Jabber: initial version of Jingle support

12 files changed, 2178 insertions, 0 deletions

diff --git a/include/gst/rtp/gstrtcpbuffer.h b/include/gst/rtp/gstrtcpbuffer.h
new file mode 100644
index 0000000000..b6410a5a1f
--- /dev/null
+++ b/include/gst/rtp/gstrtcpbuffer.h
@@ -0,0 +1,671 @@
+/* GStreamer
+ * Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
+ *
+ * gstrtcpbuffer.h: various helper functions to manipulate buffers
+ *     with RTCP payload.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTCPBUFFER_H__
+#define __GST_RTCPBUFFER_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/rtp-prelude.h>
+
+G_BEGIN_DECLS
+
+/**
+ * GST_RTCP_VERSION:
+ *
+ * The supported RTCP version 2.
+ */
+#define GST_RTCP_VERSION 2
+
+/**
+ * GstRTCPType:
+ * @GST_RTCP_TYPE_INVALID: Invalid type
+ * @GST_RTCP_TYPE_SR: Sender report
+ * @GST_RTCP_TYPE_RR: Receiver report
+ * @GST_RTCP_TYPE_SDES: Source description
+ * @GST_RTCP_TYPE_BYE: Goodbye
+ * @GST_RTCP_TYPE_APP: Application defined
+ * @GST_RTCP_TYPE_RTPFB: Transport layer feedback.
+ * @GST_RTCP_TYPE_PSFB: Payload-specific feedback.
+ * @GST_RTCP_TYPE_XR: Extended report.
+ *
+ * Different RTCP packet types.
+ */
+typedef enum
+{
+  GST_RTCP_TYPE_INVALID = 0,
+  GST_RTCP_TYPE_SR      = 200,
+  GST_RTCP_TYPE_RR      = 201,
+  GST_RTCP_TYPE_SDES    = 202,
+  GST_RTCP_TYPE_BYE     = 203,
+  GST_RTCP_TYPE_APP     = 204,
+  GST_RTCP_TYPE_RTPFB   = 205,
+  GST_RTCP_TYPE_PSFB    = 206,
+  GST_RTCP_TYPE_XR      = 207
+} GstRTCPType;
+
+/* FIXME 2.0: backwards compatibility define for enum typo */
+#define GST_RTCP_RTPFB_TYPE_RCTP_SR_REQ GST_RTCP_RTPFB_TYPE_RTCP_SR_REQ
+
+/**
+ * GstRTCPFBType:
+ * @GST_RTCP_FB_TYPE_INVALID: Invalid type
+ * @GST_RTCP_RTPFB_TYPE_NACK: Generic NACK
+ * @GST_RTCP_RTPFB_TYPE_TMMBR: Temporary Maximum Media Stream Bit Rate Request
+ * @GST_RTCP_RTPFB_TYPE_TMMBN: Temporary Maximum Media Stream Bit Rate
+ *    Notification
+ * @GST_RTCP_RTPFB_TYPE_RTCP_SR_REQ: Request an SR packet for early
+ *    synchronization
+ * @GST_RTCP_PSFB_TYPE_PLI: Picture Loss Indication
+ * @GST_RTCP_PSFB_TYPE_SLI: Slice Loss Indication
+ * @GST_RTCP_PSFB_TYPE_RPSI: Reference Picture Selection Indication
+ * @GST_RTCP_PSFB_TYPE_AFB: Application layer Feedback
+ * @GST_RTCP_PSFB_TYPE_FIR: Full Intra Request Command
+ * @GST_RTCP_PSFB_TYPE_TSTR: Temporal-Spatial Trade-off Request
+ * @GST_RTCP_PSFB_TYPE_TSTN: Temporal-Spatial Trade-off Notification
+ * @GST_RTCP_PSFB_TYPE_VBCN: Video Back Channel Message
+ *
+ * Different types of feedback messages.
+ */
+typedef enum
+{
+  /* generic */
+  GST_RTCP_FB_TYPE_INVALID        = 0,
+  /* RTPFB types */
+  GST_RTCP_RTPFB_TYPE_NACK        = 1,
+  /* RTPFB types assigned in RFC 5104 */
+  GST_RTCP_RTPFB_TYPE_TMMBR       = 3,
+  GST_RTCP_RTPFB_TYPE_TMMBN       = 4,
+  /* RTPFB types assigned in RFC 6051 */
+  GST_RTCP_RTPFB_TYPE_RTCP_SR_REQ = 5,
+  /* draft-holmer-rmcat-transport-wide-cc-extensions-01 */
+  GST_RTCP_RTPFB_TYPE_TWCC         = 15,
+
+  /* PSFB types */
+  GST_RTCP_PSFB_TYPE_PLI          = 1,
+  GST_RTCP_PSFB_TYPE_SLI          = 2,
+  GST_RTCP_PSFB_TYPE_RPSI         = 3,
+  GST_RTCP_PSFB_TYPE_AFB          = 15,
+  /* PSFB types assigned in RFC 5104 */
+  GST_RTCP_PSFB_TYPE_FIR          = 4,
+  GST_RTCP_PSFB_TYPE_TSTR         = 5,
+  GST_RTCP_PSFB_TYPE_TSTN         = 6,
+  GST_RTCP_PSFB_TYPE_VBCN         = 7,
+} GstRTCPFBType;
+
+/**
+ * GstRTCPSDESType:
+ * @GST_RTCP_SDES_INVALID: Invalid SDES entry
+ * @GST_RTCP_SDES_END: End of SDES list
+ * @GST_RTCP_SDES_CNAME: Canonical name
+ * @GST_RTCP_SDES_NAME: User name
+ * @GST_RTCP_SDES_EMAIL: User's electronic mail address
+ * @GST_RTCP_SDES_PHONE: User's phone number
+ * @GST_RTCP_SDES_LOC: Geographic user location
+ * @GST_RTCP_SDES_TOOL: Name of application or tool
+ * @GST_RTCP_SDES_NOTE: Notice about the source
+ * @GST_RTCP_SDES_PRIV: Private extensions
+ *
+ * Different types of SDES content.
+ */
+/**
+ * GST_RTCP_SDES_H323_CADDR:
+ *
+ * H.323 callable address
+ *
+ * Since: 1.20:
+ */
+/**
+ * GST_RTCP_SDES_APSI:
+ *
+ * Application Specific Identifier (RFC6776)
+ *
+ * Since: 1.20:
+ */
+/**
+ * GST_RTCP_SDES_RGRP:
+ *
+ * Reporting Group Identifier (RFC8861)
+ *
+ * Since: 1.20:
+ */
+/**
+ * GST_RTCP_SDES_RTP_STREAM_ID:
+ *
+ * RtpStreamId SDES item (RFC8852).
+ *
+ * Since: 1.20:
+ */
+/**
+ * GST_RTCP_SDES_REPAIRED_RTP_STREAM_ID:
+ *
+ * RepairedRtpStreamId SDES item (RFC8852).
+ *
+ * Since: 1.20:
+ */
+/**
+ * GST_RTCP_SDES_CCID:
+ *
+ * CLUE CaptId (RFC8849)
+ *
+ * Since: 1.20:
+ */
+/**
+ * GST_RTCP_SDES_MID:
+ *
+ * MID SDES item (RFC8843).
+ *
+ * Since: 1.20:
+ */
+typedef enum
+{
+  GST_RTCP_SDES_INVALID                 = -1,
+  GST_RTCP_SDES_END                     = 0,
+  GST_RTCP_SDES_CNAME                   = 1,
+  GST_RTCP_SDES_NAME                    = 2,
+  GST_RTCP_SDES_EMAIL                   = 3,
+  GST_RTCP_SDES_PHONE                   = 4,
+  GST_RTCP_SDES_LOC                     = 5,
+  GST_RTCP_SDES_TOOL                    = 6,
+  GST_RTCP_SDES_NOTE                    = 7,
+  GST_RTCP_SDES_PRIV                    = 8,
+  GST_RTCP_SDES_H323_CADDR              = 9,
+  GST_RTCP_SDES_APSI                    = 10,
+  GST_RTCP_SDES_RGRP                    = 11,
+  GST_RTCP_SDES_RTP_STREAM_ID           = 12,
+  GST_RTCP_SDES_REPAIRED_RTP_STREAM_ID  = 13,
+  GST_RTCP_SDES_CCID                    = 14,
+  GST_RTCP_SDES_MID                     = 15,
+} GstRTCPSDESType;
+
+/**
+ * GstRTCPXRType:
+ * @GST_RTCP_XR_TYPE_INVALID: Invalid XR Report Block
+ * @GST_RTCP_XR_TYPE_LRLE: Loss RLE Report Block
+ * @GST_RTCP_XR_TYPE_DRLE: Duplicate RLE Report Block
+ * @GST_RTCP_XR_TYPE_PRT: Packet Receipt Times Report Block
+ * @GST_RTCP_XR_TYPE_RRT: Receiver Reference Time Report Block
+ * @GST_RTCP_XR_TYPE_DLRR: Delay since the last Receiver Report
+ * @GST_RTCP_XR_TYPE_SSUMM: Statistics Summary Report Block
+ * @GST_RTCP_XR_TYPE_VOIP_METRICS: VoIP Metrics Report Block
+ *
+ * Types of RTCP Extended Reports, those are defined in RFC 3611 and other RFCs
+ * according to the [IANA registry](https://www.iana.org/assignments/rtcp-xr-block-types/rtcp-xr-block-types.xhtml).
+ *
+ * Since: 1.16
+ */
+typedef enum
+{
+  GST_RTCP_XR_TYPE_INVALID      = -1,
+  GST_RTCP_XR_TYPE_LRLE         = 1,
+  GST_RTCP_XR_TYPE_DRLE         = 2,
+  GST_RTCP_XR_TYPE_PRT          = 3,
+  GST_RTCP_XR_TYPE_RRT          = 4,
+  GST_RTCP_XR_TYPE_DLRR         = 5,
+  GST_RTCP_XR_TYPE_SSUMM        = 6,
+  GST_RTCP_XR_TYPE_VOIP_METRICS = 7
+} GstRTCPXRType;
+
+/**
+ * GST_RTCP_MAX_SDES:
+ *
+ * The maximum text length for an SDES item.
+ */
+#define GST_RTCP_MAX_SDES 255
+
+/**
+ * GST_RTCP_MAX_RB_COUNT:
+ *
+ * The maximum amount of Receiver report blocks in RR and SR messages.
+ */
+#define GST_RTCP_MAX_RB_COUNT   31
+
+/**
+ * GST_RTCP_MAX_SDES_ITEM_COUNT:
+ *
+ * The maximum amount of SDES items.
+ */
+#define GST_RTCP_MAX_SDES_ITEM_COUNT   31
+
+/**
+ * GST_RTCP_MAX_BYE_SSRC_COUNT:
+ *
+ * The maximum amount of SSRCs in a BYE packet.
+ */
+#define GST_RTCP_MAX_BYE_SSRC_COUNT   31
+
+/**
+ * GST_RTCP_VALID_MASK:
+ *
+ * Mask for version, padding bit and packet type pair
+ */
+#define GST_RTCP_VALID_MASK (0xc000 | 0x2000 | 0xfe)
+
+/**
+ * GST_RTCP_REDUCED_SIZE_VALID_MASK:
+ *
+ * Mask for version and packet type pair allowing reduced size
+ * packets, basically it accepts other types than RR and SR
+ */
+#define GST_RTCP_REDUCED_SIZE_VALID_MASK (0xc000 | 0xf8)
+
+/**
+ * GST_RTCP_VALID_VALUE:
+ *
+ * Valid value for the first two bytes of an RTCP packet after applying
+ * #GST_RTCP_VALID_MASK to them.
+ */
+#define GST_RTCP_VALID_VALUE ((GST_RTCP_VERSION << 14) | GST_RTCP_TYPE_SR)
+
+typedef struct _GstRTCPBuffer GstRTCPBuffer;
+typedef struct _GstRTCPPacket GstRTCPPacket;
+
+struct _GstRTCPBuffer
+{
+  GstBuffer   *buffer;
+  GstMapInfo   map;
+};
+
+#define GST_RTCP_BUFFER_INIT { NULL, GST_MAP_INFO_INIT }
+
+/**
+ * GstRTCPPacket:
+ * @rtcp: pointer to RTCP buffer
+ * @offset: offset of packet in buffer data
+ *
+ * Data structure that points to a packet at @offset in @buffer.
+ * The size of the structure is made public to allow stack allocations.
+ */
+struct _GstRTCPPacket
+{
+  /*< public >*/
+  GstRTCPBuffer *rtcp;
+  guint          offset;
+
+  /*< private >*/
+  gboolean       padding;      /* padding field of current packet */
+  guint8         count;        /* count field of current packet */
+  GstRTCPType    type;         /* type of current packet */
+  guint16        length;       /* length of current packet in 32-bits words minus one, this is validated when doing _get_first_packet() and _move_to_next() */
+
+  guint          item_offset;  /* current item offset for navigating SDES */
+  guint          item_count;   /* current item count */
+  guint          entry_offset; /* current entry offset for navigating SDES items */
+};
+
+/* creating buffers */
+
+GST_RTP_API
+GstBuffer*      gst_rtcp_buffer_new_take_data     (gpointer data, guint len);
+
+GST_RTP_API
+GstBuffer*      gst_rtcp_buffer_new_copy_data     (gconstpointer data, guint len);
+
+GST_RTP_API
+gboolean        gst_rtcp_buffer_validate_data     (guint8 *data, guint len);
+
+GST_RTP_API
+gboolean        gst_rtcp_buffer_validate          (GstBuffer *buffer);
+
+GST_RTP_API
+gboolean        gst_rtcp_buffer_validate_data_reduced   (guint8 *data, guint len);
+
+GST_RTP_API
+gboolean        gst_rtcp_buffer_validate_reduced        (GstBuffer *buffer);
+
+
+GST_RTP_API
+GstBuffer*      gst_rtcp_buffer_new               (guint mtu);
+
+GST_RTP_API
+gboolean        gst_rtcp_buffer_map               (GstBuffer *buffer, GstMapFlags flags, GstRTCPBuffer *rtcp);
+
+GST_RTP_API
+gboolean        gst_rtcp_buffer_unmap             (GstRTCPBuffer *rtcp);
+
+/* adding/retrieving packets */
+
+GST_RTP_API
+guint           gst_rtcp_buffer_get_packet_count  (GstRTCPBuffer *rtcp);
+
+GST_RTP_API
+gboolean        gst_rtcp_buffer_get_first_packet  (GstRTCPBuffer *rtcp, GstRTCPPacket *packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_move_to_next      (GstRTCPPacket *packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_buffer_add_packet        (GstRTCPBuffer *rtcp, GstRTCPType type,
+                                                   GstRTCPPacket *packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_remove            (GstRTCPPacket *packet);
+
+/* working with packets */
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_get_padding       (GstRTCPPacket *packet);
+
+GST_RTP_API
+guint8          gst_rtcp_packet_get_count         (GstRTCPPacket *packet);
+
+GST_RTP_API
+GstRTCPType     gst_rtcp_packet_get_type          (GstRTCPPacket *packet);
+
+GST_RTP_API
+guint16         gst_rtcp_packet_get_length        (GstRTCPPacket *packet);
+
+
+/* sender reports */
+
+GST_RTP_API
+void            gst_rtcp_packet_sr_get_sender_info    (GstRTCPPacket *packet, guint32 *ssrc,
+                                                       guint64 *ntptime, guint32 *rtptime,
+                                                       guint32 *packet_count, guint32 *octet_count);
+
+GST_RTP_API
+void            gst_rtcp_packet_sr_set_sender_info    (GstRTCPPacket *packet, guint32 ssrc,
+                                                       guint64 ntptime, guint32 rtptime,
+                                                       guint32 packet_count, guint32 octet_count);
+/* receiver reports */
+
+GST_RTP_API
+guint32         gst_rtcp_packet_rr_get_ssrc           (GstRTCPPacket *packet);
+
+GST_RTP_API
+void            gst_rtcp_packet_rr_set_ssrc           (GstRTCPPacket *packet, guint32 ssrc);
+
+
+/* report blocks for SR and RR */
+
+GST_RTP_API
+guint           gst_rtcp_packet_get_rb_count          (GstRTCPPacket *packet);
+
+GST_RTP_API
+void            gst_rtcp_packet_get_rb                (GstRTCPPacket *packet, guint nth, guint32 *ssrc,
+                                                       guint8 *fractionlost, gint32 *packetslost,
+                                                       guint32 *exthighestseq, guint32 *jitter,
+                                                       guint32 *lsr, guint32 *dlsr);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_add_rb                (GstRTCPPacket *packet, guint32 ssrc,
+                                                       guint8 fractionlost, gint32 packetslost,
+                                                       guint32 exthighestseq, guint32 jitter,
+                                                       guint32 lsr, guint32 dlsr);
+
+GST_RTP_API
+void            gst_rtcp_packet_set_rb                (GstRTCPPacket *packet, guint nth, guint32 ssrc,
+                                                       guint8 fractionlost, gint32 packetslost,
+                                                       guint32 exthighestseq, guint32 jitter,
+                                                       guint32 lsr, guint32 dlsr);
+
+/* profile-specific extensions for SR and RR */
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_add_profile_specific_ext        (GstRTCPPacket * packet,
+                                                                 const guint8 * data, guint len);
+
+GST_RTP_API
+guint16         gst_rtcp_packet_get_profile_specific_ext_length (GstRTCPPacket * packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_get_profile_specific_ext        (GstRTCPPacket * packet,
+                                                                 guint8 ** data, guint * len);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_copy_profile_specific_ext       (GstRTCPPacket * packet,
+                                                                 guint8 ** data, guint * len);
+
+/* source description packet */
+
+GST_RTP_API
+guint           gst_rtcp_packet_sdes_get_item_count   (GstRTCPPacket *packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_sdes_first_item       (GstRTCPPacket *packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_sdes_next_item        (GstRTCPPacket *packet);
+
+GST_RTP_API
+guint32         gst_rtcp_packet_sdes_get_ssrc         (GstRTCPPacket *packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_sdes_first_entry      (GstRTCPPacket *packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_sdes_next_entry       (GstRTCPPacket *packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_sdes_get_entry        (GstRTCPPacket *packet,
+                                                       GstRTCPSDESType *type, guint8 *len,
+                                                       guint8 **data);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_sdes_copy_entry       (GstRTCPPacket *packet,
+                                                       GstRTCPSDESType *type, guint8 *len,
+                                                       guint8 **data);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_sdes_add_item         (GstRTCPPacket *packet, guint32 ssrc);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_sdes_add_entry        (GstRTCPPacket *packet, GstRTCPSDESType type,
+                                                       guint8 len, const guint8 *data);
+
+/* bye packet */
+
+GST_RTP_API
+guint           gst_rtcp_packet_bye_get_ssrc_count    (GstRTCPPacket *packet);
+
+GST_RTP_API
+guint32         gst_rtcp_packet_bye_get_nth_ssrc      (GstRTCPPacket *packet, guint nth);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_bye_add_ssrc          (GstRTCPPacket *packet, guint32 ssrc);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_bye_add_ssrcs         (GstRTCPPacket *packet, guint32 *ssrc, guint len);
+
+GST_RTP_API
+guint8          gst_rtcp_packet_bye_get_reason_len    (GstRTCPPacket *packet);
+
+GST_RTP_API
+gchar*          gst_rtcp_packet_bye_get_reason        (GstRTCPPacket *packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_bye_set_reason        (GstRTCPPacket *packet, const gchar *reason);
+
+/* app packets */
+
+GST_RTP_API
+void            gst_rtcp_packet_app_set_subtype       (GstRTCPPacket * packet, guint8 subtype);
+
+GST_RTP_API
+guint8          gst_rtcp_packet_app_get_subtype       (GstRTCPPacket * packet);
+
+GST_RTP_API
+void            gst_rtcp_packet_app_set_ssrc          (GstRTCPPacket * packet, guint32 ssrc);
+
+GST_RTP_API
+guint32         gst_rtcp_packet_app_get_ssrc          (GstRTCPPacket * packet);
+
+GST_RTP_API
+void            gst_rtcp_packet_app_set_name          (GstRTCPPacket * packet, const gchar *name);
+
+GST_RTP_API
+const gchar*    gst_rtcp_packet_app_get_name          (GstRTCPPacket * packet);
+
+GST_RTP_API
+guint16         gst_rtcp_packet_app_get_data_length   (GstRTCPPacket * packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_app_set_data_length   (GstRTCPPacket * packet, guint16 wordlen);
+
+GST_RTP_API
+guint8*         gst_rtcp_packet_app_get_data          (GstRTCPPacket * packet);
+
+/* feedback packets */
+
+GST_RTP_API
+guint32         gst_rtcp_packet_fb_get_sender_ssrc    (GstRTCPPacket *packet);
+
+GST_RTP_API
+void            gst_rtcp_packet_fb_set_sender_ssrc    (GstRTCPPacket *packet, guint32 ssrc);
+
+GST_RTP_API
+guint32         gst_rtcp_packet_fb_get_media_ssrc     (GstRTCPPacket *packet);
+
+GST_RTP_API
+void            gst_rtcp_packet_fb_set_media_ssrc     (GstRTCPPacket *packet, guint32 ssrc);
+
+GST_RTP_API
+GstRTCPFBType   gst_rtcp_packet_fb_get_type           (GstRTCPPacket *packet);
+
+GST_RTP_API
+void            gst_rtcp_packet_fb_set_type           (GstRTCPPacket *packet, GstRTCPFBType type);
+
+GST_RTP_API
+guint16         gst_rtcp_packet_fb_get_fci_length     (GstRTCPPacket *packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_fb_set_fci_length     (GstRTCPPacket *packet, guint16 wordlen);
+
+GST_RTP_API
+guint8 *        gst_rtcp_packet_fb_get_fci            (GstRTCPPacket *packet);
+
+/* helper functions */
+
+GST_RTP_API
+guint64         gst_rtcp_ntp_to_unix                  (guint64 ntptime);
+
+GST_RTP_API
+guint64         gst_rtcp_unix_to_ntp                  (guint64 unixtime);
+
+GST_RTP_API
+const gchar *   gst_rtcp_sdes_type_to_name            (GstRTCPSDESType type);
+
+GST_RTP_API
+GstRTCPSDESType gst_rtcp_sdes_name_to_type            (const gchar *name);
+
+/* extended report */
+
+GST_RTP_API
+guint32         gst_rtcp_packet_xr_get_ssrc           (GstRTCPPacket *packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_first_rb           (GstRTCPPacket *packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_next_rb            (GstRTCPPacket * packet);
+
+GST_RTP_API
+GstRTCPXRType   gst_rtcp_packet_xr_get_block_type     (GstRTCPPacket * packet);
+
+GST_RTP_API
+guint16         gst_rtcp_packet_xr_get_block_length   (GstRTCPPacket * packet);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_rle_info       (GstRTCPPacket * packet,
+                                                       guint32 * ssrc, guint8 * thinning,
+                                                       guint16 * begin_seq, guint16 * end_seq,
+                                                       guint32 * chunk_count);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_rle_nth_chunk  (GstRTCPPacket * packet, guint nth,
+                                                       guint16 * chunk);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_prt_info       (GstRTCPPacket * packet,
+                                                       guint32 * ssrc, guint8 * thinning,
+                                                       guint16 * begin_seq, guint16 * end_seq);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_prt_by_seq     (GstRTCPPacket * packet, guint16 seq,
+                                                       guint32 * receipt_time);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_rrt            (GstRTCPPacket * packet, guint64 * timestamp);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_dlrr_block     (GstRTCPPacket * packet,
+                                                       guint nth, guint32 * ssrc,
+                                                       guint32 * last_rr, guint32 * delay);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_summary_info   (GstRTCPPacket * packet, guint32 * ssrc,
+                                                       guint16 * begin_seq, guint16 * end_seq);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_summary_pkt    (GstRTCPPacket * packet,
+                                                       guint32 * lost_packets, guint32 * dup_packets);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_summary_jitter (GstRTCPPacket * packet,
+                                                       guint32 * min_jitter, guint32 * max_jitter,
+                                                       guint32 * mean_jitter, guint32 * dev_jitter);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_summary_ttl    (GstRTCPPacket * packet, gboolean * is_ipv4,
+                                                       guint8 * min_ttl, guint8 * max_ttl,
+                                                       guint8 * mean_ttl, guint8 * dev_ttl);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_voip_metrics_ssrc        (GstRTCPPacket * packet, guint32 * ssrc);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_voip_packet_metrics      (GstRTCPPacket * packet,
+                                                                 guint8 * loss_rate, guint8 * discard_rate);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_voip_burst_metrics       (GstRTCPPacket * packet,
+                                                                 guint8 * burst_density, guint8 * gap_density,
+                                                                 guint16 * burst_duration, guint16 * gap_duration);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_voip_delay_metrics       (GstRTCPPacket * packet,
+                                                                 guint16 * roundtrip_delay,
+                                                                 guint16 * end_system_delay);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_voip_signal_metrics      (GstRTCPPacket * packet,
+                                                                 guint8 * signal_level, guint8 * noise_level,
+                                                                 guint8 * rerl, guint8 * gmin);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_voip_quality_metrics     (GstRTCPPacket * packet,
+                                                                 guint8 * r_factor, guint8 * ext_r_factor,
+                                                                 guint8 * mos_lq, guint8 * mos_cq);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_voip_configuration_params        (GstRTCPPacket * packet,
+                                                                         guint8 * gmin, guint8 * rx_config);
+
+GST_RTP_API
+gboolean        gst_rtcp_packet_xr_get_voip_jitter_buffer_params        (GstRTCPPacket * packet,
+                                                                         guint16 * jb_nominal,
+                                                                         guint16 * jb_maximum,
+                                                                         guint16 * jb_abs_max);
+
+G_END_DECLS
+
+#endif /* __GST_RTCPBUFFER_H__ */
+
diff --git a/include/gst/rtp/gstrtp-enumtypes.h b/include/gst/rtp/gstrtp-enumtypes.h
new file mode 100644
index 0000000000..1cb86d4825
--- /dev/null
+++ b/include/gst/rtp/gstrtp-enumtypes.h
@@ -0,0 +1,64 @@
+
+/* This file is generated by glib-mkenums, do not modify it. This code is licensed under the same license as the containing project. Note that it links to GLib, so must comply with the LGPL linking clauses. */
+
+#pragma once
+
+            #include <glib-object.h>
+            #include <gst/rtp/rtp-prelude.h>
+
+            G_BEGIN_DECLS
+
+/* enumerations from "gstrtcpbuffer.h" */
+
+GST_RTP_API
+GType gst_rtcp_type_get_type (void);
+#define GST_TYPE_RTCP_TYPE (gst_rtcp_type_get_type())
+
+GST_RTP_API
+GType gst_rtcpfb_type_get_type (void);
+#define GST_TYPE_RTCPFB_TYPE (gst_rtcpfb_type_get_type())
+
+GST_RTP_API
+GType gst_rtcpsdes_type_get_type (void);
+#define GST_TYPE_RTCPSDES_TYPE (gst_rtcpsdes_type_get_type())
+
+GST_RTP_API
+GType gst_rtcpxr_type_get_type (void);
+#define GST_TYPE_RTCPXR_TYPE (gst_rtcpxr_type_get_type())
+
+/* enumerations from "gstrtpbuffer.h" */
+
+GST_RTP_API
+GType gst_rtp_buffer_flags_get_type (void);
+#define GST_TYPE_RTP_BUFFER_FLAGS (gst_rtp_buffer_flags_get_type())
+
+GST_RTP_API
+GType gst_rtp_buffer_map_flags_get_type (void);
+#define GST_TYPE_RTP_BUFFER_MAP_FLAGS (gst_rtp_buffer_map_flags_get_type())
+
+/* enumerations from "gstrtpdefs.h" */
+
+GST_RTP_API
+GType gst_rtp_profile_get_type (void);
+#define GST_TYPE_RTP_PROFILE (gst_rtp_profile_get_type())
+
+/* enumerations from "gstrtphdrext.h" */
+
+GST_RTP_API
+GType gst_rtp_header_extension_flags_get_type (void);
+#define GST_TYPE_RTP_HEADER_EXTENSION_FLAGS (gst_rtp_header_extension_flags_get_type())
+
+GST_RTP_API
+GType gst_rtp_header_extension_direction_get_type (void);
+#define GST_TYPE_RTP_HEADER_EXTENSION_DIRECTION (gst_rtp_header_extension_direction_get_type())
+
+/* enumerations from "gstrtppayloads.h" */
+
+GST_RTP_API
+GType gst_rtp_payload_get_type (void);
+#define GST_TYPE_RTP_PAYLOAD (gst_rtp_payload_get_type())
+
+G_END_DECLS
+
+/* Generated data ends here */
+
diff --git a/include/gst/rtp/gstrtpbaseaudiopayload.h b/include/gst/rtp/gstrtpbaseaudiopayload.h
new file mode 100644
index 0000000000..773a49caba
--- /dev/null
+++ b/include/gst/rtp/gstrtpbaseaudiopayload.h
@@ -0,0 +1,124 @@
+/* GStreamer
+ * Copyright (C) <2006> Philippe Khalaf <philippe.kalaf@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTP_BASE_AUDIO_PAYLOAD_H__
+#define __GST_RTP_BASE_AUDIO_PAYLOAD_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstrtpbasepayload.h>
+#include <gst/base/gstadapter.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTPBaseAudioPayload GstRTPBaseAudioPayload;
+typedef struct _GstRTPBaseAudioPayloadClass GstRTPBaseAudioPayloadClass;
+
+typedef struct _GstRTPBaseAudioPayloadPrivate GstRTPBaseAudioPayloadPrivate;
+
+#define GST_TYPE_RTP_BASE_AUDIO_PAYLOAD \
+  (gst_rtp_base_audio_payload_get_type())
+#define GST_RTP_BASE_AUDIO_PAYLOAD(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj), \
+  GST_TYPE_RTP_BASE_AUDIO_PAYLOAD,GstRTPBaseAudioPayload))
+#define GST_RTP_BASE_AUDIO_PAYLOAD_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass), \
+  GST_TYPE_RTP_BASE_AUDIO_PAYLOAD,GstRTPBaseAudioPayloadClass))
+#define GST_IS_RTP_BASE_AUDIO_PAYLOAD(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BASE_AUDIO_PAYLOAD))
+#define GST_IS_RTP_BASE_AUDIO_PAYLOAD_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BASE_AUDIO_PAYLOAD))
+#define GST_RTP_BASE_AUDIO_PAYLOAD_CAST(obj) \
+  ((GstRTPBaseAudioPayload *) (obj))
+
+struct _GstRTPBaseAudioPayload
+{
+  GstRTPBasePayload payload;
+
+  GstRTPBaseAudioPayloadPrivate *priv;
+
+  GstClockTime base_ts;
+  gint frame_size;
+  gint frame_duration;
+
+  gint sample_size;
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTPBaseAudioPayloadClass:
+ * @parent_class: the parent class
+ *
+ * Base class for audio RTP payloader.
+ */
+struct _GstRTPBaseAudioPayloadClass
+{
+  GstRTPBasePayloadClass parent_class;
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTP_API
+GType gst_rtp_base_audio_payload_get_type (void);
+
+/* configure frame based */
+
+GST_RTP_API
+void            gst_rtp_base_audio_payload_set_frame_based        (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
+
+GST_RTP_API
+void            gst_rtp_base_audio_payload_set_frame_options      (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
+                                                                   gint frame_duration, gint frame_size);
+
+/* configure sample based */
+
+GST_RTP_API
+void            gst_rtp_base_audio_payload_set_sample_based       (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
+
+GST_RTP_API
+void            gst_rtp_base_audio_payload_set_sample_options     (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
+                                                                   gint sample_size);
+
+GST_RTP_API
+void            gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
+                                                                   gint sample_size);
+
+/* get the internal adapter */
+
+GST_RTP_API
+GstAdapter*     gst_rtp_base_audio_payload_get_adapter            (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
+
+/* push and flushing data */
+
+GST_RTP_API
+GstFlowReturn   gst_rtp_base_audio_payload_push                   (GstRTPBaseAudioPayload * baseaudiopayload,
+                                                                   const guint8 * data, guint payload_len,
+                                                                   GstClockTime timestamp);
+
+GST_RTP_API
+GstFlowReturn   gst_rtp_base_audio_payload_flush                  (GstRTPBaseAudioPayload * baseaudiopayload,
+                                                                   guint payload_len, GstClockTime timestamp);
+
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTPBaseAudioPayload, gst_object_unref)
+
+G_END_DECLS
+
+#endif /* __GST_RTP_BASE_AUDIO_PAYLOAD_H__ */
diff --git a/include/gst/rtp/gstrtpbasedepayload.h b/include/gst/rtp/gstrtpbasedepayload.h
new file mode 100644
index 0000000000..341a61551c
--- /dev/null
+++ b/include/gst/rtp/gstrtpbasedepayload.h
@@ -0,0 +1,135 @@
+/* GStreamer
+ * Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTP_BASE_DEPAYLOAD_H__
+#define __GST_RTP_BASE_DEPAYLOAD_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTP_BASE_DEPAYLOAD (gst_rtp_base_depayload_get_type())
+#define GST_RTP_BASE_DEPAYLOAD(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_BASE_DEPAYLOAD,GstRTPBaseDepayload))
+#define GST_RTP_BASE_DEPAYLOAD_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_BASE_DEPAYLOAD,GstRTPBaseDepayloadClass))
+#define GST_RTP_BASE_DEPAYLOAD_GET_CLASS(obj) \
+        (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_RTP_BASE_DEPAYLOAD,GstRTPBaseDepayloadClass))
+#define GST_IS_RTP_BASE_DEPAYLOAD(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BASE_DEPAYLOAD))
+#define GST_IS_RTP_BASE_DEPAYLOAD_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BASE_DEPAYLOAD))
+#define GST_RTP_BASE_DEPAYLOAD_CAST(obj) ((GstRTPBaseDepayload *)(obj))
+
+#define GST_RTP_BASE_DEPAYLOAD_SINKPAD(depayload) (GST_RTP_BASE_DEPAYLOAD_CAST (depayload)->sinkpad)
+#define GST_RTP_BASE_DEPAYLOAD_SRCPAD(depayload)  (GST_RTP_BASE_DEPAYLOAD_CAST (depayload)->srcpad)
+
+typedef struct _GstRTPBaseDepayload      GstRTPBaseDepayload;
+typedef struct _GstRTPBaseDepayloadClass GstRTPBaseDepayloadClass;
+typedef struct _GstRTPBaseDepayloadPrivate GstRTPBaseDepayloadPrivate;
+
+struct _GstRTPBaseDepayload
+{
+  GstElement parent;
+
+  GstPad *sinkpad, *srcpad;
+
+  /* this attribute must be set by the child */
+  guint clock_rate;
+
+  GstSegment segment;
+  gboolean need_newsegment;
+
+  /*< private >*/
+  GstRTPBaseDepayloadPrivate *priv;
+
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTPBaseDepayloadClass:
+ * @parent_class: the parent class
+ * @set_caps: configure the depayloader
+ * @process: process incoming rtp packets. Subclass must implement either
+ *   this method or @process_rtp_packet to process incoming rtp packets.
+ *   If the child returns a buffer without a valid timestamp, the timestamp
+ *   of the provided buffer will be applied to the result buffer and the
+ *   buffer will be pushed. If this function returns %NULL, nothing is pushed.
+ * @packet_lost: signal the depayloader about packet loss
+ * @handle_event: custom event handling
+ * @process_rtp_packet: Same as the process virtual function, but slightly more
+ * efficient, since it is passed the rtp buffer structure that has already
+ * been mapped (with GST_MAP_READ) by the base class and thus does not have
+ * to be mapped again by the subclass. Can be used by the subclass to process
+ * incoming rtp packets. If the subclass returns a buffer without a valid
+ * timestamp, the timestamp of the input buffer will be applied to the result
+ * buffer and the output buffer will be pushed out. If this function returns
+ * %NULL, nothing is pushed out. Since: 1.6.
+ *
+ * Base class for RTP depayloaders.
+ */
+struct _GstRTPBaseDepayloadClass
+{
+  GstElementClass parent_class;
+
+  /*< public >*/
+  /* virtuals, inform the subclass of the caps. */
+  gboolean (*set_caps) (GstRTPBaseDepayload *filter, GstCaps *caps);
+
+  /* pure virtual function */
+  GstBuffer * (*process) (GstRTPBaseDepayload *base, GstBuffer *in);
+
+  /* non-pure function used to to signal the depayloader about packet loss. the
+   * timestamp and duration are the estimated values of the lost packet.
+   * The default implementation of this message pushes a segment update. */
+  gboolean (*packet_lost) (GstRTPBaseDepayload *filter, GstEvent *event);
+
+  /* the default implementation does the default actions for events but
+   * implementation can override. */
+  gboolean (*handle_event) (GstRTPBaseDepayload * filter, GstEvent * event);
+
+  GstBuffer * (*process_rtp_packet) (GstRTPBaseDepayload *base, GstRTPBuffer * rtp_buffer);
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING - 1];
+};
+
+GST_RTP_API
+GType gst_rtp_base_depayload_get_type (void);
+
+GST_RTP_API
+GstFlowReturn   gst_rtp_base_depayload_push       (GstRTPBaseDepayload *filter, GstBuffer *out_buf);
+
+GST_RTP_API
+GstFlowReturn   gst_rtp_base_depayload_push_list  (GstRTPBaseDepayload *filter, GstBufferList *out_list);
+
+GST_RTP_API
+gboolean        gst_rtp_base_depayload_is_source_info_enabled  (GstRTPBaseDepayload * depayload);
+
+GST_RTP_API
+void            gst_rtp_base_depayload_set_source_info_enabled (GstRTPBaseDepayload * depayload,
+                                                                gboolean enable);
+
+
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTPBaseDepayload, gst_object_unref)
+
+G_END_DECLS
+
+#endif /* __GST_RTP_BASE_DEPAYLOAD_H__ */
diff --git a/include/gst/rtp/gstrtpbasepayload.h b/include/gst/rtp/gstrtpbasepayload.h
new file mode 100644
index 0000000000..00bf99e196
--- /dev/null
+++ b/include/gst/rtp/gstrtpbasepayload.h
@@ -0,0 +1,199 @@
+/* GStreamer
+ * Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTP_BASE_PAYLOAD_H__
+#define __GST_RTP_BASE_PAYLOAD_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/rtp-prelude.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTP_BASE_PAYLOAD \
+        (gst_rtp_base_payload_get_type())
+#define GST_RTP_BASE_PAYLOAD(obj) \
+        (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_BASE_PAYLOAD,GstRTPBasePayload))
+#define GST_RTP_BASE_PAYLOAD_CLASS(klass) \
+        (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_BASE_PAYLOAD,GstRTPBasePayloadClass))
+#define GST_RTP_BASE_PAYLOAD_GET_CLASS(obj) \
+        (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTP_BASE_PAYLOAD, GstRTPBasePayloadClass))
+#define GST_IS_RTP_BASE_PAYLOAD(obj) \
+        (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BASE_PAYLOAD))
+#define GST_IS_RTP_BASE_PAYLOAD_CLASS(klass) \
+        (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BASE_PAYLOAD))
+#define GST_RTP_BASE_PAYLOAD_CAST(obj) \
+        ((GstRTPBasePayload*)(obj))
+
+typedef struct _GstRTPBasePayload GstRTPBasePayload;
+typedef struct _GstRTPBasePayloadPrivate GstRTPBasePayloadPrivate;
+typedef struct _GstRTPBasePayloadClass GstRTPBasePayloadClass;
+
+/**
+ * GST_RTP_BASE_PAYLOAD_SINKPAD:
+ * @payload: a #GstRTPBasePayload
+ *
+ * Get access to the sinkpad of @payload.
+ */
+#define GST_RTP_BASE_PAYLOAD_SINKPAD(payload) (GST_RTP_BASE_PAYLOAD (payload)->sinkpad)
+/**
+ * GST_RTP_BASE_PAYLOAD_SRCPAD:
+ * @payload: a #GstRTPBasePayload
+ *
+ * Get access to the srcpad of @payload.
+ */
+#define GST_RTP_BASE_PAYLOAD_SRCPAD(payload)  (GST_RTP_BASE_PAYLOAD (payload)->srcpad)
+
+/**
+ * GST_RTP_BASE_PAYLOAD_PT:
+ * @payload: a #GstRTPBasePayload
+ *
+ * Get access to the configured payload type of @payload.
+ */
+#define GST_RTP_BASE_PAYLOAD_PT(payload)  (GST_RTP_BASE_PAYLOAD (payload)->pt)
+/**
+ * GST_RTP_BASE_PAYLOAD_MTU:
+ * @payload: a #GstRTPBasePayload
+ *
+ * Get access to the configured MTU of @payload.
+ */
+#define GST_RTP_BASE_PAYLOAD_MTU(payload) (GST_RTP_BASE_PAYLOAD (payload)->mtu)
+
+struct _GstRTPBasePayload
+{
+  GstElement element;
+
+  /*< private >*/
+  GstPad  *sinkpad;
+  GstPad  *srcpad;
+
+  guint32  ts_base;
+  guint16  seqnum_base;
+
+  gchar   *media;
+  gchar   *encoding_name;
+  gboolean dynamic;
+  guint32  clock_rate;
+
+  gint32   ts_offset;
+  guint32  timestamp;
+  gint16   seqnum_offset;
+  guint16  seqnum;
+  gint64   max_ptime;
+  guint    pt;
+  guint    ssrc;
+  guint    current_ssrc;
+  guint    mtu;
+
+  GstSegment segment;
+
+  guint64  min_ptime;
+  guint64  ptime; /* in ns */
+  guint64  ptime_multiple; /* in ns */
+
+  /*< private >*/
+  GstRTPBasePayloadPrivate *priv;
+
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTPBasePayloadClass:
+ * @parent_class: the parent class
+ * @get_caps: get desired caps
+ * @set_caps: configure the payloader
+ * @handle_buffer: process data
+ * @sink_event: custom event handling on the sinkpad
+ * @src_event: custom event handling on the srcpad
+ * @query: custom query handling
+ *
+ * Base class for audio RTP payloader.
+ */
+struct _GstRTPBasePayloadClass
+{
+  GstElementClass parent_class;
+
+  /* query accepted caps */
+  GstCaps *     (*get_caps)             (GstRTPBasePayload *payload, GstPad * pad, GstCaps * filter);
+  /* receive caps on the sink pad, configure the payloader. */
+  gboolean      (*set_caps)             (GstRTPBasePayload *payload, GstCaps *caps);
+
+  /* handle a buffer, perform 0 or more gst_rtp_base_payload_push() on
+   * the RTP buffers. This function takes ownership of the buffer. */
+  GstFlowReturn (*handle_buffer)        (GstRTPBasePayload *payload,
+                                         GstBuffer *buffer);
+  /* handle events and queries */
+  gboolean      (*sink_event)           (GstRTPBasePayload *payload, GstEvent * event);
+  gboolean      (*src_event)            (GstRTPBasePayload *payload, GstEvent * event);
+  gboolean      (*query)                (GstRTPBasePayload *payload, GstPad *pad, GstQuery * query);
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTP_API
+GType           gst_rtp_base_payload_get_type           (void);
+
+GST_RTP_API
+void            gst_rtp_base_payload_set_options        (GstRTPBasePayload *payload,
+                                                         const gchar *media,
+                                                         gboolean dynamic,
+                                                         const gchar *encoding_name,
+                                                         guint32 clock_rate);
+
+GST_RTP_API
+gboolean        gst_rtp_base_payload_set_outcaps        (GstRTPBasePayload *payload,
+                                                         const gchar *fieldname, ...);
+
+GST_RTP_API
+gboolean        gst_rtp_base_payload_set_outcaps_structure (GstRTPBasePayload *payload,
+                                                            GstStructure *s);
+
+GST_RTP_API
+gboolean        gst_rtp_base_payload_is_filled          (GstRTPBasePayload *payload,
+                                                         guint size, GstClockTime duration);
+
+GST_RTP_API
+GstFlowReturn   gst_rtp_base_payload_push               (GstRTPBasePayload *payload,
+                                                         GstBuffer *buffer);
+
+GST_RTP_API
+GstFlowReturn   gst_rtp_base_payload_push_list          (GstRTPBasePayload *payload,
+                                                         GstBufferList *list);
+
+GST_RTP_API
+GstBuffer *     gst_rtp_base_payload_allocate_output_buffer (GstRTPBasePayload * payload,
+                                                             guint payload_len, guint8 pad_len,
+                                                             guint8 csrc_count);
+
+GST_RTP_API
+void            gst_rtp_base_payload_set_source_info_enabled (GstRTPBasePayload * payload,
+                                                              gboolean enable);
+
+GST_RTP_API
+gboolean        gst_rtp_base_payload_is_source_info_enabled (GstRTPBasePayload * payload);
+
+GST_RTP_API
+guint           gst_rtp_base_payload_get_source_count (GstRTPBasePayload * payload,
+                                                       GstBuffer * buffer);
+
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTPBasePayload, gst_object_unref)
+
+G_END_DECLS
+
+#endif /* __GST_RTP_BASE_PAYLOAD_H__ */
diff --git a/include/gst/rtp/gstrtpbuffer.h b/include/gst/rtp/gstrtpbuffer.h
new file mode 100644
index 0000000000..cac8998c2b
--- /dev/null
+++ b/include/gst/rtp/gstrtpbuffer.h
@@ -0,0 +1,286 @@
+/* GStreamer
+ * Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
+ *               <2005> Wim Taymans <wim@fluendo.com>
+ *
+ * gstrtpbuffer.h: various helper functions to manipulate buffers
+ *     with RTP payload.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTPBUFFER_H__
+#define __GST_RTPBUFFER_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstrtppayloads.h>
+
+G_BEGIN_DECLS
+
+/**
+ * GST_RTP_VERSION:
+ *
+ * The supported RTP version 2.
+ */
+#define GST_RTP_VERSION 2
+
+
+typedef struct _GstRTPBuffer GstRTPBuffer;
+
+/**
+ * GstRTPBuffer:
+ * @buffer: pointer to RTP buffer
+ * @state: internal state
+ * @data: array of data
+ * @size: array of size
+ * @map: array of #GstMapInfo
+ *
+ * Data structure that points to an RTP packet.
+ * The size of the structure is made public to allow stack allocations.
+ */
+struct _GstRTPBuffer
+{
+  GstBuffer   *buffer;
+  guint        state;
+  gpointer     data[4];
+  gsize        size[4];
+  GstMapInfo   map[4];
+};
+
+#define GST_RTP_BUFFER_INIT { NULL, 0, { NULL, NULL, NULL, NULL}, { 0, 0, 0, 0 }, \
+  { GST_MAP_INFO_INIT, GST_MAP_INFO_INIT, GST_MAP_INFO_INIT, GST_MAP_INFO_INIT} }
+
+/* creating buffers */
+
+GST_RTP_API
+void            gst_rtp_buffer_allocate_data         (GstBuffer *buffer, guint payload_len,
+                                                      guint8 pad_len, guint8 csrc_count);
+
+GST_RTP_API
+GstBuffer*      gst_rtp_buffer_new_take_data         (gpointer data, gsize len);
+
+GST_RTP_API
+GstBuffer*      gst_rtp_buffer_new_copy_data         (gconstpointer data, gsize len);
+
+GST_RTP_API
+GstBuffer*      gst_rtp_buffer_new_allocate          (guint payload_len, guint8 pad_len, guint8 csrc_count);
+
+GST_RTP_API
+GstBuffer*      gst_rtp_buffer_new_allocate_len      (guint packet_len, guint8 pad_len, guint8 csrc_count);
+
+GST_RTP_API
+guint           gst_rtp_buffer_calc_header_len       (guint8 csrc_count);
+
+GST_RTP_API
+guint           gst_rtp_buffer_calc_packet_len       (guint payload_len, guint8 pad_len, guint8 csrc_count);
+
+GST_RTP_API
+guint           gst_rtp_buffer_calc_payload_len      (guint packet_len, guint8 pad_len, guint8 csrc_count);
+
+GST_RTP_API
+gboolean        gst_rtp_buffer_map                   (GstBuffer *buffer, GstMapFlags flags, GstRTPBuffer *rtp);
+
+GST_RTP_API
+void            gst_rtp_buffer_unmap                 (GstRTPBuffer *rtp);
+
+GST_RTP_API
+void            gst_rtp_buffer_set_packet_len        (GstRTPBuffer *rtp, guint len);
+
+GST_RTP_API
+guint           gst_rtp_buffer_get_packet_len        (GstRTPBuffer *rtp);
+
+GST_RTP_API
+guint           gst_rtp_buffer_get_header_len        (GstRTPBuffer *rtp);
+
+GST_RTP_API
+guint8          gst_rtp_buffer_get_version           (GstRTPBuffer *rtp);
+
+GST_RTP_API
+void            gst_rtp_buffer_set_version           (GstRTPBuffer *rtp, guint8 version);
+
+GST_RTP_API
+gboolean        gst_rtp_buffer_get_padding           (GstRTPBuffer *rtp);
+
+GST_RTP_API
+void            gst_rtp_buffer_set_padding           (GstRTPBuffer *rtp, gboolean padding);
+
+GST_RTP_API
+void            gst_rtp_buffer_pad_to                (GstRTPBuffer *rtp, guint len);
+
+GST_RTP_API
+gboolean        gst_rtp_buffer_get_extension         (GstRTPBuffer *rtp);
+
+GST_RTP_API
+void            gst_rtp_buffer_set_extension         (GstRTPBuffer *rtp, gboolean extension);
+
+GST_RTP_API
+gboolean        gst_rtp_buffer_get_extension_data    (GstRTPBuffer *rtp, guint16 *bits,
+                                                      gpointer *data, guint *wordlen);
+
+GST_RTP_API
+GBytes*         gst_rtp_buffer_get_extension_bytes   (GstRTPBuffer *rtp, guint16 *bits);
+
+GST_RTP_API
+gboolean        gst_rtp_buffer_set_extension_data    (GstRTPBuffer *rtp, guint16 bits, guint16 length);
+
+GST_RTP_API
+void            gst_rtp_buffer_remove_extension_data (GstRTPBuffer *rtp);
+
+GST_RTP_API
+guint32         gst_rtp_buffer_get_ssrc              (GstRTPBuffer *rtp);
+
+GST_RTP_API
+void            gst_rtp_buffer_set_ssrc              (GstRTPBuffer *rtp, guint32 ssrc);
+
+GST_RTP_API
+guint8          gst_rtp_buffer_get_csrc_count        (GstRTPBuffer *rtp);
+
+GST_RTP_API
+guint32         gst_rtp_buffer_get_csrc              (GstRTPBuffer *rtp, guint8 idx);
+
+GST_RTP_API
+void            gst_rtp_buffer_set_csrc              (GstRTPBuffer *rtp, guint8 idx, guint32 csrc);
+
+GST_RTP_API
+gboolean        gst_rtp_buffer_get_marker            (GstRTPBuffer *rtp);
+
+GST_RTP_API
+void            gst_rtp_buffer_set_marker            (GstRTPBuffer *rtp, gboolean marker);
+
+GST_RTP_API
+guint8          gst_rtp_buffer_get_payload_type      (GstRTPBuffer *rtp);
+
+GST_RTP_API
+void            gst_rtp_buffer_set_payload_type      (GstRTPBuffer *rtp, guint8 payload_type);
+
+GST_RTP_API
+guint16         gst_rtp_buffer_get_seq               (GstRTPBuffer *rtp);
+
+GST_RTP_API
+void            gst_rtp_buffer_set_seq               (GstRTPBuffer *rtp, guint16 seq);
+
+GST_RTP_API
+guint32         gst_rtp_buffer_get_timestamp         (GstRTPBuffer *rtp);
+
+GST_RTP_API
+void            gst_rtp_buffer_set_timestamp         (GstRTPBuffer *rtp, guint32 timestamp);
+
+GST_RTP_API
+GstBuffer*      gst_rtp_buffer_get_payload_buffer    (GstRTPBuffer *rtp);
+
+GST_RTP_API
+GstBuffer*      gst_rtp_buffer_get_payload_subbuffer (GstRTPBuffer *rtp, guint offset, guint len);
+
+GST_RTP_API
+guint           gst_rtp_buffer_get_payload_len       (GstRTPBuffer *rtp);
+
+GST_RTP_API
+gpointer        gst_rtp_buffer_get_payload           (GstRTPBuffer *rtp);
+
+GST_RTP_API
+GBytes*         gst_rtp_buffer_get_payload_bytes     (GstRTPBuffer *rtp);
+
+/* some helpers */
+
+GST_RTP_API
+guint32         gst_rtp_buffer_default_clock_rate    (guint8 payload_type);
+
+GST_RTP_API
+gint            gst_rtp_buffer_compare_seqnum        (guint16 seqnum1, guint16 seqnum2);
+
+GST_RTP_API
+guint64         gst_rtp_buffer_ext_timestamp         (guint64 *exttimestamp, guint32 timestamp);
+
+GST_RTP_API
+gboolean        gst_rtp_buffer_get_extension_onebyte_header  (GstRTPBuffer *rtp,
+                                                              guint8 id,
+                                                              guint nth,
+                                                              gpointer * data,
+                                                              guint * size);
+
+GST_RTP_API
+gboolean        gst_rtp_buffer_get_extension_twobytes_header (GstRTPBuffer *rtp,
+                                                              guint8 * appbits,
+                                                              guint8 id,
+                                                              guint nth,
+                                                              gpointer * data,
+                                                              guint * size);
+
+GST_RTP_API
+gboolean       gst_rtp_buffer_add_extension_onebyte_header  (GstRTPBuffer *rtp,
+                                                             guint8 id,
+                                                             gconstpointer data,
+                                                             guint size);
+
+GST_RTP_API
+gboolean       gst_rtp_buffer_add_extension_twobytes_header (GstRTPBuffer *rtp,
+                                                             guint8 appbits,
+                                                             guint8 id,
+                                                             gconstpointer data,
+                                                             guint size);
+
+GST_RTP_API
+gboolean gst_rtp_buffer_get_extension_onebyte_header_from_bytes (GBytes * bytes,
+                                                                 guint16 bit_pattern,
+                                                                 guint8 id,
+                                                                 guint nth,
+                                                                 gpointer * data,
+                                                                 guint * size);
+
+/**
+ * GstRTPBufferFlags:
+ * @GST_RTP_BUFFER_FLAG_RETRANSMISSION: The #GstBuffer was once wrapped
+ *           in a retransmitted packet as specified by RFC 4588.
+ * @GST_RTP_BUFFER_FLAG_REDUNDANT:      The packet represents redundant RTP packet.
+ *           The flag is used in gstrtpstorage to be able to hold the packetback
+ *           and use it only for recovery from packet loss.
+ *           Since: 1.14
+ * @GST_RTP_BUFFER_FLAG_LAST:           Offset to define more flags.
+ *
+ * Additional RTP buffer flags. These flags can potentially be used on any
+ * buffers carrying RTP packets.
+ *
+ * Note that these are only valid for #GstCaps of type: application/x-rtp (x-rtcp).
+ * They can conflict with other extended buffer flags.
+ *
+ * Since: 1.10
+ */
+typedef enum {
+  GST_RTP_BUFFER_FLAG_RETRANSMISSION = (GST_BUFFER_FLAG_LAST << 0),
+  GST_RTP_BUFFER_FLAG_REDUNDANT      = (GST_BUFFER_FLAG_LAST << 1),
+  GST_RTP_BUFFER_FLAG_LAST           = (GST_BUFFER_FLAG_LAST << 8)
+} GstRTPBufferFlags;
+
+/**
+ * GstRTPBufferMapFlags:
+ * @GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING: Skip mapping and validation of RTP
+ *           padding and RTP pad count when present. Useful for buffers where
+ *           the padding may be encrypted.
+ * @GST_RTP_BUFFER_MAP_FLAG_LAST: Offset to define more flags
+ *
+ * Additional mapping flags for gst_rtp_buffer_map().
+ *
+ * Since: 1.6.1
+ */
+typedef enum {
+  GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING = (GST_MAP_FLAG_LAST << 0),
+  GST_RTP_BUFFER_MAP_FLAG_LAST         = (GST_MAP_FLAG_LAST << 8)
+  /* 8 more flags possible afterwards */
+} GstRTPBufferMapFlags;
+
+G_END_DECLS
+
+#endif /* __GST_RTPBUFFER_H__ */
+
diff --git a/include/gst/rtp/gstrtpdefs.h b/include/gst/rtp/gstrtpdefs.h
new file mode 100644
index 0000000000..14aff5613a
--- /dev/null
+++ b/include/gst/rtp/gstrtpdefs.h
@@ -0,0 +1,58 @@
+/* GStreamer
+ * Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
+ *               <2005> Wim Taymans <wim@fluendo.com>
+ *
+ * gstrtpbuffer.h: various helper functions to manipulate buffers
+ *     with RTP payload.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTPDEFS_H__
+#define __GST_RTPDEFS_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/rtp-prelude.h>
+
+/**
+ * SECTION:gstrtpdefs
+ * @title: GstRTPdefs
+ * @short_description: common RTP defines
+ *
+ * Provides common defines for the RTP library.
+ */
+
+/**
+ * GstRTPProfile:
+ * @GST_RTP_PROFILE_UNKNOWN: invalid profile
+ * @GST_RTP_PROFILE_AVP: the Audio/Visual profile (RFC 3551)
+ * @GST_RTP_PROFILE_SAVP: the secure Audio/Visual profile (RFC 3711)
+ * @GST_RTP_PROFILE_AVPF: the Audio/Visual profile with feedback (RFC 4585)
+ * @GST_RTP_PROFILE_SAVPF: the secure Audio/Visual profile with feedback (RFC 5124)
+ *
+ * The transfer profile to use.
+ *
+ * Since: 1.6
+ */
+typedef enum {
+  GST_RTP_PROFILE_UNKNOWN = 0,
+  GST_RTP_PROFILE_AVP,
+  GST_RTP_PROFILE_SAVP,
+  GST_RTP_PROFILE_AVPF,
+  GST_RTP_PROFILE_SAVPF
+} GstRTPProfile;
+
+#endif /* __GST_RTPDEFS_H__ */
diff --git a/include/gst/rtp/gstrtphdrext.h b/include/gst/rtp/gstrtphdrext.h
new file mode 100644
index 0000000000..442c27af60
--- /dev/null
+++ b/include/gst/rtp/gstrtphdrext.h
@@ -0,0 +1,294 @@
+/* GStreamer
+ * Copyright (C) <2012> Wim Taymans <wim.taymans@gmail.com>
+ * Copyright (C) <2020> Matthew Waters <matthew@centricular.com>
+ *
+ * gstrtphdrext.h: RTP header extensions
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __GST_RTPHDREXT_H__
+#define __GST_RTPHDREXT_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+G_BEGIN_DECLS
+
+#define GST_RTP_HDREXT_BASE "urn:ietf:params:rtp-hdrext:"
+
+/* RFC 6051 */
+#define GST_RTP_HDREXT_NTP_64 "ntp-64"
+
+#define GST_RTP_HDREXT_NTP_64_SIZE 8
+
+GST_RTP_API
+gboolean       gst_rtp_hdrext_set_ntp_64  (gpointer data, guint size, guint64 ntptime);
+
+GST_RTP_API
+gboolean       gst_rtp_hdrext_get_ntp_64  (gpointer data, guint size, guint64 *ntptime);
+
+#define GST_RTP_HDREXT_NTP_56 "ntp-56"
+
+#define GST_RTP_HDREXT_NTP_56_SIZE 7
+
+GST_RTP_API
+gboolean       gst_rtp_hdrext_set_ntp_56  (gpointer data, guint size, guint64 ntptime);
+
+GST_RTP_API
+gboolean       gst_rtp_hdrext_get_ntp_56  (gpointer data, guint size, guint64 *ntptime);
+
+/**
+ * GST_RTP_HDREXT_ELEMENT_CLASS:
+ *
+ * Constant string used in element classification to signal that this element
+ * is a RTP header extension.
+ *
+ * Since: 1.20
+ */
+#define GST_RTP_HDREXT_ELEMENT_CLASS "Network/Extension/RTPHeader"
+
+GST_RTP_API
+GType gst_rtp_header_extension_get_type (void);
+#define GST_TYPE_RTP_HEADER_EXTENSION (gst_rtp_header_extension_get_type())
+#define GST_RTP_HEADER_EXTENSION(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_HEADER_EXTENSION,GstRTPHeaderExtension))
+#define GST_RTP_HEADER_EXTENSION_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_HEADER_EXTENSION,GstRTPHeaderExtensionClass))
+#define GST_RTP_HEADER_EXTENSION_GET_CLASS(obj) \
+        (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_RTP_HEADER_EXTENSION,GstRTPHeaderExtensionClass))
+#define GST_IS_RTP_HEADER_EXTENSION(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_HEADER_EXTENSION))
+#define GST_IS_RTP_HEADER_EXTENSION_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_HEADER_EXTENSION))
+/**
+ * GST_RTP_HEADER_EXTENSION_CAST:
+ *
+ * Since: 1.20
+ */
+#define GST_RTP_HEADER_EXTENSION_CAST(obj) ((GstRTPHeaderExtension *)(obj))
+
+typedef struct _GstRTPHeaderExtension      GstRTPHeaderExtension;
+typedef struct _GstRTPHeaderExtensionClass GstRTPHeaderExtensionClass;
+
+/**
+ * GstRTPHeaderExtensionFlags:
+ * @GST_RTP_HEADER_EXTENSION_ONE_BYTE: The one byte rtp extension header.
+ *              1-16 data bytes per extension with a maximum of
+ *              14 extension ids in total.
+ * @GST_RTP_HEADER_EXTENSION_TWO_BYTE: The two byte rtp extension header.
+ *              256 data bytes per extension with a maximum of 255 (or 256
+ *              including appbits) extensions in total.
+ *
+ * Flags that apply to a RTP Audio/Video header extension.
+ *
+ * Since: 1.20
+ */
+typedef enum /*< underscore_name=gst_rtp_header_extension_flags >*/
+{
+  GST_RTP_HEADER_EXTENSION_ONE_BYTE = (1 << 0),
+  GST_RTP_HEADER_EXTENSION_TWO_BYTE = (1 << 1),
+} GstRTPHeaderExtensionFlags;
+
+/**
+ * GstRTPHeaderExtensionDirection:
+ * @GST_RTP_HEADER_EXTENSION_DIRECTION_INACTIVE: Neither send nor
+ * receive RTP Header Extensions
+ * @GST_RTP_HEADER_EXTENSION_DIRECTION_SENDONLY: Only send RTP Header
+ * Extensions @GST_RTP_HEADER_EXTENSION_DIRECTION_RECVONLY: Only
+ * receive RTP Header Extensions
+ * @GST_RTP_HEADER_EXTENSION_DIRECTION_SENDRECV: Send and receive RTP
+ * Header Extensions ext
+ * @GST_RTP_HEADER_EXTENSION_DIRECTION_INHERITED: RTP header extension
+ * direction is inherited from the stream
+ *
+ * Direction to which to apply the RTP Header Extension
+ *
+ * Since: 1.20
+ */
+typedef enum /*< underscore_name=gst_rtp_header_extension_direction >*/
+{
+  GST_RTP_HEADER_EXTENSION_DIRECTION_INACTIVE = 0,
+  GST_RTP_HEADER_EXTENSION_DIRECTION_SENDONLY = (1 << 0),
+  GST_RTP_HEADER_EXTENSION_DIRECTION_RECVONLY = (1 << 1),
+  GST_RTP_HEADER_EXTENSION_DIRECTION_SENDRECV = (
+    GST_RTP_HEADER_EXTENSION_DIRECTION_SENDONLY |
+    GST_RTP_HEADER_EXTENSION_DIRECTION_RECVONLY),
+  GST_RTP_HEADER_EXTENSION_DIRECTION_INHERITED = (1 << 2)
+} GstRTPHeaderExtensionDirection;
+
+/**
+ * GstRTPHeaderExtension:
+ * @parent: the parent #GObject
+ * @ext_id: the configured extension id
+ *
+ * Instance struct for a RTP Audio/Video header extension.
+ *
+ * Since: 1.20
+ */
+struct _GstRTPHeaderExtension
+{
+  GstElement parent;
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTPHeaderExtensionClass:
+ * @parent_class: the parent class
+ * @get_uri: retrieve the RTP extension uri
+ * @get_supported_flags: retrieve the supported flags
+ * @get_max_size: retrieve the maximum size for this extension based on the
+ *     information available from input_meta.  Implementations should attempt
+ *     to provide as accurate information as possible as the returned value
+ *     will be used to control the amount of possible data in the payload.
+ *     Implementations must return the maximum as the allocated size for
+ *     writing the extension will be at least the size of the returned value.
+ *     Return the amount of data read or <0 on failure.
+ * @write: write into @data the information for this extension.  Various
+ *     information is provided to help writing extensions in particular cases.
+ * @read: read from a rtp payloaded buffer and extract the extension
+ *     information, optionally adding some meta onto the output buffer.
+ * @set_non_rtp_sink_caps: read any information from sink caps that the header
+ *     extension needs for its function.
+ * @update_non_rtp_src_caps: update depayloader non-RTP (depayloaded) caps with
+ *     the information parsed from RTP header.
+ * @set_attributes: set the necessary attributes that may be signaled e.g. with
+ *     an SDP.
+ * @set_caps_from_attributes: write the necessary caps field/s for the configured
+ *     attributes e.g. as signalled with SDP.
+ *
+ * Base class for RTP Header extensions.
+ *
+ * Since: 1.20
+ */
+
+struct _GstRTPHeaderExtensionClass
+{
+  GstElementClass parent_class;
+
+  /*< public >*/
+  GstRTPHeaderExtensionFlags (*get_supported_flags) (GstRTPHeaderExtension * ext);
+
+  gsize                 (*get_max_size)             (GstRTPHeaderExtension * ext,
+                                                     const GstBuffer * input_meta);
+
+  gssize                (*write)                    (GstRTPHeaderExtension * ext,
+                                                     const GstBuffer * input_meta,
+                                                     GstRTPHeaderExtensionFlags write_flags,
+                                                     GstBuffer * output,
+                                                     guint8 * data,
+                                                     gsize size);
+
+  gboolean              (*read)                     (GstRTPHeaderExtension * ext,
+                                                     GstRTPHeaderExtensionFlags read_flags,
+                                                     const guint8 * data,
+                                                     gsize size,
+                                                     GstBuffer * buffer);
+  gboolean              (*set_non_rtp_sink_caps)    (GstRTPHeaderExtension * ext,
+                                                     const GstCaps * caps);
+  gboolean              (*update_non_rtp_src_caps)  (GstRTPHeaderExtension * ext,
+                                                     GstCaps * caps);
+  gboolean              (*set_attributes)            (GstRTPHeaderExtension * ext,
+                                                      GstRTPHeaderExtensionDirection direction,
+                                                      const gchar * attributes);
+  gboolean              (*set_caps_from_attributes) (GstRTPHeaderExtension * ext,
+                                                     GstCaps * caps);
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTPHeaderExtension, gst_object_unref)
+
+/**
+ * GST_RTP_HEADER_EXTENSION_URI_METADATA_KEY:
+ *
+ * Since: 1.20
+ */
+#define GST_RTP_HEADER_EXTENSION_URI_METADATA_KEY "RTP-Header-Extension-URI"
+
+GST_RTP_API
+void                gst_rtp_header_extension_class_set_uri      (GstRTPHeaderExtensionClass *klass,
+                                                                 const gchar * uri);
+
+GST_RTP_API
+const gchar *       gst_rtp_header_extension_get_uri            (GstRTPHeaderExtension * ext);
+GST_RTP_API
+gsize               gst_rtp_header_extension_get_max_size       (GstRTPHeaderExtension * ext,
+                                                                 const GstBuffer * input_meta);
+GST_RTP_API
+GstRTPHeaderExtensionFlags gst_rtp_header_extension_get_supported_flags (GstRTPHeaderExtension * ext);
+GST_RTP_API
+guint               gst_rtp_header_extension_get_id             (GstRTPHeaderExtension * ext);
+GST_RTP_API
+void                gst_rtp_header_extension_set_id             (GstRTPHeaderExtension * ext,
+                                                                 guint ext_id);
+GST_RTP_API
+gssize              gst_rtp_header_extension_write              (GstRTPHeaderExtension * ext,
+                                                                 const GstBuffer * input_meta,
+                                                                 GstRTPHeaderExtensionFlags write_flags,
+                                                                 GstBuffer * output,
+                                                                 guint8 * data,
+                                                                 gsize size);
+GST_RTP_API
+gboolean            gst_rtp_header_extension_read               (GstRTPHeaderExtension * ext,
+                                                                 GstRTPHeaderExtensionFlags read_flags,
+                                                                 const guint8 * data,
+                                                                 gsize size,
+                                                                 GstBuffer * buffer);
+GST_RTP_API
+gboolean            gst_rtp_header_extension_set_non_rtp_sink_caps (GstRTPHeaderExtension * ext,
+                                                                    const GstCaps * caps);
+GST_RTP_API
+gboolean            gst_rtp_header_extension_wants_update_non_rtp_src_caps (GstRTPHeaderExtension * ext);
+GST_RTP_API
+void                gst_rtp_header_extension_set_wants_update_non_rtp_src_caps (GstRTPHeaderExtension * ext,
+                                                                                gboolean state);
+GST_RTP_API
+gboolean            gst_rtp_header_extension_update_non_rtp_src_caps (GstRTPHeaderExtension * ext,
+                                                                      GstCaps * caps);
+GST_RTP_API
+gboolean            gst_rtp_header_extension_set_caps_from_attributes (GstRTPHeaderExtension * ext,
+                                                                       GstCaps * caps);
+GST_RTP_API
+gboolean            gst_rtp_header_extension_set_attributes_from_caps (GstRTPHeaderExtension * ext,
+                                                                       const GstCaps * caps);
+
+GST_RTP_API
+GList *             gst_rtp_get_header_extension_list           (void);
+GST_RTP_API
+GstRTPHeaderExtension * gst_rtp_header_extension_create_from_uri (const gchar * uri);
+
+GST_RTP_API
+gchar *	            gst_rtp_header_extension_get_sdp_caps_field_name (GstRTPHeaderExtension * ext);
+
+GST_RTP_API
+void	            gst_rtp_header_extension_set_direction (GstRTPHeaderExtension * ext,
+                                                            GstRTPHeaderExtensionDirection direction);
+GST_RTP_API
+GstRTPHeaderExtensionDirection	gst_rtp_header_extension_get_direction (GstRTPHeaderExtension * ext);
+
+GST_RTP_API
+gboolean    gst_rtp_header_extension_set_caps_from_attributes_helper (GstRTPHeaderExtension * ext,
+                                                                      GstCaps * caps,
+                                                                      const gchar * attributes);
+
+G_END_DECLS
+
+#endif /* __GST_RTPHDREXT_H__ */
+
diff --git a/include/gst/rtp/gstrtpmeta.h b/include/gst/rtp/gstrtpmeta.h
new file mode 100644
index 0000000000..f0611d36e3
--- /dev/null
+++ b/include/gst/rtp/gstrtpmeta.h
@@ -0,0 +1,79 @@
+/* GStreamer
+ * Copyright (C) <2016> Stian Selnes <stian@pexip.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTP_META_H__
+#define __GST_RTP_META_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/rtp-prelude.h>
+
+G_BEGIN_DECLS
+
+#define GST_RTP_SOURCE_META_API_TYPE  (gst_rtp_source_meta_api_get_type())
+#define GST_RTP_SOURCE_META_INFO  (gst_rtp_source_meta_get_info())
+typedef struct _GstRTPSourceMeta GstRTPSourceMeta;
+
+#define GST_RTP_SOURCE_META_MAX_CSRC_COUNT 15
+
+/**
+ * GstRTPSourceMeta:
+ * @meta: parent #GstMeta
+ * @ssrc: the SSRC
+ * @ssrc_valid: whether @ssrc is set and valid
+ * @csrc: (allow-none): pointer to the CSRCs
+ * @csrc_count: number of elements in @csrc
+ *
+ * Meta describing the source(s) of the buffer.
+ *
+ * Since: 1.16
+ */
+struct _GstRTPSourceMeta
+{
+  GstMeta meta;
+
+  guint32 ssrc;
+  gboolean ssrc_valid;
+  guint32 csrc[GST_RTP_SOURCE_META_MAX_CSRC_COUNT];
+  guint csrc_count;
+};
+
+GST_RTP_API
+GType               gst_rtp_source_meta_api_get_type     (void);
+
+GST_RTP_API
+GstRTPSourceMeta *  gst_buffer_add_rtp_source_meta       (GstBuffer * buffer, const guint32 * ssrc,
+                                                          const guint32 * csrc, guint csrc_count);
+GST_RTP_API
+GstRTPSourceMeta *  gst_buffer_get_rtp_source_meta       (GstBuffer * buffer);
+
+GST_RTP_API
+guint               gst_rtp_source_meta_get_source_count (const GstRTPSourceMeta * meta);
+
+GST_RTP_API
+gboolean            gst_rtp_source_meta_set_ssrc         (GstRTPSourceMeta * meta, guint32 * ssrc);
+
+GST_RTP_API
+gboolean            gst_rtp_source_meta_append_csrc      (GstRTPSourceMeta * meta,
+                                                          const guint32 * csrc, guint csrc_count);
+GST_RTP_API
+const GstMetaInfo * gst_rtp_source_meta_get_info         (void);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_META_H__ */
diff --git a/include/gst/rtp/gstrtppayloads.h b/include/gst/rtp/gstrtppayloads.h
new file mode 100644
index 0000000000..84f6c3cc3c
--- /dev/null
+++ b/include/gst/rtp/gstrtppayloads.h
@@ -0,0 +1,199 @@
+/* GStreamer
+ * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * gstrtppayloads.h: various helper functions to deal with RTP payload
+ *     types.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTPPAYLOADS_H__
+#define __GST_RTPPAYLOADS_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/rtp-prelude.h>
+
+G_BEGIN_DECLS
+
+/**
+ * GstRTPPayload:
+ * @GST_RTP_PAYLOAD_PCMU: ITU-T G.711. mu-law audio (RFC 3551)
+ * @GST_RTP_PAYLOAD_1016: RFC 3551 says reserved
+ * @GST_RTP_PAYLOAD_G721: RFC 3551 says reserved
+ * @GST_RTP_PAYLOAD_GSM: GSM audio
+ * @GST_RTP_PAYLOAD_G723: ITU G.723.1 audio
+ * @GST_RTP_PAYLOAD_DVI4_8000: IMA ADPCM wave type (RFC 3551)
+ * @GST_RTP_PAYLOAD_DVI4_16000: IMA ADPCM wave type (RFC 3551)
+ * @GST_RTP_PAYLOAD_LPC: experimental linear predictive encoding
+ * @GST_RTP_PAYLOAD_PCMA: ITU-T G.711 A-law audio (RFC 3551)
+ * @GST_RTP_PAYLOAD_G722: ITU-T G.722 (RFC 3551)
+ * @GST_RTP_PAYLOAD_L16_STEREO: stereo PCM
+ * @GST_RTP_PAYLOAD_L16_MONO: mono PCM
+ * @GST_RTP_PAYLOAD_QCELP: EIA & TIA standard IS-733
+ * @GST_RTP_PAYLOAD_CN: Comfort Noise (RFC 3389)
+ * @GST_RTP_PAYLOAD_MPA: Audio MPEG 1-3.
+ * @GST_RTP_PAYLOAD_G728: ITU-T G.728 Speech coder (RFC 3551)
+ * @GST_RTP_PAYLOAD_DVI4_11025: IMA ADPCM wave type (RFC 3551)
+ * @GST_RTP_PAYLOAD_DVI4_22050: IMA ADPCM wave type (RFC 3551)
+ * @GST_RTP_PAYLOAD_G729: ITU-T G.729 Speech coder (RFC 3551)
+ * @GST_RTP_PAYLOAD_CELLB: See RFC 2029
+ * @GST_RTP_PAYLOAD_JPEG: ISO Standards 10918-1 and 10918-2 (RFC 2435)
+ * @GST_RTP_PAYLOAD_NV: nv encoding by Ron Frederick
+ * @GST_RTP_PAYLOAD_H261: ITU-T Recommendation H.261 (RFC 2032)
+ * @GST_RTP_PAYLOAD_MPV: Video MPEG 1 & 2 (RFC 2250)
+ * @GST_RTP_PAYLOAD_MP2T: MPEG-2 transport stream (RFC 2250)
+ * @GST_RTP_PAYLOAD_H263: Video H263 (RFC 2190)
+ *
+ * Standard predefined fixed payload types.
+ *
+ * The official list is at:
+ * http://www.iana.org/assignments/rtp-parameters
+ *
+ * Audio:
+ * reserved: 19
+ * unassigned: 20-23,
+ *
+ * Video:
+ * unassigned: 24, 27, 29, 30, 35-71, 77-95
+ * Reserved for RTCP conflict avoidance: 72-76
+ */
+typedef enum
+{
+  /* Audio: */
+  GST_RTP_PAYLOAD_PCMU = 0,
+  GST_RTP_PAYLOAD_1016 = 1, /* RFC 3551 says reserved */
+  GST_RTP_PAYLOAD_G721 = 2, /* RFC 3551 says reserved */
+  GST_RTP_PAYLOAD_GSM = 3,
+  GST_RTP_PAYLOAD_G723 = 4,
+  GST_RTP_PAYLOAD_DVI4_8000 = 5,
+  GST_RTP_PAYLOAD_DVI4_16000 = 6,
+  GST_RTP_PAYLOAD_LPC = 7,
+  GST_RTP_PAYLOAD_PCMA = 8,
+  GST_RTP_PAYLOAD_G722 = 9,
+  GST_RTP_PAYLOAD_L16_STEREO = 10,
+  GST_RTP_PAYLOAD_L16_MONO = 11,
+  GST_RTP_PAYLOAD_QCELP = 12,
+  GST_RTP_PAYLOAD_CN = 13,
+  GST_RTP_PAYLOAD_MPA = 14,
+  GST_RTP_PAYLOAD_G728 = 15,
+  GST_RTP_PAYLOAD_DVI4_11025 = 16,
+  GST_RTP_PAYLOAD_DVI4_22050 = 17,
+  GST_RTP_PAYLOAD_G729 = 18,
+
+  /* Video: */
+
+  GST_RTP_PAYLOAD_CELLB = 25,
+  GST_RTP_PAYLOAD_JPEG = 26,
+  GST_RTP_PAYLOAD_NV = 28,
+  GST_RTP_PAYLOAD_H261 = 31,
+  GST_RTP_PAYLOAD_MPV = 32,
+  GST_RTP_PAYLOAD_MP2T = 33,
+  GST_RTP_PAYLOAD_H263 = 34,
+
+  /* BOTH */
+} GstRTPPayload;
+
+/* backward compatibility */
+#define GST_RTP_PAYLOAD_G723_63 16
+#define GST_RTP_PAYLOAD_G723_53 17
+#define GST_RTP_PAYLOAD_TS48 18
+#define GST_RTP_PAYLOAD_TS41 19
+
+#define GST_RTP_PAYLOAD_G723_63_STRING "16"
+#define GST_RTP_PAYLOAD_G723_53_STRING "17"
+#define GST_RTP_PAYLOAD_TS48_STRING "18"
+#define GST_RTP_PAYLOAD_TS41_STRING "19"
+
+/* Defining the above as strings, to make the declaration of pad_templates
+ * easier. So if please keep these synchronized with the above.
+ */
+#define GST_RTP_PAYLOAD_PCMU_STRING "0"
+#define GST_RTP_PAYLOAD_1016_STRING "1"
+#define GST_RTP_PAYLOAD_G721_STRING "2"
+#define GST_RTP_PAYLOAD_GSM_STRING "3"
+#define GST_RTP_PAYLOAD_G723_STRING "4"
+#define GST_RTP_PAYLOAD_DVI4_8000_STRING "5"
+#define GST_RTP_PAYLOAD_DVI4_16000_STRING "6"
+#define GST_RTP_PAYLOAD_LPC_STRING "7"
+#define GST_RTP_PAYLOAD_PCMA_STRING "8"
+#define GST_RTP_PAYLOAD_G722_STRING "9"
+#define GST_RTP_PAYLOAD_L16_STEREO_STRING "10"
+#define GST_RTP_PAYLOAD_L16_MONO_STRING "11"
+#define GST_RTP_PAYLOAD_QCELP_STRING "12"
+#define GST_RTP_PAYLOAD_CN_STRING "13"
+#define GST_RTP_PAYLOAD_MPA_STRING "14"
+#define GST_RTP_PAYLOAD_G728_STRING "15"
+#define GST_RTP_PAYLOAD_DVI4_11025_STRING "16"
+#define GST_RTP_PAYLOAD_DVI4_22050_STRING "17"
+#define GST_RTP_PAYLOAD_G729_STRING "18"
+
+#define GST_RTP_PAYLOAD_CELLB_STRING "25"
+#define GST_RTP_PAYLOAD_JPEG_STRING "26"
+#define GST_RTP_PAYLOAD_NV_STRING "28"
+
+#define GST_RTP_PAYLOAD_H261_STRING "31"
+#define GST_RTP_PAYLOAD_MPV_STRING "32"
+#define GST_RTP_PAYLOAD_MP2T_STRING "33"
+#define GST_RTP_PAYLOAD_H263_STRING "34"
+
+#define GST_RTP_PAYLOAD_DYNAMIC_STRING "[96, 127]"
+
+/**
+ * GST_RTP_PAYLOAD_IS_DYNAMIC:
+ * @pt: a payload type
+ *
+ * Check if @pt is a dynamic payload type.
+ */
+#define GST_RTP_PAYLOAD_IS_DYNAMIC(pt) ((pt) >= 96 && (pt) <= 127)
+
+typedef struct _GstRTPPayloadInfo GstRTPPayloadInfo;
+
+/**
+ * GstRTPPayloadInfo:
+ * @payload_type: payload type, -1 means dynamic
+ * @media: the media type(s), usually "audio", "video", "application", "text",
+ * "message".
+ * @encoding_name: the encoding name of @pt
+ * @clock_rate: default clock rate, 0 = unknown/variable
+ * @encoding_parameters: encoding parameters. For audio this is the number of
+ * channels. NULL = not applicable.
+ * @bitrate: the bitrate of the media. 0 = unknown/variable.
+ *
+ * Structure holding default payload type information.
+ */
+struct _GstRTPPayloadInfo
+{
+  guint8       payload_type;
+  const gchar *media;
+  const gchar *encoding_name;
+  guint        clock_rate;
+  const gchar *encoding_parameters;
+  guint        bitrate;
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+GST_RTP_API
+const GstRTPPayloadInfo *  gst_rtp_payload_info_for_pt     (guint8 payload_type);
+
+GST_RTP_API
+const GstRTPPayloadInfo *  gst_rtp_payload_info_for_name   (const gchar *media, const gchar *encoding_name);
+
+G_END_DECLS
+
+#endif /* __GST_RTPPAYLOADS_H__ */
+
diff --git a/include/gst/rtp/rtp-prelude.h b/include/gst/rtp/rtp-prelude.h
new file mode 100644
index 0000000000..e130b8a476
--- /dev/null
+++ b/include/gst/rtp/rtp-prelude.h
@@ -0,0 +1,33 @@
+/* GStreamer RTP Library
+ * Copyright (C) 2018 GStreamer developers
+ *
+ * rtp-prelude.h: prelude include header for gst-rtp library
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTP_PRELUDE_H__
+#define __GST_RTP_PRELUDE_H__
+
+#include <gst/gst.h>
+
+#ifdef BUILDING_GST_RTP
+#define GST_RTP_API GST_API_EXPORT         /* from config.h */
+#else
+#define GST_RTP_API GST_API_IMPORT
+#endif
+
+#endif /* __GST_RTP_PRELUDE_H__ */
diff --git a/include/gst/rtp/rtp.h b/include/gst/rtp/rtp.h
new file mode 100644
index 0000000000..0e6633bd81
--- /dev/null
+++ b/include/gst/rtp/rtp.h
@@ -0,0 +1,36 @@
+/* GStreamer
+ * Copyright (C) 2012 GStreamer developers
+ *
+ * gstrtp.h: single include header for gst-rtp library
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTP_H__
+#define __GST_RTP_H__
+
+#include <gst/rtp/gstrtpdefs.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/rtp/gstrtcpbuffer.h>
+#include <gst/rtp/gstrtppayloads.h>
+#include <gst/rtp/gstrtphdrext.h>
+#include <gst/rtp/gstrtpbaseaudiopayload.h>
+#include <gst/rtp/gstrtpbasepayload.h>
+#include <gst/rtp/gstrtpbasedepayload.h>
+#include <gst/rtp/gstrtpmeta.h>
+#include <gst/rtp/gstrtp-enumtypes.h>
+
+#endif /* __GST_RTP_H__ */