SDP: Replace SDP telephone_event option with dtmf option

The telephone_event option was used as a flag and a bit mapped value in
different places when it is a boolean.  It is also inadequate to configure
the DTMF operation of the RTP instance created for the stream.

Change-Id: Ib1addeaf0ce86f07039f2f979cab29405dc5239b

3 files changed, 29 insertions, 35 deletions

diff --git a/main/sdp_options.c b/main/sdp_options.c
index 3f25e4326..9b57e189f 100644
--- a/main/sdp_options.c
+++ b/main/sdp_options.c
@@ -23,8 +23,8 @@
 
 #include "sdp_private.h"
 
+#define DEFAULT_DTMF AST_SDP_DTMF_NONE
 #define DEFAULT_ICE AST_SDP_ICE_DISABLED
-#define DEFAULT_TELEPHONE_EVENT 0
 #define DEFAULT_IMPL AST_SDP_IMPL_STRING
 #define DEFAULT_ENCRYPTION AST_SDP_ENCRYPTION_DISABLED
 
@@ -65,7 +65,6 @@ DEFINE_GETTERS_SETTERS_FOR(unsigned int, rtp_symmetric);
 DEFINE_GETTERS_SETTERS_FOR(unsigned int, udptl_symmetric);
 DEFINE_GETTERS_SETTERS_FOR(enum ast_t38_ec_modes, udptl_error_correction);
 DEFINE_GETTERS_SETTERS_FOR(unsigned int, udptl_far_max_datagram);
-DEFINE_GETTERS_SETTERS_FOR(unsigned int, telephone_event);
 DEFINE_GETTERS_SETTERS_FOR(unsigned int, rtp_ipv6);
 DEFINE_GETTERS_SETTERS_FOR(unsigned int, g726_non_standard);
 DEFINE_GETTERS_SETTERS_FOR(unsigned int, rtcp_mux);
@@ -73,6 +72,7 @@ DEFINE_GETTERS_SETTERS_FOR(unsigned int, tos_audio);
 DEFINE_GETTERS_SETTERS_FOR(unsigned int, cos_audio);
 DEFINE_GETTERS_SETTERS_FOR(unsigned int, tos_video);
 DEFINE_GETTERS_SETTERS_FOR(unsigned int, cos_video);
+DEFINE_GETTERS_SETTERS_FOR(enum ast_sdp_options_dtmf, dtmf);
 DEFINE_GETTERS_SETTERS_FOR(enum ast_sdp_options_ice, ice);
 DEFINE_GETTERS_SETTERS_FOR(enum ast_sdp_options_impl, impl);
 DEFINE_GETTERS_SETTERS_FOR(enum ast_sdp_options_encryption, encryption);
@@ -80,8 +80,8 @@ DEFINE_GETTERS_SETTERS_FOR(unsigned int, ssrc);
 
 static void set_defaults(struct ast_sdp_options *options)
 {
+	options->dtmf = DEFAULT_DTMF;
 	options->ice = DEFAULT_ICE;
-	options->telephone_event = DEFAULT_TELEPHONE_EVENT;
 	options->impl = DEFAULT_IMPL;
 	options->encryption = DEFAULT_ENCRYPTION;
 }
diff --git a/main/sdp_private.h b/main/sdp_private.h
index f80cefb5f..c90a57485 100644
--- a/main/sdp_private.h
+++ b/main/sdp_private.h
@@ -34,16 +34,15 @@ struct ast_sdp_options {
 		AST_STRING_FIELD(rtp_engine);
 	);
 	struct {
-		unsigned int bind_rtp_to_media_address : 1;
-		unsigned int bind_udptl_to_media_address : 1;
-		unsigned int rtp_symmetric : 1;
-		unsigned int udptl_symmetric : 1;
-		unsigned int telephone_event : 1;
-		unsigned int rtp_ipv6 : 1;
-		unsigned int g726_non_standard : 1;
-		unsigned int locally_held : 1;
-		unsigned int rtcp_mux: 1;
-		unsigned int ssrc: 1;
+		unsigned int bind_rtp_to_media_address:1;
+		unsigned int bind_udptl_to_media_address:1;
+		unsigned int rtp_symmetric:1;
+		unsigned int udptl_symmetric:1;
+		unsigned int rtp_ipv6:1;
+		unsigned int g726_non_standard:1;
+		unsigned int locally_held:1;
+		unsigned int rtcp_mux:1;
+		unsigned int ssrc:1;
 	};
 	struct {
 		unsigned int tos_audio;
@@ -52,6 +51,7 @@ struct ast_sdp_options {
 		unsigned int cos_video;
 		unsigned int udptl_far_max_datagram;
 	};
+	enum ast_sdp_options_dtmf dtmf;
 	enum ast_sdp_options_ice ice;
 	enum ast_sdp_options_impl impl;
 	enum ast_sdp_options_encryption encryption;
diff --git a/main/sdp_state.c b/main/sdp_state.c
index 953f90c36..3a87a81e2 100644
--- a/main/sdp_state.c
+++ b/main/sdp_state.c
@@ -178,9 +178,11 @@ static struct ast_rtp_instance *create_rtp(const struct ast_sdp_options *options
 		ice->stop(rtp);
 	}
 
-	if (options->telephone_event) {
+	if (options->dtmf == AST_SDP_DTMF_RFC_4733 || options->dtmf == AST_SDP_DTMF_AUTO) {
 		ast_rtp_instance_dtmf_mode_set(rtp, AST_RTP_DTMF_MODE_RFC2833);
 		ast_rtp_instance_set_prop(rtp, AST_RTP_PROPERTY_DTMF, 1);
+	} else if (options->dtmf == AST_SDP_DTMF_INBAND) {
+		ast_rtp_instance_dtmf_mode_set(rtp, AST_RTP_DTMF_MODE_INBAND);
 	}
 
 	if (media_type == AST_MEDIA_TYPE_AUDIO &&
@@ -1257,7 +1259,6 @@ static int sdp_add_m_from_rtp_stream(struct ast_sdp *sdp, const struct ast_sdp_s
 
 	ast_assert(sdp && options && stream);
 
-	media_type = ast_stream_get_type(stream);
 	if (rtp) {
 		if (ast_sdp_state_get_stream_connection_address(sdp_state, 0, &address_rtp)) {
 			return -1;
@@ -1300,31 +1301,24 @@ static int sdp_add_m_from_rtp_stream(struct ast_sdp *sdp, const struct ast_sdp_s
 		ao2_ref(format, -1);
 	}
 
-	if (rtp && media_type != AST_MEDIA_TYPE_VIDEO) {
-		for (i = 1LL; i <= AST_RTP_MAX; i <<= 1) {
-			if (!(options->telephone_event & i)) {
-				continue;
-			}
-
-			rtp_code = ast_rtp_codecs_payload_code(
-				ast_rtp_instance_get_codecs(rtp), 0, NULL, i);
-			if (rtp_code == -1) {
-				continue;
-			}
-
+	media_type = ast_stream_get_type(stream);
+	if (rtp && media_type != AST_MEDIA_TYPE_VIDEO
+		&& (options->dtmf == AST_SDP_DTMF_RFC_4733 || options->dtmf == AST_SDP_DTMF_AUTO)) {
+		i = AST_RTP_DTMF;
+		rtp_code = ast_rtp_codecs_payload_code(
+			ast_rtp_instance_get_codecs(rtp), 0, NULL, i);
+		if (-1 < rtp_code) {
 			if (ast_sdp_m_add_format(m_line, options, rtp_code, 0, NULL, i)) {
 				ast_sdp_m_free(m_line);
 				return -1;
 			}
 
-			if (i == AST_RTP_DTMF) {
-				snprintf(tmp, sizeof(tmp), "%d 0-16", rtp_code);
-				a_line = ast_sdp_a_alloc("fmtp", tmp);
-				if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
-					ast_sdp_a_free(a_line);
-					ast_sdp_m_free(m_line);
-					return -1;
-				}
+			snprintf(tmp, sizeof(tmp), "%d 0-16", rtp_code);
+			a_line = ast_sdp_a_alloc("fmtp", tmp);
+			if (!a_line || ast_sdp_m_add_a(m_line, a_line)) {
+				ast_sdp_a_free(a_line);
+				ast_sdp_m_free(m_line);
+				return -1;
 			}
 		}
 	}