config: Use one equal sign for tests in configure.ac

configure.ac

@@ -423,10 +423,10 @@ AS_IF([test "x$enable_purple" != xno],
 					[PKG_CHECK_MODULES(_SIPE_MEDIA_RECHECK_PLEASE_IGNORE, [$purple_pkgconfig >= 2.8.0],
 						dnl check purple pkgconfig for gstreamer version
 						[gstreamer_pkgconfig=`$PKG_CONFIG --variable=gstreamer $purple_pkgconfig`
-						AS_IF([test "x$gstreamer_pkgconfig" == x],
-							[AS_IF([test "x$purple_pkgconfig" == xpurple-3],
+						AS_IF([test "x$gstreamer_pkgconfig" = x],
+							[AS_IF([test "x$purple_pkgconfig" = xpurple-3],
 								[gstreamer_pkgconfig=1.0])])
-						AS_IF([test "x$gstreamer_pkgconfig" == x1.0],
+						AS_IF([test "x$gstreamer_pkgconfig" = x1.0],
 							[gstreamer_pkgconfig="gstreamer-$gstreamer_pkgconfig gstreamer-rtp-$gstreamer_pkgconfig"
 
 							AC_MSG_NOTICE([using packages "$gstreamer_pkgconfig".])
@@ -535,18 +535,18 @@ AS_IF([test "x$with_purple_vv" != xno],
 AM_CONDITIONAL(SIPE_WITH_VV, [test "x$with_purple_vv" != xno])
 
 dnl raw data RTP streams enable Lync file transfer
-AM_CONDITIONAL(SIPE_HAVE_XDATA, [test "x$ac_have_xdata" == xyes])
-AS_IF([test "x$ac_have_xdata" == xyes],
+AM_CONDITIONAL(SIPE_HAVE_XDATA, [test "x$ac_have_xdata" = xyes])
+AS_IF([test "x$ac_have_xdata" = xyes],
 	[AC_DEFINE(HAVE_XDATA, 1, [Define if we have raw data RTP in media backend.])])
 
 dnl enable appshare support
-AM_CONDITIONAL(SIPE_HAVE_APPSHARE, [test "x$ac_have_appshare" == xyes])
-AS_IF([test "x$ac_have_appshare" == xyes],
+AM_CONDITIONAL(SIPE_HAVE_APPSHARE, [test "x$ac_have_appshare" = xyes])
+AS_IF([test "x$ac_have_appshare" = xyes],
 	[AC_DEFINE(HAVE_APPSHARE, 1, [Define to 1 if we have appshare support.])])
 
 dnl RDP server for sharing local desktop
 AM_CONDITIONAL(SIPE_HAVE_APPSHARE_SERVER, [test "x$ac_have_appshare_server" = xyes])
-AS_IF([test "x$ac_have_appshare_server" == xyes],
+AS_IF([test "x$ac_have_appshare_server" = xyes],
 	[AC_DEFINE(HAVE_APPSHARE_SERVER, 1, [Define if appshare server is enabled.])])
 
 dnl telepathy code parts rely on interfaces that require GValueArray. This

src/api/sipe-backend.h

@@ -395,6 +395,7 @@ struct sipe_media_stream {
 	struct sipe_media_call *call;
 	gchar *id;
 	struct ssrc_range *ssrc_range;
+	guint32 media_source_id;
 
 	void (*candidate_pairs_established_cb)(struct sipe_media_stream *);
 	void (*read_cb)(struct sipe_media_stream *);

src/api/sipe-core.h

@@ -499,9 +499,11 @@ sipe_core_media_stream_end(struct sipe_media_stream *stream);
  *
  * @param sipe_public (in) SIPE core data.
  * @param chat_session (in) chat session structure
+ * @param with_video (in) TRUE if a video call should be created.
  */
 void sipe_core_media_connect_conference(struct sipe_core_public *sipe_public,
-					struct sipe_chat_session *chat_session);
+					struct sipe_chat_session *chat_session,
+					gboolean with_video);
 
 /**
  * Retrieves the media call in progress
@@ -683,9 +685,11 @@ void sipe_core_status_set(struct sipe_core_public *sipe_public,
  *               of @c buffer MUST be at least @c SIPE_MSRTP_VSR_HEADER_LEN +
  *               @c SIPE_MSRTP_VSR_ENTRY_LEN.
  * @param payload_type (in) payload ID of the codec negotiated with the peer.
+ * @param media_source_id (in) ID of the video stream to request.
  */
 void sipe_core_msrtp_write_video_source_request(guint8 *buffer,
-						guint8 payload_type);
+						guint8 payload_type,
+						guint32 media_source_id);
 
 /**
  * Fills @buffer with customized Payload Content Scalability Information packet

src/core/sipe-conf.c

@@ -281,7 +281,8 @@ process_invite_conf_focus_response(struct sipe_core_private *sipe_private,
 #ifdef HAVE_VV
 			if (session->is_call)
 				sipe_core_media_connect_conference(SIPE_CORE_PUBLIC,
-								   session->chat_session);
+								   session->chat_session,
+								   FALSE);
 #endif
 		}
 		sipe_xml_free(xn_response);
@@ -1169,35 +1170,32 @@ process_conference_av_endpoint(const sipe_xml *endpoint,
 			       struct sip_session *session)
 {
 	const sipe_xml *media;
-	const gchar *new_entity;
 
-	if (!sipe_strequal(user_uri, self_uri)) {
-		/* We are interested only in our own endpoint data. */
-		return;
-	}
+	if (sipe_strequal(user_uri, self_uri)) {
+		const gchar *new_entity = sipe_xml_attribute(endpoint, "entity");
 
-	new_entity = sipe_xml_attribute(endpoint, "entity");
-	if (!sipe_strequal(session->audio_video_entity, new_entity)) {
-		g_free(session->audio_video_entity);
-		session->audio_video_entity = g_strdup(new_entity);
+		if (!sipe_strequal(session->audio_video_entity, new_entity)) {
+			g_free(session->audio_video_entity);
+			session->audio_video_entity = g_strdup(new_entity);
+		}
 	}
 
-	session->audio_media_id = 0;
-
 	media = sipe_xml_child(endpoint, "media");
 	for (; media; media = sipe_xml_twin(media)) {
 		gchar *type = sipe_xml_data(sipe_xml_child(media, "type"));
 
-		if (sipe_strequal(type, "audio")) {
-			session->audio_media_id =
-					sipe_xml_int_attribute(media, "id", 0);
+		if (sipe_strequal(type, "audio") && sipe_strequal(user_uri, self_uri)) {
+			session->audio_media_id = sipe_xml_int_attribute(media, "id", 0);
+		} else if (sipe_strequal(type, "video") && session->video_media_source_id == 0) {
+			const sipe_xml *child = sipe_xml_child(media, "media-source-id");
+			const gchar *data = sipe_xml_data(child);
+
+			if (data) {
+				session->video_media_source_id = atoi(data);
+			}
 		}
 
 		g_free(type);
-
-		if (session->audio_media_id != 0) {
-			break;
-		}
 	}
 }
 
@@ -1209,7 +1207,8 @@ call_accept_cb(struct sipe_core_private *sipe_private, struct conf_accept_ctx *c
 
 	if (session) {
 		sipe_core_media_connect_conference(SIPE_CORE_PUBLIC,
-						   session->chat_session);
+						   session->chat_session,
+						   FALSE);
 	}
 }
 

src/core/sipe-media.c

@@ -106,7 +106,7 @@ struct sipe_media_stream_private {
 #define SIPE_MEDIA_STREAM         ((struct sipe_media_stream *) stream_private)
 #define SIPE_MEDIA_STREAM_PRIVATE ((struct sipe_media_stream_private *) stream)
 
-#define SIPE_MEDIA_STREAM_CONNECTION_TIMEOUT_SECONDS 30
+#define SIPE_MEDIA_STREAM_CONNECTION_TIMEOUT_SECONDS 120
 #define SIPE_MEDIA_CALL_RINGING_TIMEOUT_SECONDS 60
 #define SIPE_MEDIA_CALL_TIMEOUT_SECONDS 120
 
@@ -426,13 +426,51 @@ get_encryption_policy(struct sipe_core_private *sipe_private)
 	return result;
 }
 
+static GList *
+get_local_codecs(struct sipe_media_call_private *call_private,
+		 struct sipe_media_stream_private *stream_private)
+{
+	struct sipe_core_private *sipe_private = call_private->sipe_private;
+	gboolean is_conference = g_strstr_len(SIPE_MEDIA_CALL->with,
+					      strlen(SIPE_MEDIA_CALL->with),
+					      "app:conf:audio-video:") != NULL;
+	GList *codecs = sipe_backend_get_local_codecs(SIPE_MEDIA_CALL,
+						      SIPE_MEDIA_STREAM);
+	GList *i;
+
+	for (i = codecs; i; i = i->next) {
+		struct sipe_backend_codec *codec = i->data;
+		char *name = sipe_backend_codec_get_name(codec);
+
+		if (/* Do not announce Theora. Its optional parameters are too
+		     * long, Communicator rejects such SDP message and does not
+		     * support the codec anyway. */
+		    sipe_strequal(name,"THEORA") ||
+		    /* For an unknown reason, MS Lync A/V conferencing server
+		     * does not accept SIPE audio encoded with SIREN. Disable
+		     * the codec when SIPE is connected to a server version
+		     * prior Skype for Business. */
+		    (!SIPE_CORE_PRIVATE_FLAG_IS(SFB) && is_conference &&
+		     sipe_strequal(name,"SIREN"))) {
+			GList *tmp;
+			sipe_backend_codec_free(codec);
+			tmp = i->next;
+			codecs = g_list_delete_link(codecs, i);
+			i = tmp;
+		}
+
+		g_free(name);
+	}
+
+	return codecs;
+}
+
 static struct sdpmedia *
 media_stream_to_sdpmedia(struct sipe_media_call_private *call_private,
 			 struct sipe_media_stream_private *stream_private)
 {
 	struct sdpmedia *sdpmedia = g_new0(struct sdpmedia, 1);
-	GList *codecs = sipe_backend_get_local_codecs(SIPE_MEDIA_CALL,
-						      SIPE_MEDIA_STREAM);
+	GList *codecs = get_local_codecs(call_private, stream_private);
 	SipeEncryptionPolicy encryption_policy =
 			get_encryption_policy(call_private->sipe_private);
 	guint rtcp_port = 0;
@@ -809,6 +847,10 @@ update_call_from_remote_sdp(struct sipe_media_call_private* call_private,
 						 backend_candidates);
 	sipe_media_candidate_list_free(backend_candidates);
 
+	if (sipe_strequal(stream->id, "video") && call_private->conference_session) {
+		stream->media_source_id = call_private->conference_session->video_media_source_id;
+	}
+
 	SIPE_MEDIA_STREAM_PRIVATE->remote_candidates_and_codecs_set = TRUE;
 
 	return TRUE;
@@ -1386,6 +1428,8 @@ sipe_media_stream_add(struct sipe_media_call *call, const gchar *id,
 					    ssrc_count);
 	}
 
+	SIPE_MEDIA_STREAM->media_source_id = SIPE_MSRTP_VSR_SOURCE_ANY;
+
 	SIPE_MEDIA_STREAM->backend_private =
 			sipe_backend_media_add_stream(SIPE_MEDIA_STREAM,
 						      type, ice_version,
@@ -1529,7 +1573,8 @@ conference_audio_muted_cb(struct sipe_media_stream *stream, gboolean is_muted)
 }
 
 void sipe_core_media_connect_conference(struct sipe_core_public *sipe_public,
-					struct sipe_chat_session *chat_session)
+					struct sipe_chat_session *chat_session,
+					gboolean with_video)
 {
 	struct sipe_core_private *sipe_private = SIPE_CORE_PRIVATE;
 	struct sipe_media_call_private *call_private;
@@ -1567,6 +1612,8 @@ void sipe_core_media_connect_conference(struct sipe_core_public *sipe_public,
 	call_private->conference_session = session;
 	SIPE_MEDIA_CALL->call_reject_cb = av_call_reject_cb;
 
+	g_free(av_uri);
+
 	stream = sipe_media_stream_add(SIPE_MEDIA_CALL, "audio",
 				       SIPE_MEDIA_AUDIO,
 				       call_private->ice_version,
@@ -1577,11 +1624,23 @@ void sipe_core_media_connect_conference(struct sipe_core_public *sipe_public,
 					  _("Error creating audio stream"));
 
 		sipe_media_hangup(call_private);
+		return;
 	}
 
 	stream->mute_cb = conference_audio_muted_cb;
 
-	g_free(av_uri);
+	if (with_video) {
+		stream = sipe_media_stream_add(SIPE_MEDIA_CALL, "video",
+					       SIPE_MEDIA_VIDEO,
+					       call_private->ice_version,
+					       TRUE, VIDEO_SSRC_COUNT);
+		if (!stream) {
+			sipe_backend_notify_error(sipe_public,
+						  _("Error occurred"),
+						  _("Error creating video stream"));
+			sipe_media_hangup(call_private);
+		}
+	}
 
 	// Processing continues in stream_initialized_cb
 }

src/core/sipe-msrtp.c

@@ -104,7 +104,8 @@ enum
 
 void
 sipe_core_msrtp_write_video_source_request(guint8 *buffer,
-					   guint8 payload_type)
+					   guint8 payload_type,
+					   guint32 media_source_id)
 {
 	static guint8 bit_rate_histogram[] = {
 		0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0
@@ -123,7 +124,7 @@ sipe_core_msrtp_write_video_source_request(guint8 *buffer,
 	SIPE_WRITE_UINT16_BE(buffer, SIPE_MSRTP_VSR_HEADER_LEN +
 			     SIPE_MSRTP_VSR_ENTRY_LEN);
 	// Requested Media Source ID
-	SIPE_WRITE_UINT32_BE(buffer, SIPE_MSRTP_VSR_SOURCE_ANY);
+	SIPE_WRITE_UINT32_BE(buffer, media_source_id);
 	SIPE_WRITE_UINT16_BE(buffer, 1); // Request Id
 	SIPE_WRITE_UINT16_BE(buffer, 0); // Reserve1
 	SIPE_WRITE_UINT8(buffer, 0); // Version

src/core/sipe-session.h

@@ -70,6 +70,7 @@ struct sip_session {
 	GHashTable *conf_unconfirmed_messages;
 	gchar *audio_video_entity;
 	guint audio_media_id;
+	guint video_media_source_id;
 
 	/*
 	 * Media call related fields

src/purple/purple-chat.c

@@ -257,13 +257,26 @@ static void sipe_purple_chat_menu_lock_cb(SIPE_UNUSED_PARAMETER PurpleChat *chat
 
 #ifdef HAVE_VV
 
-static void sipe_purple_chat_menu_join_call_cb(SIPE_UNUSED_PARAMETER PurpleChat *chat,
-					       PurpleConversation *conv)
+static void
+join_conference_call(PurpleConversation *conv, gboolean with_video)
 {
 	struct sipe_core_public *sipe_public = PURPLE_CONV_TO_SIPE_CORE_PUBLIC;
 	struct sipe_chat_session *chat_session = sipe_purple_chat_get_session(conv);
-	SIPE_DEBUG_INFO("sipe_purple_chat_join_call_cb: %p %p", conv, chat_session);
-	sipe_core_media_connect_conference(sipe_public, chat_session);
+	sipe_core_media_connect_conference(sipe_public, chat_session, with_video);
+}
+
+static void sipe_purple_chat_menu_join_call_cb(SIPE_UNUSED_PARAMETER PurpleChat *chat,
+					       PurpleConversation *conv)
+{
+	SIPE_DEBUG_INFO("sipe_purple_chat_join_call_cb: %p", conv);
+	join_conference_call(conv, FALSE);
+}
+
+static void sipe_purple_chat_menu_join_video_call_cb(SIPE_UNUSED_PARAMETER PurpleChat *chat,
+						     PurpleConversation *conv)
+{
+	SIPE_DEBUG_INFO("sipe_purple_chat_join_video_call_cb: %p", conv);
+	join_conference_call(conv, TRUE);
 }
 
 #ifdef HAVE_APPSHARE
@@ -349,12 +362,15 @@ sipe_purple_chat_menu(PurpleChat *chat)
 		case SIPE_CHAT_TYPE_MULTIPARTY:
 #ifdef HAVE_VV
 			if (!sipe_core_media_get_call(PURPLE_CONV_TO_SIPE_CORE_PUBLIC)) {
-				act = NULL;
+				act = purple_menu_action_new(_("Join conference video call"),
+							     PURPLE_CALLBACK(sipe_purple_chat_menu_join_video_call_cb),
+							     conv, NULL);
+				menu = g_list_prepend(menu, act);
+
 				act = purple_menu_action_new(_("Join conference call"),
 							     PURPLE_CALLBACK(sipe_purple_chat_menu_join_call_cb),
 							     conv, NULL);
-				if (act)
-					menu = g_list_prepend(menu, act);
+				menu = g_list_prepend(menu, act);
 			}
 #ifdef HAVE_APPSHARE
 			role = sipe_core_conf_get_appshare_role(PURPLE_CONV_TO_SIPE_CORE_PUBLIC,