janus: ssrc per client

janus/src/client.c

@@ -55,6 +55,9 @@ us_janus_client_s *us_janus_client_init(janus_callbacks *gw, janus_plugin_sessio
 	US_CALLOC(client, 1);
 	client->gw = gw;
 	client->session = session;
+	client->video_ssrc = us_get_ssrc();
+	client->audio_ssrc = us_get_ssrc();
+
 	atomic_init(&client->transmit, false);
 	atomic_init(&client->transmit_acap, false);
 	atomic_init(&client->transmit_aplay, false);
@@ -181,6 +184,8 @@ static void *_video_or_acap_thread(void *v_client, bool video) {
 			atomic_load(&client->transmit)
 			&& (video || atomic_load(&client->transmit_acap))
 		) {
+			us_rtp_write_ssrc(&rtp, (rtp.video ? client->video_ssrc : client->audio_ssrc));
+
 			janus_plugin_rtp packet = {
 				.video = rtp.video,
 				.buffer = (char*)rtp.datagram,

janus/src/client.h

@@ -37,6 +37,9 @@
 typedef struct {
 	janus_callbacks			*gw;
 	janus_plugin_session	*session;
+	u32						video_ssrc;
+	u32						audio_ssrc;
+
 	atomic_bool				transmit;
 	atomic_bool				transmit_acap;
 	atomic_bool				transmit_aplay;

janus/src/plugin.c

@@ -641,14 +641,22 @@ static struct janus_plugin_result *_plugin_handle_message(
 		}
 
 		{
-			char *const sdp = us_sdp_create(
+			_LOCK_ALL;
-				_g_rtpv,
+			US_LIST_ITERATE(_g_clients, client, {
-				(with_acap ? _g_rtpa : NULL),
+				if (client->session == session) {
-				with_aplay);
+					char *const sdp = us_sdp_create(
-			json_t *const offer_jsep = json_pack("{ssss}", "type", "offer", "sdp", sdp);
+						client->video_ssrc,
-			PUSH_STATUS("started", NULL, offer_jsep);
+						client->audio_ssrc,
-			json_decref(offer_jsep);
+						with_acap,
-			free(sdp);
+						with_aplay);
+					json_t *const offer_jsep = json_pack("{ssss}", "type", "offer", "sdp", sdp);
+					PUSH_STATUS("started", NULL, offer_jsep);
+					json_decref(offer_jsep);
+					free(sdp);
+					break;
+				}
+			});
+			_UNLOCK_ALL;
 		}
 
 		{

janus/src/rtp.c

@@ -44,9 +44,11 @@ void us_rtp_destroy(us_rtp_s *rtp) {
 void us_rtp_assign(us_rtp_s *rtp, uint payload, bool video) {
 	rtp->payload = payload;
 	rtp->video = video;
-	rtp->ssrc = us_triple_u32(us_get_now_monotonic_u64());
 }
 
+#define WRITE_BE_U32(x_offset, x_value) \
+	*((u32*)(rtp->datagram + x_offset)) = __builtin_bswap32(x_value)
+
 void us_rtp_write_header(us_rtp_s *rtp, u32 pts, bool last_header) {
 	u32 word0 = 0x80000000;
 	if (last_header) {
@@ -56,10 +58,13 @@ void us_rtp_write_header(us_rtp_s *rtp, u32 pts, bool last_header) {
 	word0 |= rtp->seq;
 	++rtp->seq;
 
-#	define WRITE_BE_U32(x_offset, x_value) \
-		*((u32*)(rtp->datagram + x_offset)) = __builtin_bswap32(x_value)
 	WRITE_BE_U32(0, word0);
 	WRITE_BE_U32(4, pts);
-	WRITE_BE_U32(8, rtp->ssrc);
+	WRITE_BE_U32(8, 0); // SSRC should be written separately for each client
-#	undef WRITE_BE_U32
 }
+
+void us_rtp_write_ssrc(us_rtp_s *rtp, u32 ssrc) {
+	WRITE_BE_U32(8, ssrc);
+}
+
+#undef WRITE_BE_U32

janus/src/rtp.h

@@ -23,6 +23,7 @@
 #pragma once
 
 #include "uslibs/types.h"
+#include "uslibs/tools.h"
 
 
 // Max RTP size for WebRTC is 1200 bytes:
@@ -47,7 +48,6 @@
 typedef struct {
 	uint	payload;
 	bool	video;
-	u32		ssrc;
 
 	u16		seq;
 	u8		datagram[US_RTP_TOTAL_SIZE];
@@ -62,8 +62,13 @@ typedef struct {
 typedef void (*us_rtp_callback_f)(const us_rtp_s *rtp);
 
 
+static inline u32 us_get_ssrc(void) {
+	return us_triple_u32(us_get_now_monotonic_u64());
+}
+
 us_rtp_s *us_rtp_init(void);
 void us_rtp_destroy(us_rtp_s *rtp);
 
 void us_rtp_assign(us_rtp_s *rtp, uint payload, bool video);
 void us_rtp_write_header(us_rtp_s *rtp, u32 pts, bool last_header);
+void us_rtp_write_ssrc(us_rtp_s *rtp, u32 ssrc);

janus/src/sdp.c

@@ -34,12 +34,12 @@
 #include "rtpa.h"
 
 
-char *us_sdp_create(us_rtpv_s *rtpv, us_rtpa_s *rtpa, bool mic) {
+char *us_sdp_create(u32 video_ssrc, u32 audio_ssrc, bool acap, bool aplay) {
 	char *video_sdp;
 	{
 		// https://tools.ietf.org/html/rfc6184
 		// https://github.com/meetecho/janus-gateway/issues/2443
-		const uint pl = rtpv->rtp->payload;
+		const uint pl = US_RTP_H264_PAYLOAD;
 		US_ASPRINTF(
 			video_sdp,
 			"m=video 1 RTP/SAVPF %u" RN
@@ -58,31 +58,20 @@ char *us_sdp_create(us_rtpv_s *rtpv, us_rtpa_s *rtpa, bool mic) {
 			"a=extmap:3/sendonly http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time" RN
 			"a=sendonly" RN,
 			pl, pl, pl, pl, pl, pl, pl,
-			rtpv->rtp->ssrc);
+			video_ssrc);
 	}
 
 	char *audio_sdp;
-	if (rtpa == NULL) {
+	if (acap || aplay) {
-		audio_sdp = us_strdup("");
+		const uint pl = US_RTP_OPUS_PAYLOAD;
-	} else {
 
-
+		char *dir = "";
-	if (rtpa != NULL || mic) {
+		if (acap && aplay) {
-		uint pl;
+			dir = "sendrecv";
-		u32 ssrc;
+		} else if (acap && !aplay) {
-		if (rtpa != NULL) {
-			pl = rtpa->rtp->payload;
-			ssrc = rtpa->rtp->ssrc;
-		} else {
-			pl = US_RTP_OPUS_PAYLOAD;
-			ssrc = us_triple_u32(us_get_now_monotonic_u64());
-		}
-
-		char *dir = "sendrecv";
-		if (rtpa == NULL) {
-			dir = "recvonly";
-		} else if (!mic) {
 			dir = "sendonly";
+		} else if (!acap && aplay) {
+			dir = "recvonly";
 		}
 
 		US_ASPRINTF(
@@ -98,8 +87,10 @@ char *us_sdp_create(us_rtpv_s *rtpv, us_rtpa_s *rtpa, bool mic) {
 			pl, pl,
 			US_RTP_OPUS_HZ, US_RTP_OPUS_CH,
 			pl,
-			ssrc,
+			audio_ssrc,
 			dir);
+	} else {
+		audio_sdp = us_strdup("");
 	}
 
 	char *sdp;

janus/src/sdp.h

@@ -24,8 +24,5 @@
 
 #include "uslibs/types.h"
 
-#include "rtpv.h"
-#include "rtpa.h"
 
-
+char *us_sdp_create(u32 video_ssrc, u32 audio_ssrc, bool acap, bool aplay);
-char *us_sdp_create(us_rtpv_s *rtpv, us_rtpa_s *rtpa, bool mic);