diff --git a/janus/src/client.c b/janus/src/client.c
index 526b446..ac686b6 100644
--- a/janus/src/client.c
+++ b/janus/src/client.c
@@ -103,14 +103,13 @@ static void *_common_thread(void *v_client, bool video) {
 				packet.buffer = (char *)rtp->datagram;
 				packet.length = rtp->used;
 				janus_plugin_rtp_extensions_reset(&packet.extensions);
-				// FIXME: See rtpv.c
-				// Это очень эффективный способ уменьшить задержку, но WebRTC стек в хроме и фоксе
-				// слишком корявый, чтобы обработать это, тз-за чего на кейфреймах начинаются заикания.
+				// FIXME: Это очень эффективный способ уменьшить задержку, но WebRTC стек в хроме и фоксе
+				// слишком корявый, чтобы обработать это, из-за чего на кейфреймах начинаются заикания.
 				//   - https://github.com/Glimesh/janus-ftl-plugin/issues/101
-				/*if (video) {
+				if (rtp->zero_playout_delay) {
 					packet.extensions.min_delay = 0;
 					packet.extensions.max_delay = 0;
-				}*/
+				}
 				client->gw->relay_rtp(client->session, &packet);
 			}
 			rtp_destroy(rtp);
diff --git a/janus/src/config.c b/janus/src/config.c
index 1bfef2d..4c6776b 100644
--- a/janus/src/config.c
+++ b/janus/src/config.c
@@ -24,6 +24,7 @@
 
 
 static char *_get_value(janus_config *jcfg, const char *section, const char *option);
+static bool _get_bool(janus_config *jcfg, const char *section, const char *option, bool def);
 
 
 plugin_config_s *plugin_config_init(const char *config_dir_path) {
@@ -50,6 +51,9 @@ plugin_config_s *plugin_config_init(const char *config_dir_path) {
 		JLOG_ERROR("config", "Missing config value: video.sink (ex. memsink.object)");
 		goto error;
 	}
+	if ((config->video_zero_playout_delay = _get_bool(jcfg, "video", "zero_playout_delay", false)) == true) {
+		JLOG_INFO("config", "Enabled the experimental Playout-Delay=0 RTP extension for VIDEO");
+	}
 	if ((config->audio_dev_name = _get_value(jcfg, "audio", "device")) != NULL) {
 		JLOG_INFO("config", "Enabled the experimental AUDIO feature");
 		if ((config->tc358743_dev_path = _get_value(jcfg, "audio", "tc358743")) == NULL) {
@@ -85,3 +89,13 @@ static char *_get_value(janus_config *jcfg, const char *section, const char *opt
 	}
 	return strdup(option_obj->value);
 }
+
+static bool _get_bool(janus_config *jcfg, const char *section, const char *option, bool def) {
+	char *tmp = _get_value(jcfg, section, option);
+	bool value = def;
+	if (tmp != NULL) {
+		value = (!strcasecmp(tmp, "1") || !strcasecmp(tmp, "true") || !strcasecmp(tmp, "yes"));
+		free(tmp);
+	}
+	return value;
+}
diff --git a/janus/src/config.h b/janus/src/config.h
index 69668f8..9f9fe5c 100644
--- a/janus/src/config.h
+++ b/janus/src/config.h
@@ -36,6 +36,8 @@
 
 typedef struct {
 	char	*video_sink_name;
+	bool	video_zero_playout_delay;
+
 	char	*audio_dev_name;
 	char	*tc358743_dev_path;
 } plugin_config_s;
diff --git a/janus/src/plugin.c b/janus/src/plugin.c
index acfe6f7..da125f9 100644
--- a/janus/src/plugin.c
+++ b/janus/src/plugin.c
@@ -261,7 +261,7 @@ static int _plugin_init(janus_callbacks *gw, const char *config_dir_path) {
 	_g_gw = gw;
 
 	_g_video_queue = queue_init(1024);
-	_g_rtpv = rtpv_init(_relay_rtp_clients);
+	_g_rtpv = rtpv_init(_relay_rtp_clients, _g_config->video_zero_playout_delay);
 	if (_g_config->audio_dev_name) {
 		_g_rtpa = rtpa_init(_relay_rtp_clients);
 		A_THREAD_CREATE(&_g_audio_tid, _audio_thread, NULL);
diff --git a/janus/src/rtp.c b/janus/src/rtp.c
index 800d87d..cc11f60 100644
--- a/janus/src/rtp.c
+++ b/janus/src/rtp.c
@@ -26,11 +26,12 @@
 #include "rtp.h"
 
 
-rtp_s *rtp_init(unsigned payload, bool video) {
+rtp_s *rtp_init(unsigned payload, bool video, bool zero_playout_delay) {
 	rtp_s *rtp;
 	A_CALLOC(rtp, 1);
 	rtp->payload = payload;
 	rtp->video = video;
+	rtp->zero_playout_delay = zero_playout_delay; // See client.c
 	rtp->ssrc = triple_u32(get_now_monotonic_u64());
 	return rtp;
 }
diff --git a/janus/src/rtp.h b/janus/src/rtp.h
index 454224f..45a6d63 100644
--- a/janus/src/rtp.h
+++ b/janus/src/rtp.h
@@ -39,6 +39,7 @@
 typedef struct {
 	unsigned	payload;
 	bool		video;
+	bool		zero_playout_delay;
 	uint32_t	ssrc;
 
 	uint16_t	seq;
@@ -49,7 +50,7 @@ typedef struct {
 typedef void (*rtp_callback_f)(const rtp_s *rtp);
 
 
-rtp_s *rtp_init(unsigned payload, bool video);
+rtp_s *rtp_init(unsigned payload, bool video, bool zero_playout_delay);
 rtp_s *rtp_dup(const rtp_s *rtp);
 void rtp_destroy(rtp_s *rtp);
 
diff --git a/janus/src/rtpa.c b/janus/src/rtpa.c
index 46e9960..6d18354 100644
--- a/janus/src/rtpa.c
+++ b/janus/src/rtpa.c
@@ -26,7 +26,7 @@
 rtpa_s *rtpa_init(rtp_callback_f callback) {
 	rtpa_s *rtpa;
 	A_CALLOC(rtpa, 1);
-	rtpa->rtp = rtp_init(111, false);
+	rtpa->rtp = rtp_init(111, false, false);
 	rtpa->callback = callback;
 	return rtpa;
 }
diff --git a/janus/src/rtpv.c b/janus/src/rtpv.c
index 381eb12..ec86088 100644
--- a/janus/src/rtpv.c
+++ b/janus/src/rtpv.c
@@ -31,10 +31,10 @@ void _rtpv_process_nalu(rtpv_s *rtpv, const uint8_t *data, size_t size, uint32_t
 static ssize_t _find_annexb(const uint8_t *data, size_t size);
 
 
-rtpv_s *rtpv_init(rtp_callback_f callback) {
+rtpv_s *rtpv_init(rtp_callback_f callback, bool zero_playout_delay) {
 	rtpv_s *rtpv;
 	A_CALLOC(rtpv, 1);
-	rtpv->rtp = rtp_init(96, true);
+	rtpv->rtp = rtp_init(96, true, zero_playout_delay);
 	rtpv->callback = callback;
 	rtpv->sps = frame_init();
 	rtpv->pps = frame_init();
@@ -81,14 +81,14 @@ char *rtpv_make_sdp(rtpv_s *rtpv) {
 		"a=rtcp-fb:%u nack pli" RN
 		"a=rtcp-fb:%u goog-remb" RN
 		"a=ssrc:%" PRIu32 " cname:ustreamer" RN
-		// XXX: See client.c
-		// "a=extmap:1 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay" RN
+		"%s" // playout-delay
 		"a=sendonly" RN,
 		PAYLOAD, PAYLOAD, PAYLOAD, PAYLOAD,
 		PAYLOAD, sps,
 		PAYLOAD, pps,
 		PAYLOAD, PAYLOAD, PAYLOAD,
-		rtpv->rtp->ssrc
+		rtpv->rtp->ssrc,
+		(rtpv->rtp->zero_playout_delay ? "a=extmap:1 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay" RN : "")
 	);
 #	undef PAYLOAD
 
diff --git a/janus/src/rtpv.h b/janus/src/rtpv.h
index 7be712b..53fbea7 100644
--- a/janus/src/rtpv.h
+++ b/janus/src/rtpv.h
@@ -50,7 +50,7 @@ typedef struct {
 } rtpv_s;
 
 
-rtpv_s *rtpv_init(rtp_callback_f callback);
+rtpv_s *rtpv_init(rtp_callback_f callback, bool zero_playout_delay);
 void rtpv_destroy(rtpv_s *rtpv);
 
 char *rtpv_make_sdp(rtpv_s *rtpv);