sdp: Add support for setting connection address and clean up state.

This change cleans up state management for media streams by moving
RTP instances into their own session structure and adding additional
details that are not relevant to the core (such as connection address).
These can live either in the local capabilities or joint capabilities.

The ability to set explicit connection address information for
the purposes of direct media and NAT has also been added at the
global and stream specific level.

ASTERISK-26900

Change-Id: If7e5307239a9534420732de11c451a2705b6b681

1 files changed, 14 insertions, 79 deletions

diff --git a/main/sdp.c b/main/sdp.c
index 246763edf..1ef6400b9 100644
--- a/main/sdp.c
+++ b/main/sdp.c
@@ -500,69 +500,10 @@ int ast_sdp_m_add_format(struct ast_sdp_m_line *m_line, const struct ast_sdp_opt
 	return 0;
 }
 
-/* TODO
- * This isn't set anywhere yet.
- */
-/*! \brief Scheduler for RTCP purposes */
-static struct ast_sched_context *sched;
-
-/*! \brief Internal function which creates an RTP instance */
-static struct ast_rtp_instance *create_rtp(const struct ast_sdp_options *options,
-	enum ast_media_type media_type)
-{
-	struct ast_rtp_instance *rtp;
-	struct ast_rtp_engine_ice *ice;
-	struct ast_sockaddr temp_media_address;
-	static struct ast_sockaddr address_rtp;
-	struct ast_sockaddr *media_address =  &address_rtp;
-
-	if (options->bind_rtp_to_media_address && !ast_strlen_zero(options->media_address)) {
-		ast_sockaddr_parse(&temp_media_address, options->media_address, 0);
-		media_address = &temp_media_address;
-	} else {
-		if (ast_check_ipv6()) {
-			ast_sockaddr_parse(&address_rtp, "::", 0);
-		} else {
-			ast_sockaddr_parse(&address_rtp, "0.0.0.0", 0);
-		}
-	}
-
-	if (!(rtp = ast_rtp_instance_new(options->rtp_engine, sched, media_address, NULL))) {
-		ast_log(LOG_ERROR, "Unable to create RTP instance using RTP engine '%s'\n",
-			options->rtp_engine);
-		return NULL;
-	}
-
-	ast_rtp_instance_set_prop(rtp, AST_RTP_PROPERTY_RTCP, 1);
-	ast_rtp_instance_set_prop(rtp, AST_RTP_PROPERTY_NAT, options->rtp_symmetric);
-
-	if (options->ice == AST_SDP_ICE_DISABLED && (ice = ast_rtp_instance_get_ice(rtp))) {
-		ice->stop(rtp);
-	}
-
-	if (options->telephone_event) {
-		ast_rtp_instance_dtmf_mode_set(rtp, AST_RTP_DTMF_MODE_RFC2833);
-		ast_rtp_instance_set_prop(rtp, AST_RTP_PROPERTY_DTMF, 1);
-	}
-
-	if (media_type == AST_MEDIA_TYPE_AUDIO &&
-			(options->tos_audio || options->cos_audio)) {
-		ast_rtp_instance_set_qos(rtp, options->tos_audio,
-			options->cos_audio, "SIP RTP Audio");
-	} else if (media_type == AST_MEDIA_TYPE_VIDEO &&
-			(options->tos_video || options->cos_video)) {
-		ast_rtp_instance_set_qos(rtp, options->tos_video,
-			options->cos_video, "SIP RTP Video");
-	}
-
-	ast_rtp_instance_set_last_rx(rtp, time(NULL));
-
-	return rtp;
-}
-
-int ast_sdp_add_m_from_stream(struct ast_sdp *sdp, const struct ast_sdp_options *options,
-	struct ast_rtp_instance *rtp, const struct ast_stream *stream)
+int ast_sdp_add_m_from_rtp_stream(struct ast_sdp *sdp, const struct ast_sdp_state *sdp_state,
+	const struct ast_sdp_options *options, int stream_index)
 {
+	struct ast_stream *stream = ast_stream_topology_get_stream(ast_sdp_state_get_local_topology(sdp_state), stream_index);
 	struct ast_sdp_m_line *m_line;
 	struct ast_format_cap *caps;
 	int i;
@@ -572,13 +513,15 @@ int ast_sdp_add_m_from_stream(struct ast_sdp *sdp, const struct ast_sdp_options
 	enum ast_media_type media_type;
 	char tmp[64];
 	struct ast_sockaddr address_rtp;
+	struct ast_rtp_instance *rtp = ast_sdp_state_get_rtp_instance(sdp_state, stream_index);
 	struct ast_sdp_a_line *a_line;
 
-
-	ast_assert(sdp && options && rtp && stream);
+	ast_assert(sdp && options && stream);
 
 	media_type = ast_stream_get_type(stream);
-	ast_rtp_instance_get_local_address(rtp, &address_rtp);
+	if (ast_sdp_state_get_stream_connection_address(sdp_state, 0, &address_rtp)) {
+		return -1;
+	}
 
 	m_line = ast_sdp_m_alloc(
 		ast_codec_media_type2str(ast_stream_get_type(stream)),
@@ -673,7 +616,7 @@ int ast_sdp_add_m_from_stream(struct ast_sdp *sdp, const struct ast_sdp_options
 		}
 	}
 
-	a_line = ast_sdp_a_alloc(options->locally_held ? "sendonly" : "sendrecv", "");
+	a_line = ast_sdp_a_alloc(ast_sdp_state_get_locally_held(sdp_state, stream_index) ? "sendonly" : "sendrecv", "");
 	if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
 		ast_sdp_a_free(a_line);
 		ast_sdp_m_free(m_line);
@@ -699,7 +642,6 @@ struct ast_sdp *ast_sdp_create_from_state(const struct ast_sdp_state *sdp_state)
 	struct ast_sdp_c_line *c_line = NULL;
 	struct ast_sdp_s_line *s_line = NULL;
 	struct ast_sdp_t_line *t_line = NULL;
-	struct ast_rtp_instance *rtp = NULL;
 	char *address_type;
 	struct timeval tv = ast_tvnow();
 	uint32_t t;
@@ -732,25 +674,18 @@ struct ast_sdp *ast_sdp_create_from_state(const struct ast_sdp_state *sdp_state)
 	}
 
 	for (stream_num = 0; stream_num < stream_count; stream_num++) {
-		struct ast_stream *stream = ast_stream_topology_get_stream(topology, stream_num);
-
-		rtp = create_rtp(options, ast_stream_get_type(stream));
-		if (!rtp) {
-			goto error;
-		}
-
-		ast_stream_set_data(stream, AST_STREAM_DATA_RTP_INSTANCE,
-			rtp, (ast_stream_data_free_fn)&ast_rtp_instance_destroy);
+		enum ast_media_type type = ast_stream_get_type(ast_stream_topology_get_stream(topology, stream_num));
 
-		if (ast_sdp_add_m_from_stream(sdp, options, rtp, stream)) {
-			goto error;
+		if (type == AST_MEDIA_TYPE_AUDIO || type == AST_MEDIA_TYPE_VIDEO) {
+			if (ast_sdp_add_m_from_rtp_stream(sdp, sdp_state, options, stream_num)) {
+				goto error;
+			}
 		}
 	}
 
 	return sdp;
 
 error:
-	ao2_cleanup(rtp);
 	if (sdp) {
 		ast_sdp_free(sdp);
 	} else {