MT#60386 sdp_create: don't carry stream_params

Don't carry parameters required for processing
in the `sdp_create()` via the `stream_params`,
but rahter handle them like:
- parsing in `sdp_parse()`
- `sdp_media` -> `stream_params` in sdp_streams()
- `stream_params` -> `call_media` in ` __media_init_from_flags()`

Additionally: update the test "subscribe_request AMR asymmetric".
This is because we seem to never actually take into account
presence of bandwidth data in offer/answer model preceding
the subscribe request.

Change-Id: I5b4b19ae244c6bbf961d5ea7c18b6747519144db

daemon/call.c

@@ -2879,6 +2879,26 @@ static void __media_init_from_flags(struct call_media *other_media, struct call_
 		if (sp->desired_family)
 			media->desired_family = sp->desired_family;
 	}
+
+	/* desired endpoint's address */
+	other_media->desired_address = NULL;
+	if (media)
+		media->desired_address = NULL;
+	if (sp->rtp_endpoint.address.family) {
+		other_media->desired_address = &sp->rtp_endpoint.address;
+		if (media)
+			media->desired_address = &sp->rtp_endpoint.address;
+	}
+
+	/* desired bandwidth */
+	other_media->desired_bandwidth_as = sp->media_session_as;
+	other_media->desired_bandwidth_rr = sp->media_session_rr;
+	other_media->desired_bandwidth_rs = sp->media_session_rs;
+	if (media) {
+		media->desired_bandwidth_as = sp->media_session_as;
+		media->desired_bandwidth_rr = sp->media_session_rr;
+		media->desired_bandwidth_rs = sp->media_session_rs;
+	}
 }
 
 unsigned int proto_num_ports(unsigned int sp_ports, struct call_media *media, sdp_ng_flags *flags,

daemon/call_interfaces.c

@@ -3690,7 +3690,7 @@ const char *call_publish_ng(ng_buffer *ngbuf, bencode_item_t *input, bencode_ite
 	if (ret)
 		ilog(LOG_ERR, "Publish error"); // XXX close call? handle errors?
 
-	ret = sdp_create(&sdp_out, ml, &streams, &flags);
+	ret = sdp_create(&sdp_out, ml, &flags);
 	if (!ret) {
 		save_last_sdp(ml, &sdp_in, &parsed, &streams);
 		bencode_buffer_destroy_add(output->buffer, g_free, sdp_out.s);
@@ -3771,7 +3771,7 @@ const char *call_subscribe_request_ng(bencode_item_t *input, bencode_item_t *out
 			return "Failed to rewrite SDP";
 	} else {
 		/* create new SDP */
-		ret = sdp_create(&sdp_out, dest_ml, NULL, &flags);
+		ret = sdp_create(&sdp_out, dest_ml, &flags);
 	}
 
 	/* place return output SDP */

daemon/janus.c

@@ -644,7 +644,7 @@ static const char *janus_videoroom_join(struct websocket_message *wm, struct jan
 			return "Subscribe error";
 
 		/* create SDP */
-		ret = sdp_create(jsep_sdp_out, dest_ml, NULL, &flags);
+		ret = sdp_create(jsep_sdp_out, dest_ml, &flags);
 
 		if (!dest_ml->janus_session)
 			dest_ml->janus_session = obj_get(session);
@@ -874,7 +874,7 @@ static const char *janus_videoroom_configure(struct websocket_message *wm, struc
 		// XXX check there's only one audio and one video stream?
 
 		g_auto(str) sdp_out = STR_NULL;
-		ret = sdp_create(&sdp_out, ml, &streams, &flags);
+		ret = sdp_create(&sdp_out, ml, &flags);
 		if (ret)
 			return "Publish error";
 

daemon/sdp.c

@@ -1837,6 +1837,9 @@ int sdp_streams(const sdp_sessions_q *sessions, sdp_streams_q *streams, sdp_ng_f
 				flags->trust_address = 1;
 			}
 
+			/*
+			 * pass important context parameters: sdp_media -> stream_params
+			 */
 			sp->consecutive_ports = media->port_count;
 			sp->num_ports = sp->consecutive_ports * 2; // only do *=2 for RTP streams?
 			sp->protocol_str = media->transport;
@@ -1850,7 +1853,7 @@ int sdp_streams(const sdp_sessions_q *sessions, sdp_streams_q *streams, sdp_ng_f
 			bf_set_clear(&sp->sp_flags, SP_FLAG_STRICT_SOURCE, flags->strict_source);
 			bf_set_clear(&sp->sp_flags, SP_FLAG_MEDIA_HANDOVER, flags->media_handover);
 
-			/* b= (bandwidth) */
+			/* b= (bandwidth), is parsed in sdp_parse() */
 			sp->media_session_as = media->as;
 			sp->media_session_rr = media->rr;
 			sp->media_session_rs = media->rs;
@@ -3472,7 +3475,7 @@ static void sdp_out_add_timing(GString *out, struct call_monologue *monologue)
 }
 
 static void sdp_out_add_bandwidth(GString *out, struct call_monologue *monologue,
-		struct stream_params *sp, struct call_media *media)
+		struct call_media *media)
 {
 	struct call_monologue *ml = monologue;
 	struct media_subscription *ms = call_get_top_media_subscription(monologue);
@@ -3481,12 +3484,12 @@ static void sdp_out_add_bandwidth(GString *out, struct call_monologue *monologue
 
 	/* sdp bandwidth per media level */
 	if (media) {
-		if (sp && sp->media_session_as >= 0)
-			g_string_append_printf(out, "b=AS:%d\r\n", sp->media_session_as);
-		if (sp && sp->media_session_rr >= 0)
-			g_string_append_printf(out, "b=RR:%d\r\n", sp->media_session_rr);
-		if (sp && sp->media_session_rs >= 0)
-			g_string_append_printf(out, "b=RS:%d\r\n", sp->media_session_rs);
+		if (media->desired_bandwidth_as >= 0)
+			g_string_append_printf(out, "b=AS:%d\r\n", media->desired_bandwidth_as);
+		if (media->desired_bandwidth_rr >= 0)
+			g_string_append_printf(out, "b=RR:%d\r\n", media->desired_bandwidth_rr);
+		if (media->desired_bandwidth_rs >= 0)
+			g_string_append_printf(out, "b=RS:%d\r\n", media->desired_bandwidth_rs);
 	}
 	else {
 		/* sdp bandwidth per session/media level
@@ -3499,17 +3502,16 @@ static void sdp_out_add_bandwidth(GString *out, struct call_monologue *monologue
 }
 
 static void sdp_out_add_media_connection(GString *out, struct call_media *media,
-		struct packet_stream *rtp_ps, struct stream_params *sp,
-		sdp_ng_flags *flags)
+		struct packet_stream *rtp_ps, sdp_ng_flags *flags)
 {
 	const char *media_conn_address = NULL;
 	const char *media_conn_address_type = rtp_ps->selected_sfd->local_intf->advertised_address.addr.family->rfc_name;
 
 	/* we want to keep an original media connection for message / force relay */
-	if (sp && (media->type_id == MT_MESSAGE || flags->ice_option == ICE_FORCE_RELAY))
+	if (media->desired_address && (media->type_id == MT_MESSAGE || flags->ice_option == ICE_FORCE_RELAY))
 	{
-		media_conn_address = sockaddr_print_buf(&sp->rtp_endpoint.address);
-		media_conn_address_type = sp->rtp_endpoint.address.family->rfc_name;
+		media_conn_address = sockaddr_print_buf(media->desired_address);
+		media_conn_address_type = media->desired_family->rfc_name;
 	}
 	else {
 		media_conn_address = sockaddr_print_buf(&rtp_ps->selected_sfd->local_intf->advertised_address.addr);
@@ -3529,12 +3531,10 @@ static void sdp_out_add_media_connection(GString *out, struct call_media *media,
  * For the rest of cases (publish, subscribe, janus etc.) this works as usual:
  * given monologue is a monologue which is being processed.
  */
-int sdp_create(str *out, struct call_monologue *monologue,
-	 sdp_streams_q *streams, sdp_ng_flags *flags)
+int sdp_create(str *out, struct call_monologue *monologue, sdp_ng_flags *flags)
 {
 	const char *err = NULL;
 	GString *s = NULL;
-	struct stream_params *sp;
 
 	err = "Need at least one media";
 	if (!monologue->medias->len)
@@ -3563,7 +3563,7 @@ int sdp_create(str *out, struct call_monologue *monologue,
 	 * but instead per media, below */
 
 	/* add bandwidth control per session level */
-	sdp_out_add_bandwidth(s, monologue, NULL, NULL);
+	sdp_out_add_bandwidth(s, monologue, NULL);
 
 	/* set timing to always be: 0 0 */
 	sdp_out_add_timing(s, monologue);
@@ -3575,11 +3575,6 @@ int sdp_create(str *out, struct call_monologue *monologue,
 	{
 		media = monologue->medias->pdata[i];
 
-		/* take corresponding stream to get per media flags */
-		sp = NULL;
-		if (streams)
-			sp = t_queue_peek_nth(streams, i);
-
 		/* check call media existence */
 		err = "Empty media stream";
 		if (!media)
@@ -3618,10 +3613,10 @@ int sdp_create(str *out, struct call_monologue *monologue,
 		g_string_append_printf(s, "\r\n");
 
 		/* add actual media connection */
-		sdp_out_add_media_connection(s, media, rtp_ps, sp, flags);
+		sdp_out_add_media_connection(s, media, rtp_ps, flags);
 
 		/* add per media bandwidth */
-		sdp_out_add_bandwidth(s, monologue, sp, media);
+		sdp_out_add_bandwidth(s, monologue, media);
 
 		/* print media level attributes */
 		print_sdp_media_section(s, media, NULL, flags, rtp_ps_link, true, false);

include/call.h

@@ -505,6 +505,7 @@ struct call_media {
 	const struct transport_protocol *protocol;
 	str			format_str;
 	sockfamily_t		*desired_family;
+	sockaddr_t			*desired_address;
 	const struct logical_intf *logical_intf;
 
 	struct ice_agent	*ice_agent;
@@ -550,6 +551,9 @@ struct call_media {
 	dtmf_event_q		dtmf_send;
 	int					media_sdp_id;
 
+	/* bandwidth */
+	int desired_bandwidth_as, desired_bandwidth_rr, desired_bandwidth_rs;
+
 #ifdef WITH_TRANSCODING
 	encoder_callback_t	encoder_callback;
 #endif

include/sdp.h

@@ -81,7 +81,7 @@ void sdp_streams_clear(sdp_streams_q *);
 void sdp_sessions_clear(sdp_sessions_q *sessions);
 int sdp_replace(struct sdp_chopper *, sdp_sessions_q *, struct call_monologue *, sdp_ng_flags *);
 int sdp_is_duplicate(sdp_sessions_q *sessions);
-int sdp_create(str *out, struct call_monologue *, sdp_streams_q *streams, sdp_ng_flags *flags);
+int sdp_create(str *out, struct call_monologue *, sdp_ng_flags *flags);
 const char *sdp_get_sendrecv(struct call_media *media);
 
 int sdp_parse_candidate(struct ice_candidate *cand, const str *s); // returns -1, 0, 1

t/auto-daemon-tests.pl

@@ -1559,6 +1559,8 @@ s=-
 t=0 0
 m=audio PORT RTP/AVP 96 98
 c=IN IP4 203.0.113.1
+b=RR:1087
+b=RS:362
 a=label:0
 a=rtpmap:96 AMR/8000
 a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0
@@ -1569,6 +1571,9 @@ a=rtcp:PORT
 a=ptime:20
 m=audio PORT RTP/AVP 118 98
 c=IN IP4 203.0.113.1
+b=AS:80
+b=RR:1087
+b=RS:362
 a=label:1
 a=rtpmap:118 AMR/8000
 a=fmtp:118 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0