diff --git a/janus/src/client.c b/janus/src/client.c
index 6ce9147..28be3b1 100644
--- a/janus/src/client.c
+++ b/janus/src/client.c
@@ -108,12 +108,13 @@ static void *_common_thread(void *v_client, bool video) {
 				packet.mindex = (rtp->video ? 0 : 1);
 #				endif
 				janus_plugin_rtp_extensions_reset(&packet.extensions);
-				// 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 (rtp->zero_playout_delay) {
+					// https://github.com/pikvm/pikvm/issues/784
 					packet.extensions.min_delay = 0;
 					packet.extensions.max_delay = 0;
+				} else {
+					packet.extensions.min_delay = 0;
+					packet.extensions.max_delay = 1000;
 				}
 				client->gw->relay_rtp(client->session, &packet);
 			}
diff --git a/janus/src/config.c b/janus/src/config.c
index 9806818..2c7e9d1 100644
--- a/janus/src/config.c
+++ b/janus/src/config.c
@@ -24,7 +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);
+// static bool _get_bool(janus_config *jcfg, const char *section, const char *option, bool def);
 
 
 us_config_s *us_config_init(const char *config_dir_path) {
@@ -51,9 +51,6 @@ us_config_s *us_config_init(const char *config_dir_path) {
 		US_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) {
-		US_JLOG_INFO("config", "Enabled the experimental Playout-Delay=0 RTP extension for VIDEO");
-	}
 	if ((config->audio_dev_name = _get_value(jcfg, "audio", "device")) != NULL) {
 		US_JLOG_INFO("config", "Enabled the experimental AUDIO feature");
 		if ((config->tc358743_dev_path = _get_value(jcfg, "audio", "tc358743")) == NULL) {
@@ -88,7 +85,7 @@ static char *_get_value(janus_config *jcfg, const char *section, const char *opt
 	return us_strdup(option_obj->value);
 }
 
-static bool _get_bool(janus_config *jcfg, const char *section, const char *option, bool def) {
+/*static bool _get_bool(janus_config *jcfg, const char *section, const char *option, bool def) {
 	char *const tmp = _get_value(jcfg, section, option);
 	bool value = def;
 	if (tmp != NULL) {
@@ -96,4 +93,4 @@ static bool _get_bool(janus_config *jcfg, const char *section, const char *optio
 		free(tmp);
 	}
 	return value;
-}
+}*/
diff --git a/janus/src/config.h b/janus/src/config.h
index 52de3d1..4bdb852 100644
--- a/janus/src/config.h
+++ b/janus/src/config.h
@@ -36,7 +36,6 @@
 
 typedef struct {
 	char	*video_sink_name;
-	bool	video_zero_playout_delay;
 
 	char	*audio_dev_name;
 	char	*tc358743_dev_path;
diff --git a/janus/src/plugin.c b/janus/src/plugin.c
index 8537f8e..ee34c94 100644
--- a/janus/src/plugin.c
+++ b/janus/src/plugin.c
@@ -262,7 +262,7 @@ static int _plugin_init(janus_callbacks *gw, const char *config_dir_path) {
 	_g_gw = gw;
 
 	_g_video_queue = us_queue_init(1024);
-	_g_rtpv = us_rtpv_init(_relay_rtp_clients, _g_config->video_zero_playout_delay);
+	_g_rtpv = us_rtpv_init(_relay_rtp_clients);
 	if (_g_config->audio_dev_name != NULL) {
 		_g_rtpa = us_rtpa_init(_relay_rtp_clients);
 		US_THREAD_CREATE(_g_audio_tid, _audio_thread, NULL);
diff --git a/janus/src/rtp.c b/janus/src/rtp.c
index b7f0ea9..1c31b84 100644
--- a/janus/src/rtp.c
+++ b/janus/src/rtp.c
@@ -26,12 +26,11 @@
 #include "rtp.h"
 
 
-us_rtp_s *us_rtp_init(unsigned payload, bool video, bool zero_playout_delay) {
+us_rtp_s *us_rtp_init(unsigned payload, bool video) {
 	us_rtp_s *rtp;
 	US_CALLOC(rtp, 1);
 	rtp->payload = payload;
 	rtp->video = video;
-	rtp->zero_playout_delay = zero_playout_delay; // See client.c
 	rtp->ssrc = us_triple_u32(us_get_now_monotonic_u64());
 	return rtp;
 }
diff --git a/janus/src/rtp.h b/janus/src/rtp.h
index 372bbbe..06cc1d7 100644
--- a/janus/src/rtp.h
+++ b/janus/src/rtp.h
@@ -39,18 +39,18 @@
 typedef struct {
 	unsigned	payload;
 	bool		video;
-	bool		zero_playout_delay;
 	uint32_t	ssrc;
 
 	uint16_t	seq;
 	uint8_t		datagram[US_RTP_DATAGRAM_SIZE];
 	size_t		used;
+	bool		zero_playout_delay;
 } us_rtp_s;
 
 typedef void (*us_rtp_callback_f)(const us_rtp_s *rtp);
 
 
-us_rtp_s *us_rtp_init(unsigned payload, bool video, bool zero_playout_delay);
+us_rtp_s *us_rtp_init(unsigned payload, bool video);
 us_rtp_s *us_rtp_dup(const us_rtp_s *rtp);
 void us_rtp_destroy(us_rtp_s *rtp);
 
diff --git a/janus/src/rtpa.c b/janus/src/rtpa.c
index 8c2305d..c3be0b0 100644
--- a/janus/src/rtpa.c
+++ b/janus/src/rtpa.c
@@ -26,7 +26,7 @@
 us_rtpa_s *us_rtpa_init(us_rtp_callback_f callback) {
 	us_rtpa_s *rtpa;
 	US_CALLOC(rtpa, 1);
-	rtpa->rtp = us_rtp_init(111, false, false);
+	rtpa->rtp = us_rtp_init(111, false);
 	rtpa->callback = callback;
 	return rtpa;
 }
diff --git a/janus/src/rtpv.c b/janus/src/rtpv.c
index e81b93f..f21c9b3 100644
--- a/janus/src/rtpv.c
+++ b/janus/src/rtpv.c
@@ -31,10 +31,10 @@ void _rtpv_process_nalu(us_rtpv_s *rtpv, const uint8_t *data, size_t size, uint3
 static ssize_t _find_annexb(const uint8_t *data, size_t size);
 
 
-us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback, bool zero_playout_delay) {
+us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback) {
 	us_rtpv_s *rtpv;
 	US_CALLOC(rtpv, 1);
-	rtpv->rtp = us_rtp_init(96, true, zero_playout_delay);
+	rtpv->rtp = us_rtp_init(96, true);
 	rtpv->callback = callback;
 	return rtpv;
 }
@@ -59,12 +59,11 @@ char *us_rtpv_make_sdp(us_rtpv_s *rtpv) {
 		"a=rtcp-fb:%u nack pli" RN
 		"a=rtcp-fb:%u goog-remb" RN
 		"a=ssrc:%" PRIu32 " cname:ustreamer" RN
-		"%s" // playout-delay
+		"a=extmap:1 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay" RN
 		"a=sendonly" RN,
 		PAYLOAD, PAYLOAD, PAYLOAD, PAYLOAD,
 		PAYLOAD, PAYLOAD, PAYLOAD,
-		rtpv->rtp->ssrc,
-		(rtpv->rtp->zero_playout_delay ? "a=extmap:1 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay" RN : "")
+		rtpv->rtp->ssrc
 	);
 	return sdp;
 #	undef PAYLOAD
@@ -78,6 +77,8 @@ void us_rtpv_wrap(us_rtpv_s *rtpv, const us_frame_s *frame) {
 
 	assert(frame->format == V4L2_PIX_FMT_H264);
 
+	rtpv->rtp->zero_playout_delay = (frame->gop == 0);
+
 	const uint32_t pts = us_get_now_monotonic_u64() * 9 / 100; // PTS units are in 90 kHz
 	ssize_t last_offset = -_PRE;
 
diff --git a/janus/src/rtpv.h b/janus/src/rtpv.h
index e24736d..01c2010 100644
--- a/janus/src/rtpv.h
+++ b/janus/src/rtpv.h
@@ -43,7 +43,7 @@ typedef struct {
 } us_rtpv_s;
 
 
-us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback, bool zero_playout_delay);
+us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback);
 void us_rtpv_destroy(us_rtpv_s *rtpv);
 
 char *us_rtpv_make_sdp(us_rtpv_s *rtpv);