, /* path */
@@ -403,12 +410,127 @@ struct ast_json *stasis_app_recording_to_json(
 	return ast_json_ref(json);
 }
 
+typedef int (*recording_operation_cb)(struct stasis_app_recording *recording);
+
+static int recording_noop(struct stasis_app_recording *recording)
+{
+	return 0;
+}
+
+static int recording_disregard(struct stasis_app_recording *recording)
+{
+	recording->state = STASIS_APP_RECORDING_STATE_CANCELED;
+	return 0;
+}
+
+static int recording_cancel(struct stasis_app_recording *recording)
+{
+	int res = 0;
+	recording->state = STASIS_APP_RECORDING_STATE_CANCELED;
+	res |= stasis_app_control_queue_control(recording->control,
+		AST_CONTROL_RECORD_CANCEL);
+	res |= ast_filedelete(recording->absolute_name, NULL);
+	return res;
+}
+
+static int recording_stop(struct stasis_app_recording *recording)
+{
+	recording->state = STASIS_APP_RECORDING_STATE_COMPLETE;
+	return stasis_app_control_queue_control(recording->control,
+		AST_CONTROL_RECORD_STOP);
+}
+
+static int recording_pause(struct stasis_app_recording *recording)
+{
+	recording->state = STASIS_APP_RECORDING_STATE_PAUSED;
+	return stasis_app_control_queue_control(recording->control,
+		AST_CONTROL_RECORD_SUSPEND);
+}
+
+static int recording_unpause(struct stasis_app_recording *recording)
+{
+	recording->state = STASIS_APP_RECORDING_STATE_RECORDING;
+	return stasis_app_control_queue_control(recording->control,
+		AST_CONTROL_RECORD_SUSPEND);
+}
+
+static int recording_mute(struct stasis_app_recording *recording)
+{
+	if (recording->muted) {
+		/* already muted */
+		return 0;
+	}
+
+	recording->muted = 1;
+	return stasis_app_control_queue_control(recording->control,
+		AST_CONTROL_RECORD_MUTE);
+}
+
+static int recording_unmute(struct stasis_app_recording *recording)
+{
+	if (!recording->muted) {
+		/* already unmuted */
+		return 0;
+	}
+
+	return stasis_app_control_queue_control(recording->control,
+		AST_CONTROL_RECORD_MUTE);
+}
+
+recording_operation_cb operations[STASIS_APP_RECORDING_STATE_MAX][STASIS_APP_RECORDING_OPER_MAX] = {
+	[STASIS_APP_RECORDING_STATE_QUEUED][STASIS_APP_RECORDING_CANCEL] = recording_disregard,
+	[STASIS_APP_RECORDING_STATE_QUEUED][STASIS_APP_RECORDING_STOP] = recording_disregard,
+	[STASIS_APP_RECORDING_STATE_RECORDING][STASIS_APP_RECORDING_CANCEL] = recording_cancel,
+	[STASIS_APP_RECORDING_STATE_RECORDING][STASIS_APP_RECORDING_STOP] = recording_stop,
+	[STASIS_APP_RECORDING_STATE_RECORDING][STASIS_APP_RECORDING_PAUSE] = recording_pause,
+	[STASIS_APP_RECORDING_STATE_RECORDING][STASIS_APP_RECORDING_UNPAUSE] = recording_noop,
+	[STASIS_APP_RECORDING_STATE_RECORDING][STASIS_APP_RECORDING_MUTE] = recording_mute,
+	[STASIS_APP_RECORDING_STATE_RECORDING][STASIS_APP_RECORDING_UNMUTE] = recording_unmute,
+	[STASIS_APP_RECORDING_STATE_PAUSED][STASIS_APP_RECORDING_CANCEL] = recording_cancel,
+	[STASIS_APP_RECORDING_STATE_PAUSED][STASIS_APP_RECORDING_STOP] = recording_stop,
+	[STASIS_APP_RECORDING_STATE_PAUSED][STASIS_APP_RECORDING_PAUSE] = recording_noop,
+	[STASIS_APP_RECORDING_STATE_PAUSED][STASIS_APP_RECORDING_UNPAUSE] = recording_unpause,
+	[STASIS_APP_RECORDING_STATE_PAUSED][STASIS_APP_RECORDING_MUTE] = recording_mute,
+	[STASIS_APP_RECORDING_STATE_PAUSED][STASIS_APP_RECORDING_UNMUTE] = recording_unmute,
+};
+
 enum stasis_app_recording_oper_results stasis_app_recording_operation(
 	struct stasis_app_recording *recording,
 	enum stasis_app_recording_media_operation operation)
 {
-	ast_assert(0); // TODO
-	return STASIS_APP_RECORDING_OPER_FAILED;
+	recording_operation_cb cb;
+	SCOPED_AO2LOCK(lock, recording);
+
+	if (recording->state < 0 || recording->state >= STASIS_APP_RECORDING_STATE_MAX) {
+		ast_log(LOG_WARNING, "Invalid recording state %d\n",
+			recording->state);
+		return -1;
+	}
+
+	if (operation < 0 || operation >= STASIS_APP_RECORDING_OPER_MAX) {
+		ast_log(LOG_WARNING, "Invalid recording operation %d\n",
+			operation);
+		return -1;
+	}
+
+	cb = operations[recording->state][operation];
+
+	if (!cb) {
+		if (recording->state != STASIS_APP_RECORDING_STATE_RECORDING) {
+			/* So we can be specific in our error message. */
+			return STASIS_APP_RECORDING_OPER_NOT_RECORDING;
+		} else {
+			/* And, really, all operations should be valid during
+			 * recording */
+			ast_log(LOG_ERROR,
+				"Unhandled operation during recording: %d\n",
+				operation);
+			return STASIS_APP_RECORDING_OPER_FAILED;
+		}
+	}
+
+	return cb(recording) ?
+		STASIS_APP_RECORDING_OPER_FAILED : STASIS_APP_RECORDING_OPER_OK;
 }
 
 static int load_module(void)
diff --git a/rest-api/api-docs/recordings.json b/rest-api/api-docs/recordings.json
index b564edee0..b4dd6d090 100644
--- a/rest-api/api-docs/recordings.json
+++ b/rest-api/api-docs/recordings.json
@@ -87,6 +87,12 @@
 							"allowMultiple": false,
 							"dataType": "string"
 						}
+					],
+					"errorResponses": [
+						{
+							"code": 404,
+							"reason": "Recording not found"
+						}
 					]
 				},
 				{
@@ -103,6 +109,12 @@
 							"allowMultiple": false,
 							"dataType": "string"
 						}
+					],
+					"errorResponses": [
+						{
+							"code": 404,
+							"reason": "Recording not found"
+						}
 					]
 				}
 			]
@@ -124,6 +136,12 @@
 							"allowMultiple": false,
 							"dataType": "string"
 						}
+					],
+					"errorResponses": [
+						{
+							"code": 404,
+							"reason": "Recording not found"
+						}
 					]
 				}
 			]
@@ -134,7 +152,7 @@
 				{
 					"httpMethod": "POST",
 					"summary": "Pause a live recording.",
-					"notes": "Pausing a recording suspends silence detection, which will be restarted when the recording is unpaused.",
+					"notes": "Pausing a recording suspends silence detection, which will be restarted when the recording is unpaused. Paused time is not included in the accounting for maxDurationSeconds.",
 					"nickname": "pauseRecording",
 					"responseClass": "void",
 					"parameters": [
@@ -146,6 +164,16 @@
 							"allowMultiple": false,
 							"dataType": "string"
 						}
+					],
+					"errorResponses": [
+						{
+							"code": 404,
+							"reason": "Recording not found"
+						},
+						{
+							"code": 409,
+							"reason": "Recording not in session"
+						}
 					]
 				}
 			]
@@ -167,6 +195,16 @@
 							"allowMultiple": false,
 							"dataType": "string"
 						}
+					],
+					"errorResponses": [
+						{
+							"code": 404,
+							"reason": "Recording not found"
+						},
+						{
+							"code": 409,
+							"reason": "Recording not in session"
+						}
 					]
 				}
 			]
@@ -189,6 +227,16 @@
 							"allowMultiple": false,
 							"dataType": "string"
 						}
+					],
+					"errorResponses": [
+						{
+							"code": 404,
+							"reason": "Recording not found"
+						},
+						{
+							"code": 409,
+							"reason": "Recording not in session"
+						}
 					]
 				}
 			]
@@ -210,6 +258,16 @@
 							"allowMultiple": false,
 							"dataType": "string"
 						}
+					],
+					"errorResponses": [
+						{
+							"code": 404,
+							"reason": "Recording not found"
+						},
+						{
+							"code": 409,
+							"reason": "Recording not in session"
+						}
 					]
 				}
 			]