sip: Add media manager and sipify origin

* pipeline: we should bind the used socket of our udpsink to the socket udpsrc
2021-02-17 23:48:33 +01:00
parent 2dfa42d48d
commit 967f30d688
13 changed files with 1669 additions and 48 deletions
--- a/plugins/sip/calls-sip-media-manager.c
+++ b/plugins/sip/calls-sip-media-manager.c
@@ -0,0 +1,127 @@
+/*
+ * Copyright (C) 2021 Purism SPC
+ *
+ * This file is part of Calls.
+ *
+ * Calls is free software: you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * Calls is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with Calls.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * Author: Evangelos Ribeiro Tzaras <evangelos.tzaras@puri.sm>
+ *
+ * SPDX-License-Identifier: GPL-3.0-or-later
+ *
+ */
+
+#define G_LOG_DOMAIN "CallsCallsSipMediaManager"
+
+#include "calls-sip-media-pipeline.h"
+#include "gst-rfc3551.h"
+#include "calls-sip-media-manager.h"
+
+#include <gst/gst.h>
+
+typedef struct _CallsSipMediaManager
+{
+  GObject parent;
+} CallsSipMediaManager;
+
+G_DEFINE_TYPE (CallsSipMediaManager, calls_sip_media_manager, G_TYPE_OBJECT);
+
+
+MediaCodecInfo*
+get_best_codec (CallsSipMediaManager *self)
+{
+  return media_codec_by_name ("PCMU");
+}
+
+
+static void
+calls_sip_media_manager_finalize (GObject *object)
+{
+  gst_deinit ();
+
+  G_OBJECT_CLASS (calls_sip_media_manager_parent_class)->finalize (object);
+}
+
+
+static void
+calls_sip_media_manager_class_init (CallsSipMediaManagerClass *klass)
+{
+  GObjectClass *object_class = G_OBJECT_CLASS (klass);
+
+  object_class->finalize = calls_sip_media_manager_finalize;
+}
+
+
+static void
+calls_sip_media_manager_init (CallsSipMediaManager *self)
+{
+  gst_init (NULL, NULL);
+}
+
+
+/* Public functions */
+
+CallsSipMediaManager *
+calls_sip_media_manager_default ()
+{
+  static CallsSipMediaManager *instance = NULL;
+
+  if (instance == NULL) {
+    g_debug ("Creating CallsSipMediaManager");
+    instance = g_object_new (CALLS_TYPE_SIP_MEDIA_MANAGER, NULL);
+    g_object_add_weak_pointer (G_OBJECT (instance), (gpointer *) &instance);
+  }
+  return instance;
+}
+
+
+/* calls_sip_media_manager_static_capabilities:
+ *
+ * @port: Should eventually come from the ICE stack
+ * @use_srtp: Whether to use srtp (not really handled)
+ *
+ * Returns: (full-control) string describing capabilities
+ * to be used in the session description (SDP)
+ */
+char *
+calls_sip_media_manager_static_capabilities (CallsSipMediaManager *self,
+                                             guint                 port,
+                                             gboolean              use_srtp)
+{
+  char *attribute_line = "rtpmap:0 PCMU/8000";
+  char *payload_type = use_srtp ? "SAVP" : "AVP";
+  g_autofree char *media_line = NULL;
+
+  g_return_val_if_fail (CALLS_IS_SIP_MEDIA_MANAGER (self), NULL);
+
+  media_line = g_strdup_printf ("audio %d RTP/%s 0", port, payload_type);
+
+  /* TODO we can have multiple attribute lines (or media lines for that matter) */
+  /* TODO add attribute describing RTCP stream */
+  return g_strdup_printf ("v=0\r\n"
+                          "m=%s\r\n"
+                          "a=%s\r\n",
+                          media_line,
+                          attribute_line);
+}
+
+
+/* TODO lookup plugins in GStreamer */
+gboolean
+calls_sip_media_manager_supports_media (CallsSipMediaManager *self,
+                                        const char *media_type)
+{
+  return TRUE;
+}
+