Feat/add websocket text support

bridges/bridge_native_rtp.c

@@ -643,7 +643,7 @@ static int native_rtp_bridge_compatible_check(struct ast_bridge *bridge, struct
 	RAII_VAR(struct rtp_glue_data *, glue0, NULL, rtp_glue_data_destroy);
 	RAII_VAR(struct rtp_glue_data *, glue1, NULL, rtp_glue_data_destroy);
 
-	ast_debug(1, "Bridge '%s'.  Checking compatability for channels '%s' and '%s'\n",
+	ast_debug(1, "Bridge '%s'. Checking compatibility for channels '%s' and '%s'\n",
 		bridge->uniqueid, ast_channel_name(bc0->chan), ast_channel_name(bc1->chan));
 
 	if (!native_rtp_bridge_capable(bc0->chan)) {

configure

@@ -714,7 +714,7 @@ AST_NO_STRINGOP_TRUNCATION
 AST_NO_FORMAT_Y2K
 AST_NO_FORMAT_TRUNCATION
 AST_NO_STRICT_OVERFLOW
-AST_FORTIFY_SOURCE
+# AST_FORTIFY_SOURCE
 AST_TRAMPOLINES
 AST_DECLARATION_AFTER_STATEMENT
 AST_UNDEFINED_SANITIZER

include/asterisk/res_pjsip.h

@@ -887,6 +887,26 @@ struct ast_sip_t38_configuration {
 	unsigned int bind_udptl_to_media_address;
 };
 
+/*!
+ * \brief WebRTC Datachannel configuration for SIP endpoints
+ */
+struct ast_sip_webrtc_datachannel_configuration
+{
+	/*! Whether WebRTC Datachannel support is enabled or not */
+	unsigned int enabled;
+	/*! Whether to use IPv6 for WebRTC Datachannel or not */
+	unsigned int ipv6;
+};
+
+/*!
+ * \brief Websocket text configuration for SIP endpoints
+ */
+struct ast_sip_websocket_text_configuration
+{
+	/*! Whether websocket text support is enabled or not */
+	unsigned int enabled;
+};
+
 /*!
  * \brief Media configuration for SIP endpoints
  */
@@ -905,6 +925,10 @@ struct ast_sip_endpoint_media_configuration {
 	struct ast_sip_direct_media_configuration direct_media;
 	/*! T.38 (FoIP) options */
 	struct ast_sip_t38_configuration t38;
+	/*! WebRTC Datachannel configuration options */
+	struct ast_sip_webrtc_datachannel_configuration webrtc_datachannel_configuration;
+	/*! Websocket text configuration options */
+	struct ast_sip_websocket_text_configuration websocket_text_configuration;
 	/*! Configured codecs */
 	struct ast_format_cap *codecs;
 	/*! Capabilities in topology form */
@@ -921,7 +945,7 @@ struct ast_sip_endpoint_media_configuration {
 	unsigned int tos_text;
 	/*! Priority for text streams */
 	unsigned int cos_text;
-	/*! Indicate if text stream supports RED */
+	/*! Flag indicate if text stream supports RED */
 	unsigned int red_enabled;
 	/*! Is g.726 packed in a non standard way */
 	unsigned int g726_non_standard;

include/asterisk/res_pjsip_session.h

@@ -51,6 +51,7 @@ struct pjmedia_sdp_media;
 struct pjmedia_sdp_session;
 struct ast_dsp;
 struct ast_udptl;
+struct ast_websocket_session_text;
 
 /*! \brief T.38 states for a session */
 enum ast_sip_session_t38state {
@@ -127,6 +128,9 @@ struct ast_sip_session_media {
 	char *remote_label;
 	/*! \brief Stream name */
 	char *stream_name;
+
+	/*! \brief Websocket session text instance itself */
+	struct ast_websocket_session_text *websocket_session_text;
 };
 
 /*!

include/asterisk/strings.h

@@ -94,7 +94,7 @@ static force_inline int attribute_pure ast_strlen_zero(const char *s)
   \retval 1 if \a str begins with \a prefix
   \retval 0 otherwise.
  */
-static int force_inline attribute_pure ast_begins_with(const char *str, const char *prefix)
+static force_inline int attribute_pure ast_begins_with(const char *str, const char *prefix)
 {
 	ast_assert(str != NULL);
 	ast_assert(prefix != NULL);
@@ -113,7 +113,7 @@ static int force_inline attribute_pure ast_begins_with(const char *str, const ch
   \retval 1 if \a str ends with \a suffix
   \retval 0 otherwise.
  */
-static int force_inline attribute_pure ast_ends_with(const char *str, const char *suffix)
+static force_inline int attribute_pure ast_ends_with(const char *str, const char *suffix)
 {
 	size_t str_len;
 	size_t suffix_len;

main/channel.c

@@ -4818,6 +4818,7 @@ int ast_sendtext_data(struct ast_channel *chan, struct ast_msg_data *msg)
 		f.datalen = body_len;
 		f.mallocd = AST_MALLOCD_DATA;
 		f.data.ptr = ast_strdup(body);
+		f.stream_num = ast_stream_get_position(ast_channel_get_default_stream(chan, AST_MEDIA_TYPE_TEXT));
 		if (f.data.ptr) {
 			res = ast_channel_tech(chan)->write_text(chan, &f);
 		} else {

res/res_pjsip/pjsip_config.xml

@@ -998,7 +998,7 @@
 				<configOption name="tos_text">
 					<synopsis>DSCP TOS bits for text streams</synopsis>
 					<description><para>
-						See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
+						See https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service for more information about QoS settings
 					</para></description>
 				</configOption>
 				<configOption name="cos_audio">
@@ -1016,9 +1016,12 @@
 				<configOption name="cos_text">
 					<synopsis>Priority for text streams</synopsis>
 					<description><para>
-							See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings
+						See https://docs.asterisk.org/Configuration/Channel-Drivers/IP-Quality-of-Service for more information about QoS settings
 					</para></description>
 				</configOption>
+				<configOption name="ws_text" default="no">
+					<synopsis>Allow use of websocket text for WebRTC traffic</synopsis>
+				</configOption>
 				<configOption name="allow_subscribe" default="yes">
 					<synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis>
 				</configOption>

res/res_pjsip/pjsip_configuration.c

@@ -2217,6 +2217,9 @@ int ast_res_pjsip_initialize_configuration(void)
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "named_call_group", "", named_groups_handler, named_callgroups_to_str, NULL, 0, 0);
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "named_pickup_group", "", named_groups_handler, named_pickupgroups_to_str, NULL, 0, 0);
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "device_state_busy_at", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, devicestate_busy_at));
+
+	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "ws_text", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.websocket_text_configuration.enabled));
+
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "t38_udptl", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.t38.enabled));
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "t38_udptl_ec", "none", t38udptl_ec_handler, t38udptl_ec_to_str, NULL, 0, 0);
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "t38_udptl_maxdatagram", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, media.t38.maxdatagram));

res/res_pjsip/pjsip_manager.xml

@@ -252,6 +252,9 @@
 				<parameter name="RewriteContact">
 					<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='rewrite_contact']/synopsis/node())"/></para>
 				</parameter>
+				<parameter name="WsText">
+					<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='ws_text']/synopsis/node())"/></para>
+				</parameter>
 				<parameter name="Transport">
 					<para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='transport']/synopsis/node())"/></para>
 				</parameter>

res/res_pjsip_sdp_rtp.c

@@ -384,7 +384,7 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
 						red_cp = strtok_r(NULL, "/", &rest);
 					}
 
-					if (++red_num_gen > 0) {
+					if (red_num_gen > 0) {
 						ast_log(AST_LOG_NOTICE, "T.140/RED enabled (pt=%d) with %d generations\n", num, red_num_gen);
 						session->endpoint->media.red_enabled = 1;
 						ast_rtp_red_init(session_media->rtp, 300, red_data_pt, red_num_gen);
@@ -975,7 +975,7 @@ static enum ast_sip_session_media_encryption get_media_encryption_type(pj_str_t
 
 	*optimistic = 0;
 
-	if (!transport_str) {
+	if (!transport_str || !strstr(transport_str, "AVP")) {
 		return AST_SIP_MEDIA_TRANSPORT_INVALID;
 	}
 	if (strstr(transport_str, "UDP/TLS")) {
@@ -1536,6 +1536,12 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
 		SCOPE_EXIT_RTN_VALUE(0, "Endpoint has no codecs\n");
 	}
 
+	RAII_VAR(char *, transport_str, ast_strndup(stream->desc.transport.ptr, stream->desc.transport.slen), ast_free);
+
+	if (!transport_str || !strstr(transport_str, "AVP")) {
+		SCOPE_EXIT_RTN_VALUE(0, "Incompatible transport\n");
+	}
+
 	/* Ensure incoming transport is compatible with the endpoint's configuration */
 	if (!session->endpoint->media.rtp.use_received_transport) {
 		encryption = check_endpoint_media_transport(session->endpoint, stream);
@@ -1974,6 +1980,10 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 			media->attr[media->attr_count++] = attr;
 		}
 
+		if (media_type == AST_MEDIA_TYPE_TEXT) {
+			ast_debug(3, "SDP generate RTP frame text %d %s\n", rtp_code, ast_format_get_codec_name(format));
+		}
+
 		if (media_type == AST_MEDIA_TYPE_TEXT && !strcasecmp(ast_format_get_codec_name(format), "red")) {
 			// TODO jpb: En attendant de faire mieux.
 			if (rtp_code == 105 || rtp_code == 96) {
@@ -2150,6 +2160,12 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
 		SCOPE_EXIT_RTN_VALUE(1, "No channel\n");
 	}
 
+	RAII_VAR(char *, transport_str, ast_strndup(remote_stream->desc.transport.ptr, remote_stream->desc.transport.slen), ast_free);
+
+	if (!transport_str || !strstr(transport_str, "AVP")) {
+		SCOPE_EXIT_RTN_VALUE(0, "Incompatible transport\n");
+	}
+
 	/* Ensure incoming transport is compatible with the endpoint's configuration */
 	if (!session->endpoint->media.rtp.use_received_transport &&
 		check_endpoint_media_transport(session->endpoint, remote_stream) == AST_SIP_MEDIA_TRANSPORT_INVALID) {
@@ -2347,7 +2363,7 @@ static void stream_destroy(struct ast_sip_session_media *session_media)
 
 /*! \brief SDP handler for 'audio' media stream */
 static struct ast_sip_session_sdp_handler audio_sdp_handler = {
-	.id = STR_AUDIO,
+	.id = "audio_rtp", //STR_AUDIO, 
 	.negotiate_incoming_sdp_stream = negotiate_incoming_sdp_stream,
 	.create_outgoing_sdp_stream = create_outgoing_sdp_stream,
 	.apply_negotiated_sdp_stream = apply_negotiated_sdp_stream,
@@ -2358,7 +2374,7 @@ static struct ast_sip_session_sdp_handler audio_sdp_handler = {
 
 /*! \brief SDP handler for 'video' media stream */
 static struct ast_sip_session_sdp_handler video_sdp_handler = {
-	.id = STR_VIDEO,
+	.id = "video_rtp", //STR_VIDEO,
 	.negotiate_incoming_sdp_stream = negotiate_incoming_sdp_stream,
 	.create_outgoing_sdp_stream = create_outgoing_sdp_stream,
 	.apply_negotiated_sdp_stream = apply_negotiated_sdp_stream,
@@ -2370,7 +2386,7 @@ static struct ast_sip_session_sdp_handler video_sdp_handler = {
 
 /*! \brief SDP handler for 'text' media stream */
 static struct ast_sip_session_sdp_handler text_sdp_handler = {
-	.id = STR_TEXT,
+	.id = "test_rtp", //STR_TEXT,
 	.negotiate_incoming_sdp_stream = negotiate_incoming_sdp_stream,
 	.create_outgoing_sdp_stream = create_outgoing_sdp_stream,
 	.apply_negotiated_sdp_stream = apply_negotiated_sdp_stream,
@@ -2458,7 +2474,7 @@ static int load_module(void)
 		ast_log(LOG_ERROR, "Unable to register SDP handler for %s stream type\n", STR_VIDEO);
 		goto end;
 	}
-
+	
 	if (ast_sip_session_register_sdp_handler(&text_sdp_handler, STR_TEXT)) {
 		ast_log(LOG_ERROR, "Unable to register SDP handler for %s stream type\n", STR_TEXT);
 		goto end;

res/res_pjsip_session.c

@@ -59,8 +59,8 @@
 #define MOD_DATA_ON_RESPONSE "on_response"
 #define MOD_DATA_NAT_HOOK "nat_hook"
 
-/* Most common case is one audio and one video stream */
-#define DEFAULT_NUM_SESSION_MEDIA 2
+/* Most common case is one audio, one video stream and one text stream */
+#define DEFAULT_NUM_SESSION_MEDIA 3
 
 /* Some forward declarations */
 static void handle_session_begin(struct ast_sip_session *session);