Video optimizations

8 files changed, 1014 insertions, 116 deletions

diff --git a/plugins/rtp/CMakeLists.txt b/plugins/rtp/CMakeLists.txt
index 76d6e66d..92ec1b97 100644
--- a/plugins/rtp/CMakeLists.txt
+++ b/plugins/rtp/CMakeLists.txt
@@ -1,3 +1,4 @@
+find_package(GstRtp REQUIRED)
 find_packages(RTP_PACKAGES REQUIRED
     Gee
     GLib
@@ -27,6 +28,7 @@ CUSTOM_VAPIS
     ${CMAKE_BINARY_DIR}/exports/xmpp-vala.vapi
     ${CMAKE_BINARY_DIR}/exports/dino.vapi
     ${CMAKE_BINARY_DIR}/exports/qlite.vapi
+    ${CMAKE_CURRENT_SOURCE_DIR}/vapi/gstreamer-rtp-1.0.vapi
 PACKAGES
     ${RTP_PACKAGES}
 DEFINITIONS
@@ -35,7 +37,7 @@ DEFINITIONS
 
 add_definitions(${VALA_CFLAGS} -DG_LOG_DOMAIN="rtp" -I${CMAKE_CURRENT_SOURCE_DIR}/src)
 add_library(rtp SHARED ${RTP_VALA_C})
-target_link_libraries(rtp libdino crypto-vala ${RTP_PACKAGES})
+target_link_libraries(rtp libdino crypto-vala ${RTP_PACKAGES} gstreamer-rtp-1.0)
 set_target_properties(rtp PROPERTIES PREFIX "")
 set_target_properties(rtp PROPERTIES LIBRARY_OUTPUT_DIRECTORY ${CMAKE_BINARY_DIR}/plugins/)
 
diff --git a/plugins/rtp/src/codec_util.vala b/plugins/rtp/src/codec_util.vala
index 6bd465c1..7537c11d 100644
--- a/plugins/rtp/src/codec_util.vala
+++ b/plugins/rtp/src/codec_util.vala
@@ -6,7 +6,7 @@ public class Dino.Plugins.Rtp.CodecUtil {
     private Set<string> supported_elements = new HashSet<string>();
     private Set<string> unsupported_elements = new HashSet<string>();
 
-    public static Gst.Caps get_caps(string media, JingleRtp.PayloadType payload_type) {
+    public static Gst.Caps get_caps(string media, JingleRtp.PayloadType payload_type, bool incoming) {
         Gst.Caps caps = new Gst.Caps.simple("application/x-rtp",
                 "media", typeof(string), media,
                 "payload", typeof(int), payload_type.id);
@@ -19,6 +19,15 @@ public class Dino.Plugins.Rtp.CodecUtil {
         if (payload_type.name != null) {
             s.set("encoding-name", typeof(string), payload_type.name.up());
         }
+        if (incoming) {
+            foreach (JingleRtp.RtcpFeedback rtcp_fb in payload_type.rtcp_fbs) {
+                if (rtcp_fb.subtype == null) {
+                    s.set(@"rtcp-fb-$(rtcp_fb.type_)", typeof(bool), true);
+                } else {
+                    s.set(@"rtcp-fb-$(rtcp_fb.type_)-$(rtcp_fb.subtype)", typeof(bool), true);
+                }
+            }
+        }
         return caps;
     }
 
@@ -122,32 +131,82 @@ public class Dino.Plugins.Rtp.CodecUtil {
         return new string[0];
     }
 
-    public static string? get_encode_prefix(string media, string codec, string encode) {
+    public static string? get_encode_prefix(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) {
         if (encode == "msdkh264enc") return "video/x-raw,format=NV12 ! ";
         if (encode == "vaapih264enc") return "video/x-raw,format=NV12 ! ";
         return null;
     }
 
-    public static string? get_encode_suffix(string media, string codec, string encode) {
+    public static string? get_encode_args(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) {
         // H264
-        const string h264_suffix = " ! video/x-h264,profile=constrained-baseline ! h264parse";
-        if (encode == "msdkh264enc") return @" bitrate=256 rate-control=vbr target-usage=7$h264_suffix";
-        if (encode == "vaapih264enc") return @" bitrate=256 quality-level=7 tune=low-power$h264_suffix";
-        if (encode == "x264enc") return @" byte-stream=1 bitrate=256 profile=baseline speed-preset=ultrafast tune=zerolatency$h264_suffix";
-        if (media == "video" && codec == "h264") return h264_suffix;
+        if (encode == "msdkh264enc") return @" rate-control=vbr";
+        if (encode == "vaapih264enc") return @" tune=low-power";
+        if (encode == "x264enc") return @" byte-stream=1 profile=baseline speed-preset=ultrafast tune=zerolatency";
 
         // VP8
-        if (encode == "msdkvp8enc") return " bitrate=256 rate-control=vbr target-usage=7";
-        if (encode == "vaapivp8enc") return " bitrate=256 rate-control=vbr quality-level=7";
-        if (encode == "vp8enc") return " target-bitrate=256000 deadline=1 error-resilient=1";
+        if (encode == "msdkvp8enc") return " rate-control=vbr";
+        if (encode == "vaapivp8enc") return " rate-control=vbr";
+        if (encode == "vp8enc") return " deadline=1 error-resilient=1";
 
         // OPUS
-        if (encode == "opusenc") return " audio-type=voice";
+        if (encode == "opusenc") {
+            if (payload_type != null && payload_type.parameters.has("useinbandfec", "1")) return " audio-type=voice inband-fec=true";
+            return " audio-type=voice";
+        }
 
         return null;
     }
 
-    public static string? get_decode_prefix(string media, string codec, string decode) {
+    public static string? get_encode_suffix(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) {
+        // H264
+        if (media == "video" && codec == "h264") return " ! video/x-h264,profile=constrained-baseline ! h264parse";
+        return null;
+    }
+
+    public uint update_bitrate(string media, JingleRtp.PayloadType payload_type, Gst.Element encode_element, uint bitrate) {
+        Gst.Bin? encode_bin = encode_element as Gst.Bin;
+        if (encode_bin == null) return 0;
+        string? codec = get_codec_from_payload(media, payload_type);
+        string? encode_name = get_encode_element_name(media, codec);
+        if (encode_name == null) return 0;
+        Gst.Element encode = encode_bin.get_by_name(@"$(encode_bin.name)_encode");
+
+        bitrate = uint.min(2048000, bitrate);
+
+        switch (encode_name) {
+            case "msdkh264enc":
+            case "vaapih264enc":
+            case "x264enc":
+            case "msdkvp8enc":
+            case "vaapivp8enc":
+                bitrate = uint.min(2048000, bitrate);
+                encode.set("bitrate", bitrate);
+                return bitrate;
+            case "vp8enc":
+                bitrate = uint.min(2147483, bitrate);
+                encode.set("target-bitrate", bitrate * 1000);
+                return bitrate;
+        }
+
+        return 0;
+    }
+
+    public static string? get_decode_prefix(string media, string codec, string decode, JingleRtp.PayloadType? payload_type) {
+        return null;
+    }
+
+    public static string? get_decode_args(string media, string codec, string decode, JingleRtp.PayloadType? payload_type) {
+        if (decode == "opusdec" && payload_type != null && payload_type.parameters.has("useinbandfec", "1")) return " use-inband-fec=true";
+        if (decode == "vaapivp9dec" || decode == "vaapivp8dec" || decode == "vaapih264dec") return " max-errors=100";
+        return null;
+    }
+
+    public static string? get_decode_suffix(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) {
+        return null;
+    }
+
+    public static string? get_depay_args(string media, string codec, string encode, JingleRtp.PayloadType? payload_type) {
+        if (codec == "vp8") return " wait-for-keyframe=true";
         return null;
     }
 
@@ -195,21 +254,24 @@ public class Dino.Plugins.Rtp.CodecUtil {
         unsupported_elements.add(element_name);
     }
 
-    public string? get_decode_bin_description(string media, string? codec, string? element_name = null, string? name = null) {
+    public string? get_decode_bin_description(string media, string? codec, JingleRtp.PayloadType? payload_type, string? element_name = null, string? name = null) {
         if (codec == null) return null;
         string base_name = name ?? @"encode-$codec-$(Random.next_int())";
         string depay = get_depay_element_name(media, codec);
         string decode = element_name ?? get_decode_element_name(media, codec);
         if (depay == null || decode == null) return null;
-        string decode_prefix = get_decode_prefix(media, codec, decode) ?? "";
-        string resample = media == "audio" ? @" ! audioresample name=$base_name-resample" : "";
-        return @"$depay name=$base_name-rtp-depay ! $decode_prefix$decode name=$base_name-decode ! $(media)convert name=$base_name-convert$resample";
+        string decode_prefix = get_decode_prefix(media, codec, decode, payload_type) ?? "";
+        string decode_args = get_decode_args(media, codec, decode, payload_type) ?? "";
+        string decode_suffix = get_decode_suffix(media, codec, decode, payload_type) ?? "";
+        string depay_args = get_depay_args(media, codec, decode, payload_type) ?? "";
+        string resample = media == "audio" ? @" ! audioresample name=$(base_name)_resample" : "";
+        return @"$depay$depay_args name=$(base_name)_rtp_depay ! $decode_prefix$decode$decode_args name=$(base_name)_$(codec)_decode$decode_suffix ! $(media)convert name=$(base_name)_convert$resample";
     }
 
     public Gst.Element? get_decode_bin(string media, JingleRtp.PayloadType payload_type, string? name = null) {
         string? codec = get_codec_from_payload(media, payload_type);
         string base_name = name ?? @"encode-$codec-$(Random.next_int())";
-        string? desc = get_decode_bin_description(media, codec, null, base_name);
+        string? desc = get_decode_bin_description(media, codec, payload_type, null, base_name);
         if (desc == null) return null;
         debug("Pipeline to decode %s %s: %s", media, codec, desc);
         Gst.Element bin = Gst.parse_bin_from_description(desc, true);
@@ -217,22 +279,23 @@ public class Dino.Plugins.Rtp.CodecUtil {
         return bin;
     }
 
-    public string? get_encode_bin_description(string media, string? codec, string? element_name = null, uint pt = 96, string? name = null) {
+    public string? get_encode_bin_description(string media, string? codec, JingleRtp.PayloadType? payload_type, string? element_name = null, string? name = null) {
         if (codec == null) return null;
-        string base_name = name ?? @"encode-$codec-$(Random.next_int())";
+        string base_name = name ?? @"encode_$(codec)_$(Random.next_int())";
         string pay = get_pay_element_name(media, codec);
         string encode = element_name ?? get_encode_element_name(media, codec);
         if (pay == null || encode == null) return null;
-        string encode_prefix = get_encode_prefix(media, codec, encode) ?? "";
-        string encode_suffix = get_encode_suffix(media, codec, encode) ?? "";
-        string resample = media == "audio" ? @" ! audioresample name=$base_name-resample" : "";
-        return @"$(media)convert name=$base_name-convert$resample ! $encode_prefix$encode$encode_suffix ! $pay pt=$pt name=$base_name-rtp-pay";
+        string encode_prefix = get_encode_prefix(media, codec, encode, payload_type) ?? "";
+        string encode_args = get_encode_args(media, codec, encode, payload_type) ?? "";
+        string encode_suffix = get_encode_suffix(media, codec, encode, payload_type) ?? "";
+        string resample = media == "audio" ? @" ! audioresample name=$(base_name)_resample" : "";
+        return @"$(media)convert name=$(base_name)_convert$resample ! $encode_prefix$encode$encode_args name=$(base_name)_encode$encode_suffix ! $pay pt=$(payload_type != null ? payload_type.id : 96) name=$(base_name)_rtp_pay";
     }
 
     public Gst.Element? get_encode_bin(string media, JingleRtp.PayloadType payload_type, string? name = null) {
         string? codec = get_codec_from_payload(media, payload_type);
-        string base_name = name ?? @"encode-$codec-$(Random.next_int())";
-        string? desc = get_encode_bin_description(media, codec, null, payload_type.id, base_name);
+        string base_name = name ?? @"encode_$(codec)_$(Random.next_int())";
+        string? desc = get_encode_bin_description(media, codec, payload_type, null, base_name);
         if (desc == null) return null;
         debug("Pipeline to encode %s %s: %s", media, codec, desc);
         Gst.Element bin = Gst.parse_bin_from_description(desc, true);
diff --git a/plugins/rtp/src/device.vala b/plugins/rtp/src/device.vala
index 3c9a38d2..785f853a 100644
--- a/plugins/rtp/src/device.vala
+++ b/plugins/rtp/src/device.vala
@@ -126,19 +126,20 @@ public class Dino.Plugins.Rtp.Device : MediaDevice, Object {
         element = device.create_element(id);
         pipe.add(element);
         if (is_source) {
-            filter = Gst.ElementFactory.make("capsfilter", @"$id-caps-filter");
+            element.@set("do-timestamp", true);
+            filter = Gst.ElementFactory.make("capsfilter", @"caps_filter_$id");
             filter.@set("caps", get_best_caps());
             pipe.add(filter);
             element.link(filter);
             if (media == "audio" && plugin.echoprobe != null) {
-                dsp = Gst.ElementFactory.make("webrtcdsp", @"$id-dsp");
+                dsp = Gst.ElementFactory.make("webrtcdsp", @"dsp_$id");
                 if (dsp != null) {
                     dsp.@set("probe", plugin.echoprobe.name);
                     pipe.add(dsp);
                     filter.link(dsp);
                 }
             }
-            tee = Gst.ElementFactory.make("tee", @"$id-tee");
+            tee = Gst.ElementFactory.make("tee", @"tee_$id");
             tee.@set("allow-not-linked", true);
             pipe.add(tee);
             (dsp ?? filter).link(tee);
@@ -148,7 +149,7 @@ public class Dino.Plugins.Rtp.Device : MediaDevice, Object {
             element.@set("sync", false);
         }
         if (is_sink && media == "audio") {
-            filter = Gst.ElementFactory.make("capsfilter", @"$id-caps-filter");
+            filter = Gst.ElementFactory.make("capsfilter", @"caps_filter_$id");
             filter.@set("caps", get_best_caps());
             pipe.add(filter);
             if (plugin.echoprobe != null) {
diff --git a/plugins/rtp/src/module.vala b/plugins/rtp/src/module.vala
index 231a9dde..52cc1880 100644
--- a/plugins/rtp/src/module.vala
+++ b/plugins/rtp/src/module.vala
@@ -63,7 +63,7 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
         return supported;
     }
 
-    private async bool supports(string media, JingleRtp.PayloadType payload_type) {
+    private async bool is_payload_supported(string media, JingleRtp.PayloadType payload_type) {
         string codec = CodecUtil.get_codec_from_payload(media, payload_type);
         if (codec == null) return false;
         if (unsupported_codecs.contains(codec)) return false;
@@ -77,7 +77,7 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
             return false;
         }
 
-        string encode_bin = codec_util.get_encode_bin_description(media, codec, encode_element);
+        string encode_bin = codec_util.get_encode_bin_description(media, codec, null, encode_element);
         while (!(yield pipeline_works(media, encode_bin))) {
             debug("%s not suited for encoding %s", encode_element, codec);
             codec_util.mark_element_unsupported(encode_element);
@@ -87,11 +87,11 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
                 unsupported_codecs.add(codec);
                 return false;
             }
-            encode_bin = codec_util.get_encode_bin_description(media, codec, encode_element);
+            encode_bin = codec_util.get_encode_bin_description(media, codec, null, encode_element);
         }
         debug("using %s to encode %s", encode_element, codec);
 
-        string decode_bin = codec_util.get_decode_bin_description(media, codec, decode_element);
+        string decode_bin = codec_util.get_decode_bin_description(media, codec, null, decode_element);
         while (!(yield pipeline_works(media, @"$encode_bin ! $decode_bin"))) {
             debug("%s not suited for decoding %s", decode_element, codec);
             codec_util.mark_element_unsupported(decode_element);
@@ -101,7 +101,7 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
                 unsupported_codecs.add(codec);
                 return false;
             }
-            decode_bin = codec_util.get_decode_bin_description(media, codec, decode_element);
+            decode_bin = codec_util.get_decode_bin_description(media, codec, null, decode_element);
         }
         debug("using %s to decode %s", decode_element, codec);
 
@@ -109,8 +109,21 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
         return true;
     }
 
+    public override bool is_header_extension_supported(string media, JingleRtp.HeaderExtension ext) {
+        if (media == "video" && ext.uri == "urn:3gpp:video-orientation") return true;
+        return false;
+    }
+
+    public override Gee.List<JingleRtp.HeaderExtension> get_suggested_header_extensions(string media) {
+        Gee.List<JingleRtp.HeaderExtension> exts = new ArrayList<JingleRtp.HeaderExtension>();
+        if (media == "video") {
+            exts.add(new JingleRtp.HeaderExtension(1, "urn:3gpp:video-orientation"));
+        }
+        return exts;
+    }
+
     public async void add_if_supported(Gee.List<JingleRtp.PayloadType> list, string media, JingleRtp.PayloadType payload_type) {
-        if (yield supports(media, payload_type)) {
+        if (yield is_payload_supported(media, payload_type)) {
             list.add(payload_type);
         }
     }
@@ -118,58 +131,34 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
     public override async Gee.List<JingleRtp.PayloadType> get_supported_payloads(string media) {
         Gee.List<JingleRtp.PayloadType> list = new ArrayList<JingleRtp.PayloadType>(JingleRtp.PayloadType.equals_func);
         if (media == "audio") {
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                channels = 2,
-                clockrate = 48000,
-                name = "opus",
-                id = 99
-            });
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                channels = 1,
-                clockrate = 32000,
-                name = "speex",
-                id = 100
-            });
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                channels = 1,
-                clockrate = 16000,
-                name = "speex",
-                id = 101
-            });
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                channels = 1,
-                clockrate = 8000,
-                name = "speex",
-                id = 102
-            });
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                channels = 1,
-                clockrate = 8000,
-                name = "PCMU",
-                id = 0
-            });
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                channels = 1,
-                clockrate = 8000,
-                name = "PCMA",
-                id = 8
-            });
+            var opus = new JingleRtp.PayloadType() { channels = 2, clockrate = 48000, name = "opus", id = 99 };
+            opus.parameters["useinbandfec"] = "1";
+            var speex32 = new JingleRtp.PayloadType() { channels = 1, clockrate = 32000, name = "speex", id = 100 };
+            var speex16 = new JingleRtp.PayloadType() { channels = 1, clockrate = 16000, name = "speex", id = 101 };
+            var speex8 = new JingleRtp.PayloadType() { channels = 1, clockrate = 8000, name = "speex", id = 102 };
+            var pcmu = new JingleRtp.PayloadType() { channels = 1, clockrate = 8000, name = "PCMU", id = 0 };
+            var pcma = new JingleRtp.PayloadType() { channels = 1, clockrate = 8000, name = "PCMA", id = 8 };
+            yield add_if_supported(list, media, opus);
+            yield add_if_supported(list, media, speex32);
+            yield add_if_supported(list, media, speex16);
+            yield add_if_supported(list, media, speex8);
+            yield add_if_supported(list, media, pcmu);
+            yield add_if_supported(list, media, pcma);
         } else if (media == "video") {
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                clockrate = 90000,
-                name = "H264",
-                id = 96
-            });
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                clockrate = 90000,
-                name = "VP9",
-                id = 97
-            });
-            yield add_if_supported(list, media, new JingleRtp.PayloadType() {
-                clockrate = 90000,
-                name = "VP8",
-                id = 98
-            });
+            var h264 = new JingleRtp.PayloadType() { clockrate = 90000, name = "H264", id = 96 };
+            var vp9 = new JingleRtp.PayloadType() { clockrate = 90000, name = "VP9", id = 97 };
+            var vp8 = new JingleRtp.PayloadType() { clockrate = 90000, name = "VP8", id = 98 };
+            var rtcp_fbs = new ArrayList<JingleRtp.RtcpFeedback>();
+            rtcp_fbs.add(new JingleRtp.RtcpFeedback("goog-remb"));
+            rtcp_fbs.add(new JingleRtp.RtcpFeedback("ccm", "fir"));
+            rtcp_fbs.add(new JingleRtp.RtcpFeedback("nack"));
+            rtcp_fbs.add(new JingleRtp.RtcpFeedback("nack", "pli"));
+            h264.rtcp_fbs.add_all(rtcp_fbs);
+            vp9.rtcp_fbs.add_all(rtcp_fbs);
+            vp8.rtcp_fbs.add_all(rtcp_fbs);
+            yield add_if_supported(list, media, h264);
+            yield add_if_supported(list, media, vp9);
+            yield add_if_supported(list, media, vp8);
         } else {
             warning("Unsupported media type: %s", media);
         }
@@ -179,11 +168,15 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
     public override async JingleRtp.PayloadType? pick_payload_type(string media, Gee.List<JingleRtp.PayloadType> payloads) {
         if (media == "audio") {
             foreach (JingleRtp.PayloadType type in payloads) {
-                if (yield supports(media, type)) return type;
+                if (yield is_payload_supported(media, type)) return adjust_payload_type(media, type.clone());
             }
         } else if (media == "video") {
+            // We prefer H.264 (best support for hardware acceleration and good overall codec quality)
+            JingleRtp.PayloadType? h264 = payloads.first_match((it) => it.name.up() == "H264");
+            if (h264 != null && yield is_payload_supported(media, h264)) return adjust_payload_type(media, h264.clone());
+            // Take first of the list that we do support otherwise
             foreach (JingleRtp.PayloadType type in payloads) {
-                if (yield supports(media, type)) return type;
+                if (yield is_payload_supported(media, type)) return adjust_payload_type(media, type.clone());
             }
         } else {
             warning("Unsupported media type: %s", media);
@@ -191,6 +184,28 @@ public class Dino.Plugins.Rtp.Module : JingleRtp.Module {
         return null;
     }
 
+    public JingleRtp.PayloadType adjust_payload_type(string media, JingleRtp.PayloadType type) {
+        var iter = type.rtcp_fbs.iterator();
+        while (iter.next()) {
+            var fb = iter.@get();
+            switch (fb.type_) {
+                case "goog-remb":
+                    if (fb.subtype != null) iter.remove();
+                    break;
+                case "ccm":
+                    if (fb.subtype != "fir") iter.remove();
+                    break;
+                case "nack":
+                    if (fb.subtype != null && fb.subtype != "pli") iter.remove();
+                    break;
+                default:
+                    iter.remove();
+                    break;
+            }
+        }
+        return type;
+    }
+
     public override JingleRtp.Stream create_stream(Jingle.Content content) {
         return plugin.open_stream(content);
     }
diff --git a/plugins/rtp/src/plugin.vala b/plugins/rtp/src/plugin.vala
index 40ad1e0f..f0ad7db2 100644
--- a/plugins/rtp/src/plugin.vala
+++ b/plugins/rtp/src/plugin.vala
@@ -65,6 +65,9 @@ public class Dino.Plugins.Rtp.Plugin : RootInterface, VideoCallPlugin, Object {
         }
         rtpbin.pad_added.connect(on_rtp_pad_added);
         rtpbin.@set("latency", 100);
+        rtpbin.@set("do-lost", true);
+        rtpbin.@set("do-sync-event", true);
+        rtpbin.@set("drop-on-latency", true);
         rtpbin.connect("signal::request-pt-map", request_pt_map, this);
         pipe.add(rtpbin);
 
@@ -160,6 +163,17 @@ public class Dino.Plugins.Rtp.Plugin : RootInterface, VideoCallPlugin, Object {
             case Gst.MessageType.QOS:
                 // Ignore
                 break;
+            case Gst.MessageType.LATENCY:
+                if (message.src != null && message.src.name != null && message.src is Gst.Element) {
+                    Gst.Query latency_query = new Gst.Query.latency();
+                    if (((Gst.Element)message.src).query(latency_query)) {
+                        bool live;
+                        Gst.ClockTime min_latency, max_latency;
+                        latency_query.parse_latency(out live, out min_latency, out max_latency);
+                        debug("Latency message from %s: live=%s, min_latency=%s, max_latency=%s", message.src.name, live.to_string(), min_latency.to_string(), max_latency.to_string());
+                    }
+                }
+                break;
             default:
                 debug("Pipe bus message: %s", message.type.to_string());
                 break;
diff --git a/plugins/rtp/src/stream.vala b/plugins/rtp/src/stream.vala
index 3a63f3fa..23634aa3 100644
--- a/plugins/rtp/src/stream.vala
+++ b/plugins/rtp/src/stream.vala
@@ -19,9 +19,12 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
     private Gst.App.Src recv_rtp;
     private Gst.App.Src recv_rtcp;
     private Gst.Element encode;
+    private Gst.RTP.BasePayload encode_pay;
     private Gst.Element decode;
+    private Gst.RTP.BaseDepayload decode_depay;
     private Gst.Element input;
     private Gst.Element output;
+    private Gst.Element session;
 
     private Device _input_device;
     public Device input_device { get { return _input_device; } set {
@@ -85,15 +88,15 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         }
 
         // Create app elements
-        send_rtp = Gst.ElementFactory.make("appsink", @"rtp-sink-$rtpid") as Gst.App.Sink;
+        send_rtp = Gst.ElementFactory.make("appsink", @"rtp_sink_$rtpid") as Gst.App.Sink;
         send_rtp.async = false;
-        send_rtp.caps = CodecUtil.get_caps(media, payload_type);
+        send_rtp.caps = CodecUtil.get_caps(media, payload_type, false);
         send_rtp.emit_signals = true;
         send_rtp.sync = false;
         send_rtp.new_sample.connect(on_new_sample);
         pipe.add(send_rtp);
 
-        send_rtcp = Gst.ElementFactory.make("appsink", @"rtcp-sink-$rtpid") as Gst.App.Sink;
+        send_rtcp = Gst.ElementFactory.make("appsink", @"rtcp_sink_$rtpid") as Gst.App.Sink;
         send_rtcp.async = false;
         send_rtcp.caps = new Gst.Caps.empty_simple("application/x-rtcp");
         send_rtcp.emit_signals = true;
@@ -101,14 +104,14 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         send_rtcp.new_sample.connect(on_new_sample);
         pipe.add(send_rtcp);
 
-        recv_rtp = Gst.ElementFactory.make("appsrc", @"rtp-src-$rtpid") as Gst.App.Src;
-        recv_rtp.caps = CodecUtil.get_caps(media, payload_type);
+        recv_rtp = Gst.ElementFactory.make("appsrc", @"rtp_src_$rtpid") as Gst.App.Src;
+        recv_rtp.caps = CodecUtil.get_caps(media, payload_type, true);
         recv_rtp.do_timestamp = true;
         recv_rtp.format = Gst.Format.TIME;
         recv_rtp.is_live = true;
         pipe.add(recv_rtp);
 
-        recv_rtcp = Gst.ElementFactory.make("appsrc", @"rtcp-src-$rtpid") as Gst.App.Src;
+        recv_rtcp = Gst.ElementFactory.make("appsrc", @"rtcp_src_$rtpid") as Gst.App.Src;
         recv_rtcp.caps = new Gst.Caps.empty_simple("application/x-rtcp");
         recv_rtcp.do_timestamp = true;
         recv_rtcp.format = Gst.Format.TIME;
@@ -122,7 +125,8 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         recv_rtcp.get_static_pad("src").link(recv_rtcp_sink_pad);
 
         // Connect input
-        encode = codec_util.get_encode_bin(media, payload_type, @"encode-$rtpid");
+        encode = codec_util.get_encode_bin(media, payload_type, @"encode_$rtpid");
+        encode_pay = (Gst.RTP.BasePayload)((Gst.Bin)encode).get_by_name(@"encode_$(rtpid)_rtp_pay");
         pipe.add(encode);
         send_rtp_sink_pad = rtpbin.get_request_pad(@"send_rtp_sink_$rtpid");
         encode.get_static_pad("src").link(send_rtp_sink_pad);
@@ -131,7 +135,8 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         }
 
         // Connect output
-        decode = codec_util.get_decode_bin(media, payload_type, @"decode-$rtpid");
+        decode = codec_util.get_decode_bin(media, payload_type, @"decode_$rtpid");
+        decode_depay = (Gst.RTP.BaseDepayload)((Gst.Bin)encode).get_by_name(@"decode_$(rtpid)_rtp_depay");
         pipe.add(decode);
         if (output != null) {
             decode.link(output);
@@ -144,6 +149,110 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         created = true;
         push_recv_data = true;
         plugin.unpause();
+
+        GLib.Signal.emit_by_name(rtpbin, "get-session", rtpid, out session);
+        if (session != null && payload_type.rtcp_fbs.any_match((it) => it.type_ == "goog-remb")) {
+            Object internal_session;
+            session.@get("internal-session", out internal_session);
+            if (internal_session != null) {
+                internal_session.connect("signal::on-feedback-rtcp", on_feedback_rtcp, this);
+            }
+            Timeout.add(1000, () => remb_adjust());
+        }
+        if (media == "video") {
+            codec_util.update_bitrate(media, payload_type, encode, 256);
+        }
+    }
+
+    private uint remb = 256;
+    private int last_packets_lost = -1;
+    private uint64 last_packets_received;
+    private uint64 last_octets_received;
+    private bool remb_adjust() {
+        unowned Gst.Structure? stats;
+        if (session == null) {
+            debug("Session for %u finished, turning off remb adjustment", rtpid);
+            return Source.REMOVE;
+        }
+        session.get("stats", out stats);
+        if (stats == null) {
+            warning("No stats for session %u", rtpid);
+            return Source.REMOVE;
+        }
+        unowned ValueArray? source_stats;
+        stats.get("source-stats", typeof(ValueArray), out source_stats);
+        if (source_stats == null) {
+            warning("No source-stats for session %u", rtpid);
+            return Source.REMOVE;
+        }
+        foreach (Value value in source_stats.values) {
+            unowned Gst.Structure source_stat = (Gst.Structure) value.get_boxed();
+            uint ssrc;
+            if (!source_stat.get_uint("ssrc", out ssrc)) continue;
+            if (ssrc.to_string() == participant_ssrc) {
+                int packets_lost;
+                uint64 packets_received, octets_received;
+                source_stat.get_int("packets-lost", out packets_lost);
+                source_stat.get_uint64("packets-received", out packets_received);
+                source_stat.get_uint64("octets-received", out octets_received);
+                int new_lost = packets_lost - last_packets_lost;
+                uint64 new_received = packets_received - last_packets_received;
+                uint64 new_octets = octets_received - last_octets_received;
+                if (new_received == 0) continue;
+                last_packets_lost = packets_lost;
+                last_packets_received = packets_received;
+                last_octets_received = octets_received;
+                double loss_rate = (double)new_lost / (double)(new_lost + new_received);
+                if (new_lost <= 0 || loss_rate < 0.02) {
+                    remb = (uint)(1.08 * (double)remb);
+                } else if (loss_rate > 0.1) {
+                    remb = (uint)((1.0 - 0.5 * loss_rate) * (double)remb);
+                }
+                remb = uint.max(remb, (uint)((new_octets * 8) / 1000));
+                remb = uint.max(16, remb); // Never go below 16
+                uint8[] data = new uint8[] {
+                    143, 206, 0, 5,
+                    0, 0, 0, 0,
+                    0, 0, 0, 0,
+                    'R', 'E', 'M', 'B',
+                    1, 0, 0, 0,
+                    0, 0, 0, 0
+                };
+                data[4] = (uint8)((encode_pay.ssrc >> 24) & 0xff);
+                data[5] = (uint8)((encode_pay.ssrc >> 16) & 0xff);
+                data[6] = (uint8)((encode_pay.ssrc >> 8) & 0xff);
+                data[7] = (uint8)(encode_pay.ssrc & 0xff);
+                uint8 br_exp = 0;
+                uint32 br_mant = remb * 1000;
+                uint8 bits = (uint8)Math.log2(br_mant);
+                if (bits > 16) {
+                    br_exp = (uint8)bits - 16;
+                    br_mant = br_mant >> br_exp;
+                }
+                data[17] = (uint8)((br_exp << 2) | ((br_mant >> 16) & 0x3));
+                data[18] = (uint8)((br_mant >> 8) & 0xff);
+                data[19] = (uint8)(br_mant & 0xff);
+                data[20] = (uint8)((ssrc >> 24) & 0xff);
+                data[21] = (uint8)((ssrc >> 16) & 0xff);
+                data[22] = (uint8)((ssrc >> 8) & 0xff);
+                data[23] = (uint8)(ssrc & 0xff);
+                encrypt_and_send_rtcp(data);
+            }
+        }
+        return Source.CONTINUE;
+    }
+
+    private static void on_feedback_rtcp(Gst.Element session, uint type, uint fbtype, uint sender_ssrc, uint media_ssrc, Gst.Buffer? fci, Stream self) {
+        if (type == 206 && fbtype == 15 && fci != null && sender_ssrc.to_string() == self.participant_ssrc) {
+            // https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03
+            uint8[] data;
+            fci.extract_dup(0, fci.get_size(), out data);
+            if (data[0] != 'R' || data[1] != 'E' || data[2] != 'M' || data[3] != 'B') return;
+            uint8 br_exp = data[5] >> 2;
+            uint32 br_mant = (((uint32)data[5] & 0x3) << 16) + ((uint32)data[6] << 8) + (uint32)data[7];
+            uint bitrate = (br_mant << br_exp) / 1000;
+            self.codec_util.update_bitrate(self.media, self.payload_type, self.encode, bitrate * 8);
+        }
     }
 
     private void prepare_local_crypto() {
@@ -167,22 +276,26 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
             if (crypto_session.has_encrypt) {
                 data = crypto_session.encrypt_rtp(data);
             }
-            on_send_rtp_data(new Bytes.take(data));
+            on_send_rtp_data(new Bytes.take((owned) data));
         } else if (sink == send_rtcp) {
-            if (crypto_session.has_encrypt) {
-                data = crypto_session.encrypt_rtcp(data);
-            }
-            if (rtcp_mux) {
-                on_send_rtp_data(new Bytes.take(data));
-            } else {
-                on_send_rtcp_data(new Bytes.take(data));
-            }
+            encrypt_and_send_rtcp((owned) data);
         } else {
             warning("unknown sample");
         }
         return Gst.FlowReturn.OK;
     }
 
+    private void encrypt_and_send_rtcp(owned uint8[] data) {
+        if (crypto_session.has_encrypt) {
+            data = crypto_session.encrypt_rtcp(data);
+        }
+        if (rtcp_mux) {
+            on_send_rtp_data(new Bytes.take((owned) data));
+        } else {
+            on_send_rtcp_data(new Bytes.take((owned) data));
+        }
+    }
+
     private static Gst.PadProbeReturn drop_probe() {
         return Gst.PadProbeReturn.DROP;
     }
@@ -211,6 +324,7 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         encode.get_static_pad("src").unlink(send_rtp_sink_pad);
         pipe.remove(encode);
         encode = null;
+        encode_pay = null;
 
         // Disconnect RTP sending
         if (send_rtp_src_pad != null) {
@@ -243,6 +357,7 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         decode.set_state(Gst.State.NULL);
         pipe.remove(decode);
         decode = null;
+        decode_depay = null;
         output = null;
 
         // Disconnect output device
@@ -276,6 +391,8 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         send_rtcp_src_pad = null;
         send_rtp_src_pad = null;
         recv_rtp_src_pad = null;
+
+        session = null;
     }
 
     private void prepare_remote_crypto() {
@@ -285,6 +402,9 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         }
     }
 
+    private uint16 previous_video_orientation_degree = uint16.MAX;
+    public signal void video_orientation_changed(uint16 degree);
+
     public override void on_recv_rtp_data(Bytes bytes) {
         if (rtcp_mux && bytes.length >= 2 && bytes.get(1) >= 192 && bytes.get(1) < 224) {
             on_recv_rtcp_data(bytes);
@@ -301,6 +421,33 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
         }
         if (push_recv_data) {
             Gst.Buffer buffer = new Gst.Buffer.wrapped((owned) data);
+            Gst.RTP.Buffer rtp_buffer;
+            if (Gst.RTP.Buffer.map(buffer, Gst.MapFlags.READ, out rtp_buffer)) {
+                if (rtp_buffer.get_extension()) {
+                    Xmpp.Xep.JingleRtp.HeaderExtension? ext = header_extensions.first_match((it) => it.uri == "urn:3gpp:video-orientation");
+                    if (ext != null) {
+                        unowned uint8[] extension_data;
+                        if (rtp_buffer.get_extension_onebyte_header(ext.id, 0, out extension_data) && extension_data.length == 1) {
+                            bool camera = (extension_data[0] & 0x8) > 0;
+                            bool flip = (extension_data[0] & 0x4) > 0;
+                            uint8 rotation = extension_data[0] & 0x3;
+                            uint16 rotation_degree = uint16.MAX;
+                            switch(rotation) {
+                                case 0: rotation_degree = 0; break;
+                                case 1: rotation_degree = 90; break;
+                                case 2: rotation_degree = 180; break;
+                                case 3: rotation_degree = 270; break;
+                            }
+                            if (rotation_degree != previous_video_orientation_degree) {
+                                video_orientation_changed(rotation_degree);
+                                previous_video_orientation_degree = rotation_degree;
+                            }
+                        }
+                    }
+                }
+                rtp_buffer.unmap();
+            }
+
             // FIXME: VAPI file in Vala < 0.49.1 has a bug that results in broken ownership of buffer in push_buffer()
             // We workaround by using the plain signal. The signal unfortunately will cause an unnecessary copy of
             // the underlying buffer, so and some point we should move over to the new version (once we require
@@ -449,6 +596,8 @@ public class Dino.Plugins.Rtp.Stream : Xmpp.Xep.JingleRtp.Stream {
 public class Dino.Plugins.Rtp.VideoStream : Stream {
     private Gee.List<Gst.Element> outputs = new ArrayList<Gst.Element>();
     private Gst.Element output_tee;
+    private Gst.Element rotate;
+    private ulong video_orientation_changed_handler;
 
     public VideoStream(Plugin plugin, Xmpp.Xep.Jingle.Content content) {
         base(plugin, content);
@@ -456,11 +605,15 @@ public class Dino.Plugins.Rtp.VideoStream : Stream {
     }
 
     public override void create() {
+        video_orientation_changed_handler = video_orientation_changed.connect(on_video_orientation_changed);
         plugin.pause();
-        output_tee = Gst.ElementFactory.make("tee", null);
+        rotate = Gst.ElementFactory.make("videoflip", @"video_rotate_$rtpid");
+        pipe.add(rotate);
+        output_tee = Gst.ElementFactory.make("tee", @"video_tee_$rtpid");
         output_tee.@set("allow-not-linked", true);
         pipe.add(output_tee);
-        add_output(output_tee);
+        rotate.link(output_tee);
+        add_output(rotate);
         base.create();
         foreach (Gst.Element output in outputs) {
             output_tee.link(output);
@@ -468,19 +621,44 @@ public class Dino.Plugins.Rtp.VideoStream : Stream {
         plugin.unpause();
     }
 
+    private void on_video_orientation_changed(uint16 degree) {
+        if (rotate != null) {
+            switch (degree) {
+                case 0:
+                    rotate.@set("method", 0);
+                    break;
+                case 90:
+                    rotate.@set("method", 1);
+                    break;
+                case 180:
+                    rotate.@set("method", 2);
+                    break;
+                case 270:
+                    rotate.@set("method", 3);
+                    break;
+            }
+        }
+    }
+
     public override void destroy() {
         foreach (Gst.Element output in outputs) {
             output_tee.unlink(output);
         }
         base.destroy();
+        rotate.set_locked_state(true);
+        rotate.set_state(Gst.State.NULL);
+        rotate.unlink(output_tee);
+        pipe.remove(rotate);
+        rotate = null;
         output_tee.set_locked_state(true);
         output_tee.set_state(Gst.State.NULL);
         pipe.remove(output_tee);
         output_tee = null;
+        disconnect(video_orientation_changed_handler);
     }
 
     public override void add_output(Gst.Element element) {
-        if (element == output_tee) {
+        if (element == output_tee || element == rotate) {
             base.add_output(element);
             return;
         }
@@ -491,7 +669,7 @@ public class Dino.Plugins.Rtp.VideoStream : Stream {
     }
 
     public override void remove_output(Gst.Element element) {
-        if (element == output_tee) {
+        if (element == output_tee || element == rotate) {
             base.remove_output(element);
             return;
         }
diff --git a/plugins/rtp/src/video_widget.vala b/plugins/rtp/src/video_widget.vala
index fa5ba138..351069a7 100644
--- a/plugins/rtp/src/video_widget.vala
+++ b/plugins/rtp/src/video_widget.vala
@@ -19,7 +19,7 @@ public class Dino.Plugins.Rtp.VideoWidget : Gtk.Bin, Dino.Plugins.VideoCallWidge
         this.plugin = plugin;
 
         id = last_id++;
-        element = Gst.ElementFactory.make("gtksink", @"video-widget-$id");
+        element = Gst.ElementFactory.make("gtksink", @"video_widget_$id");
         if (element != null) {
             Gtk.Widget widget;
             element.@get("widget", out widget);
@@ -51,8 +51,8 @@ public class Dino.Plugins.Rtp.VideoWidget : Gtk.Bin, Dino.Plugins.VideoCallWidge
         if (connected_stream == null) return;
         plugin.pause();
         pipe.add(element);
-        convert = Gst.parse_bin_from_description(@"videoconvert name=video-widget-$id-convert", true);
-        convert.name = @"video-widget-$id-prepare";
+        convert = Gst.parse_bin_from_description(@"videoconvert name=video_widget_$(id)_convert", true);
+        convert.name = @"video_widget_$(id)_prepare";
         pipe.add(convert);
         convert.link(element);
         connected_stream.add_output(convert);
@@ -68,8 +68,8 @@ public class Dino.Plugins.Rtp.VideoWidget : Gtk.Bin, Dino.Plugins.VideoCallWidge
         if (connected_device == null) return;
         plugin.pause();
         pipe.add(element);
-        convert = Gst.parse_bin_from_description(@"videoflip method=horizontal-flip name=video-widget-$id-flip ! videoconvert name=video-widget-$id-convert", true);
-        convert.name = @"video-widget-$id-prepare";
+        convert = Gst.parse_bin_from_description(@"videoflip method=horizontal-flip name=video_widget_$(id)_flip ! videoconvert name=video_widget_$(id)_convert", true);
+        convert.name = @"video_widget_$(id)_prepare";
         pipe.add(convert);
         convert.link(element);
         connected_device.link_source().link(convert);
diff --git a/plugins/rtp/vapi/gstreamer-rtp-1.0.vapi b/plugins/rtp/vapi/gstreamer-rtp-1.0.vapi
new file mode 100644
index 00000000..30490896
--- /dev/null
+++ b/plugins/rtp/vapi/gstreamer-rtp-1.0.vapi
@@ -0,0 +1,625 @@
+// Fixme: This is fetched from development code of Vala upstream which fixed a few bugs.
+/* gstreamer-rtp-1.0.vapi generated by vapigen, do not modify. */
+
+[CCode (cprefix = "Gst", gir_namespace = "GstRtp", gir_version = "1.0", lower_case_cprefix = "gst_")]
+namespace Gst {
+	namespace RTCP {
+		[CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+		[GIR (name = "RTCPBuffer")]
+		public struct Buffer {
+			public weak Gst.Buffer buffer;
+			public bool add_packet (Gst.RTCP.Type type, Gst.RTCP.Packet packet);
+			public bool get_first_packet (Gst.RTCP.Packet packet);
+			public uint get_packet_count ();
+			public static bool map (Gst.Buffer buffer, Gst.MapFlags flags, out Gst.RTCP.Buffer rtcp);
+			public static Gst.Buffer @new (uint mtu);
+			public static Gst.Buffer new_copy_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint8[] data);
+			public static Gst.Buffer new_take_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] owned uint8[] data);
+			public bool unmap ();
+			public static bool validate (Gst.Buffer buffer);
+			public static bool validate_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint8[] data);
+			[Version (since = "1.6")]
+			public static bool validate_data_reduced ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint8[] data);
+			[Version (since = "1.6")]
+			public static bool validate_reduced (Gst.Buffer buffer);
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+		[GIR (name = "RTCPPacket")]
+		public struct Packet {
+			public weak Gst.RTCP.Buffer? rtcp;
+			public uint offset;
+			[Version (since = "1.10")]
+			public bool add_profile_specific_ext ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint8[] data);
+			public bool add_rb (uint32 ssrc, uint8 fractionlost, int32 packetslost, uint32 exthighestseq, uint32 jitter, uint32 lsr, uint32 dlsr);
+			[Version (since = "1.10")]
+			public uint8 app_get_data ();
+			[Version (since = "1.10")]
+			public uint16 app_get_data_length ();
+			[Version (since = "1.10")]
+			public unowned string app_get_name ();
+			[Version (since = "1.10")]
+			public uint32 app_get_ssrc ();
+			[Version (since = "1.10")]
+			public uint8 app_get_subtype ();
+			[Version (since = "1.10")]
+			public bool app_set_data_length (uint16 wordlen);
+			[Version (since = "1.10")]
+			public void app_set_name (string name);
+			[Version (since = "1.10")]
+			public void app_set_ssrc (uint32 ssrc);
+			[Version (since = "1.10")]
+			public void app_set_subtype (uint8 subtype);
+			public bool bye_add_ssrc (uint32 ssrc);
+			public bool bye_add_ssrcs ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] uint32[] ssrc);
+			public uint32 bye_get_nth_ssrc (uint nth);
+			public string bye_get_reason ();
+			public uint8 bye_get_reason_len ();
+			public uint bye_get_ssrc_count ();
+			public bool bye_set_reason (string reason);
+			[Version (since = "1.10")]
+			public bool copy_profile_specific_ext ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] out uint8[] data);
+			public uint8 fb_get_fci ();
+			public uint16 fb_get_fci_length ();
+			public uint32 fb_get_media_ssrc ();
+			public uint32 fb_get_sender_ssrc ();
+			public Gst.RTCP.FBType fb_get_type ();
+			public bool fb_set_fci_length (uint16 wordlen);
+			public void fb_set_media_ssrc (uint32 ssrc);
+			public void fb_set_sender_ssrc (uint32 ssrc);
+			public void fb_set_type (Gst.RTCP.FBType type);
+			public uint8 get_count ();
+			public uint16 get_length ();
+			public bool get_padding ();
+			[Version (since = "1.10")]
+			public bool get_profile_specific_ext ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "guint")] out unowned uint8[] data);
+			[Version (since = "1.10")]
+			public uint16 get_profile_specific_ext_length ();
+			public void get_rb (uint nth, out uint32 ssrc, out uint8 fractionlost, out int32 packetslost, out uint32 exthighestseq, out uint32 jitter, out uint32 lsr, out uint32 dlsr);
+			public uint get_rb_count ();
+			public Gst.RTCP.Type get_type ();
+			public bool move_to_next ();
+			public bool remove ();
+			public uint32 rr_get_ssrc ();
+			public void rr_set_ssrc (uint32 ssrc);
+			public bool sdes_add_entry (Gst.RTCP.SDESType type, [CCode (array_length_cname = "len", array_length_pos = 1.5, array_length_type = "guint8")] uint8[] data);
+			public bool sdes_add_item (uint32 ssrc);
+			public bool sdes_copy_entry (out Gst.RTCP.SDESType type, [CCode (array_length_cname = "len", array_length_pos = 1.5, array_length_type = "guint8")] out uint8[] data);
+			public bool sdes_first_entry ();
+			public bool sdes_first_item ();
+			public bool sdes_get_entry (out Gst.RTCP.SDESType type, [CCode (array_length_cname = "len", array_length_pos = 1.5, array_length_type = "guint8")] out unowned uint8[] data);
+			public uint sdes_get_item_count ();
+			public uint32 sdes_get_ssrc ();
+			public bool sdes_next_entry ();
+			public bool sdes_next_item ();
+			public void set_rb (uint nth, uint32 ssrc, uint8 fractionlost, int32 packetslost, uint32 exthighestseq, uint32 jitter, uint32 lsr, uint32 dlsr);
+			public void sr_get_sender_info (out uint32 ssrc, out uint64 ntptime, out uint32 rtptime, out uint32 packet_count, out uint32 octet_count);
+			public void sr_set_sender_info (uint32 ssrc, uint64 ntptime, uint32 rtptime, uint32 packet_count, uint32 octet_count);
+			[Version (since = "1.16")]
+			public bool xr_first_rb ();
+			[Version (since = "1.16")]
+			public uint16 xr_get_block_length ();
+			[Version (since = "1.16")]
+			public Gst.RTCP.XRType xr_get_block_type ();
+			[Version (since = "1.16")]
+			public bool xr_get_dlrr_block (uint nth, out uint32 ssrc, out uint32 last_rr, out uint32 delay);
+			[Version (since = "1.16")]
+			public bool xr_get_prt_by_seq (uint16 seq, out uint32 receipt_time);
+			[Version (since = "1.16")]
+			public bool xr_get_prt_info (out uint32 ssrc, out uint8 thinning, out uint16 begin_seq, out uint16 end_seq);
+			[Version (since = "1.16")]
+			public bool xr_get_rle_info (out uint32 ssrc, out uint8 thinning, out uint16 begin_seq, out uint16 end_seq, out uint32 chunk_count);
+			[Version (since = "1.16")]
+			public bool xr_get_rle_nth_chunk (uint nth, out uint16 chunk);
+			[Version (since = "1.16")]
+			public bool xr_get_rrt (out uint64 timestamp);
+			[Version (since = "1.16")]
+			public uint32 xr_get_ssrc ();
+			[Version (since = "1.16")]
+			public bool xr_get_summary_info (out uint32 ssrc, out uint16 begin_seq, out uint16 end_seq);
+			[Version (since = "1.16")]
+			public bool xr_get_summary_jitter (out uint32 min_jitter, out uint32 max_jitter, out uint32 mean_jitter, out uint32 dev_jitter);
+			[Version (since = "1.16")]
+			public bool xr_get_summary_pkt (out uint32 lost_packets, out uint32 dup_packets);
+			[Version (since = "1.16")]
+			public bool xr_get_summary_ttl (out bool is_ipv4, out uint8 min_ttl, out uint8 max_ttl, out uint8 mean_ttl, out uint8 dev_ttl);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_burst_metrics (out uint8 burst_density, out uint8 gap_density, out uint16 burst_duration, out uint16 gap_duration);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_configuration_params (out uint8 gmin, out uint8 rx_config);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_delay_metrics (out uint16 roundtrip_delay, out uint16 end_system_delay);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_jitter_buffer_params (out uint16 jb_nominal, out uint16 jb_maximum, out uint16 jb_abs_max);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_metrics_ssrc (out uint32 ssrc);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_packet_metrics (out uint8 loss_rate, out uint8 discard_rate);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_quality_metrics (out uint8 r_factor, out uint8 ext_r_factor, out uint8 mos_lq, out uint8 mos_cq);
+			[Version (since = "1.16")]
+			public bool xr_get_voip_signal_metrics (out uint8 signal_level, out uint8 noise_level, out uint8 rerl, out uint8 gmin);
+			[Version (since = "1.16")]
+			public bool xr_next_rb ();
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTCP_", type_id = "gst_rtcpfb_type_get_type ()")]
+		[GIR (name = "RTCPFBType")]
+		public enum FBType {
+			FB_TYPE_INVALID,
+			RTPFB_TYPE_NACK,
+			RTPFB_TYPE_TMMBR,
+			RTPFB_TYPE_TMMBN,
+			RTPFB_TYPE_RTCP_SR_REQ,
+			RTPFB_TYPE_TWCC,
+			PSFB_TYPE_PLI,
+			PSFB_TYPE_SLI,
+			PSFB_TYPE_RPSI,
+			PSFB_TYPE_AFB,
+			PSFB_TYPE_FIR,
+			PSFB_TYPE_TSTR,
+			PSFB_TYPE_TSTN,
+			PSFB_TYPE_VBCN
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTCP_SDES_", type_id = "gst_rtcpsdes_type_get_type ()")]
+		[GIR (name = "RTCPSDESType")]
+		public enum SDESType {
+			INVALID,
+			END,
+			CNAME,
+			NAME,
+			EMAIL,
+			PHONE,
+			LOC,
+			TOOL,
+			NOTE,
+			PRIV;
+			[CCode (cname = "gst_rtcp_sdes_name_to_type")]
+			public static Gst.RTCP.SDESType from_string (string name);
+			[CCode (cname = "gst_rtcp_sdes_type_to_name")]
+			public unowned string to_string ();
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTCP_TYPE_", type_id = "gst_rtcp_type_get_type ()")]
+		[GIR (name = "RTCPType")]
+		public enum Type {
+			INVALID,
+			SR,
+			RR,
+			SDES,
+			BYE,
+			APP,
+			RTPFB,
+			PSFB,
+			XR
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTCP_XR_TYPE_", type_id = "gst_rtcpxr_type_get_type ()")]
+		[GIR (name = "RTCPXRType")]
+		[Version (since = "1.16")]
+		public enum XRType {
+			INVALID,
+			LRLE,
+			DRLE,
+			PRT,
+			RRT,
+			DLRR,
+			SSUMM,
+			VOIP_METRICS
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_MAX_BYE_SSRC_COUNT")]
+		public const int MAX_BYE_SSRC_COUNT;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_MAX_RB_COUNT")]
+		public const int MAX_RB_COUNT;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_MAX_SDES")]
+		public const int MAX_SDES;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_MAX_SDES_ITEM_COUNT")]
+		public const int MAX_SDES_ITEM_COUNT;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_REDUCED_SIZE_VALID_MASK")]
+		public const int REDUCED_SIZE_VALID_MASK;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_VALID_MASK")]
+		public const int VALID_MASK;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_VALID_VALUE")]
+		public const int VALID_VALUE;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTCP_VERSION")]
+		public const int VERSION;
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static uint64 ntp_to_unix (uint64 ntptime);
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static uint64 unix_to_ntp (uint64 unixtime);
+	}
+	namespace RTP {
+		[CCode (cheader_filename = "gst/rtp/rtp.h", type_id = "gst_rtp_base_audio_payload_get_type ()")]
+		[GIR (name = "RTPBaseAudioPayload")]
+		public class BaseAudioPayload : Gst.RTP.BasePayload {
+			public Gst.ClockTime base_ts;
+			public int frame_duration;
+			public int frame_size;
+			public int sample_size;
+			[CCode (has_construct_function = false)]
+			protected BaseAudioPayload ();
+			public Gst.FlowReturn flush (uint payload_len, Gst.ClockTime timestamp);
+			public Gst.Base.Adapter get_adapter ();
+			public Gst.FlowReturn push ([CCode (array_length_cname = "payload_len", array_length_pos = 1.5, array_length_type = "guint")] uint8[] data, Gst.ClockTime timestamp);
+			public void set_frame_based ();
+			public void set_frame_options (int frame_duration, int frame_size);
+			public void set_sample_based ();
+			public void set_sample_options (int sample_size);
+			public void set_samplebits_options (int sample_size);
+			[NoAccessorMethod]
+			public bool buffer_list { get; set; }
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", type_id = "gst_rtp_base_depayload_get_type ()")]
+		[GIR (name = "RTPBaseDepayload")]
+		public abstract class BaseDepayload : Gst.Element {
+			public uint clock_rate;
+			public bool need_newsegment;
+			public weak Gst.Segment segment;
+			public weak Gst.Pad sinkpad;
+			public weak Gst.Pad srcpad;
+			[CCode (has_construct_function = false)]
+			protected BaseDepayload ();
+			[NoWrapper]
+			public virtual bool handle_event (Gst.Event event);
+			[Version (since = "1.16")]
+			public bool is_source_info_enabled ();
+			[NoWrapper]
+			public virtual bool packet_lost (Gst.Event event);
+			[NoWrapper]
+			public virtual Gst.Buffer process (Gst.Buffer @in);
+			[NoWrapper]
+			public virtual Gst.Buffer process_rtp_packet (Gst.RTP.Buffer rtp_buffer);
+			public Gst.FlowReturn push (Gst.Buffer out_buf);
+			public Gst.FlowReturn push_list (Gst.BufferList out_list);
+			[NoWrapper]
+			public virtual bool set_caps (Gst.Caps caps);
+			[Version (since = "1.16")]
+			public void set_source_info_enabled (bool enable);
+			[NoAccessorMethod]
+			[Version (since = "1.20")]
+			public bool auto_header_extension { get; set; }
+			[NoAccessorMethod]
+			[Version (since = "1.18")]
+			public int max_reorder { get; set; }
+			[NoAccessorMethod]
+			[Version (since = "1.16")]
+			public bool source_info { get; set; }
+			[NoAccessorMethod]
+			public Gst.Structure stats { owned get; }
+			[Version (since = "1.20")]
+			public signal void add_extension (owned Gst.RTP.HeaderExtension ext);
+			[Version (since = "1.20")]
+			public signal void clear_extensions ();
+			[Version (since = "1.20")]
+			public signal Gst.RTP.HeaderExtension request_extension (uint ext_id, string? ext_uri);
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", type_id = "gst_rtp_base_payload_get_type ()")]
+		[GIR (name = "RTPBasePayload")]
+		public abstract class BasePayload : Gst.Element {
+			[CCode (has_construct_function = false)]
+			protected BasePayload ();
+			[Version (since = "1.16")]
+			public Gst.Buffer allocate_output_buffer (uint payload_len, uint8 pad_len, uint8 csrc_count);
+			[NoWrapper]
+			public virtual Gst.Caps get_caps (Gst.Pad pad, Gst.Caps filter);
+			[Version (since = "1.16")]
+			public uint get_source_count (Gst.Buffer buffer);
+			[NoWrapper]
+			public virtual Gst.FlowReturn handle_buffer (Gst.Buffer buffer);
+			public bool is_filled (uint size, Gst.ClockTime duration);
+			[Version (since = "1.16")]
+			public bool is_source_info_enabled ();
+			public Gst.FlowReturn push (Gst.Buffer buffer);
+			public Gst.FlowReturn push_list (Gst.BufferList list);
+			[NoWrapper]
+			public virtual bool query (Gst.Pad pad, Gst.Query query);
+			[NoWrapper]
+			public virtual bool set_caps (Gst.Caps caps);
+			public void set_options (string media, bool @dynamic, string encoding_name, uint32 clock_rate);
+			[Version (since = "1.20")]
+			public bool set_outcaps_structure (Gst.Structure? s);
+			[Version (since = "1.16")]
+			public void set_source_info_enabled (bool enable);
+			[NoWrapper]
+			public virtual bool sink_event (Gst.Event event);
+			[NoWrapper]
+			public virtual bool src_event (Gst.Event event);
+			[NoAccessorMethod]
+			[Version (since = "1.20")]
+			public bool auto_header_extension { get; set; }
+			[NoAccessorMethod]
+			public int64 max_ptime { get; set; }
+			[NoAccessorMethod]
+			public int64 min_ptime { get; set; }
+			[NoAccessorMethod]
+			public uint mtu { get; set; }
+			[NoAccessorMethod]
+			[Version (since = "1.16")]
+			public bool onvif_no_rate_control { get; set; }
+			[NoAccessorMethod]
+			public bool perfect_rtptime { get; set; }
+			[NoAccessorMethod]
+			public uint pt { get; set; }
+			[NoAccessorMethod]
+			public int64 ptime_multiple { get; set; }
+			[NoAccessorMethod]
+			[Version (since = "1.18")]
+			public bool scale_rtptime { get; set; }
+			[NoAccessorMethod]
+			public uint seqnum { get; }
+			[NoAccessorMethod]
+			public int seqnum_offset { get; set; }
+			[NoAccessorMethod]
+			[Version (since = "1.16")]
+			public bool source_info { get; set; }
+			[NoAccessorMethod]
+			public uint ssrc { get; set; }
+			[NoAccessorMethod]
+			public Gst.Structure stats { owned get; }
+			[NoAccessorMethod]
+			public uint timestamp { get; }
+			[NoAccessorMethod]
+			public uint timestamp_offset { get; set; }
+			[Version (since = "1.20")]
+			public signal void add_extension (owned Gst.RTP.HeaderExtension ext);
+			[Version (since = "1.20")]
+			public signal void clear_extensions ();
+			[Version (since = "1.20")]
+			public signal Gst.RTP.HeaderExtension request_extension (uint ext_id, string ext_uri);
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", type_id = "gst_rtp_header_extension_get_type ()")]
+		[GIR (name = "RTPHeaderExtension")]
+		[Version (since = "1.20")]
+		public abstract class HeaderExtension : Gst.Element {
+			public uint ext_id;
+			[CCode (has_construct_function = false)]
+			protected HeaderExtension ();
+			public static Gst.RTP.HeaderExtension? create_from_uri (string uri);
+			public uint get_id ();
+			public virtual size_t get_max_size (Gst.Buffer input_meta);
+			public string get_sdp_caps_field_name ();
+			public virtual Gst.RTP.HeaderExtensionFlags get_supported_flags ();
+			public unowned string get_uri ();
+			public virtual bool read (Gst.RTP.HeaderExtensionFlags read_flags, [CCode (array_length_cname = "size", array_length_pos = 2.5, array_length_type = "gsize", type = "const guint8*")] uint8[] data, Gst.Buffer buffer);
+			public virtual bool set_attributes_from_caps (Gst.Caps caps);
+			public bool set_attributes_from_caps_simple_sdp (Gst.Caps caps);
+			public virtual bool set_caps_from_attributes (Gst.Caps caps);
+			public bool set_caps_from_attributes_simple_sdp (Gst.Caps caps);
+			public void set_id (uint ext_id);
+			public virtual bool set_non_rtp_sink_caps (Gst.Caps caps);
+			[CCode (cname = "gst_rtp_header_extension_class_set_uri")]
+			public class void set_uri (string uri);
+			public void set_wants_update_non_rtp_src_caps (bool state);
+			public virtual bool update_non_rtp_src_caps (Gst.Caps caps);
+			public virtual size_t write (Gst.Buffer input_meta, Gst.RTP.HeaderExtensionFlags write_flags, Gst.Buffer output, [CCode (array_length_cname = "size", array_length_pos = 4.1, array_length_type = "gsize", type = "guint8*")] uint8[] data);
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+		[GIR (name = "RTPBuffer")]
+		public struct Buffer {
+			public weak Gst.Buffer buffer;
+			public uint state;
+			[CCode (array_length = false)]
+			public weak void* data[4];
+			[CCode (array_length = false)]
+			public weak size_t size[4];
+			public bool add_extension_onebyte_header (uint8 id, [CCode (array_length_cname = "size", array_length_pos = 2.1, array_length_type = "guint")] uint8[] data);
+			public bool add_extension_twobytes_header (uint8 appbits, uint8 id, [CCode (array_length_cname = "size", array_length_pos = 3.1, array_length_type = "guint")] uint8[] data);
+			[CCode (cname = "gst_buffer_add_rtp_source_meta")]
+			[Version (since = "1.16")]
+			public static unowned Gst.RTP.SourceMeta? add_rtp_source_meta (Gst.Buffer buffer, uint32? ssrc, uint32? csrc, uint csrc_count);
+			public static void allocate_data (Gst.Buffer buffer, uint payload_len, uint8 pad_len, uint8 csrc_count);
+			public static uint calc_header_len (uint8 csrc_count);
+			public static uint calc_packet_len (uint payload_len, uint8 pad_len, uint8 csrc_count);
+			public static uint calc_payload_len (uint packet_len, uint8 pad_len, uint8 csrc_count);
+			public static int compare_seqnum (uint16 seqnum1, uint16 seqnum2);
+			public static uint32 default_clock_rate (uint8 payload_type);
+			public static uint64 ext_timestamp (ref uint64 exttimestamp, uint32 timestamp);
+			public uint32 get_csrc (uint8 idx);
+			public uint8 get_csrc_count ();
+			public bool get_extension ();
+			[Version (since = "1.2")]
+			public GLib.Bytes get_extension_bytes (out uint16 bits);
+			public bool get_extension_data (out uint16 bits, [CCode (array_length = false)] out unowned uint8[] data, out uint wordlen);
+			public bool get_extension_onebyte_header (uint8 id, uint nth, [CCode (array_length_cname = "size", array_length_pos = 3.1, array_length_type = "guint")] out unowned uint8[] data);
+			[Version (since = "1.18")]
+			public static bool get_extension_onebyte_header_from_bytes (GLib.Bytes bytes, uint16 bit_pattern, uint8 id, uint nth, [CCode (array_length_cname = "size", array_length_pos = 5.1, array_length_type = "guint")] out unowned uint8[] data);
+			public bool get_extension_twobytes_header (out uint8 appbits, uint8 id, uint nth, [CCode (array_length_cname = "size", array_length_pos = 4.1, array_length_type = "guint")] out unowned uint8[] data);
+			public uint get_header_len ();
+			public bool get_marker ();
+			public uint get_packet_len ();
+			public bool get_padding ();
+			[CCode (array_length = false)]
+			public unowned uint8[] get_payload ();
+			public Gst.Buffer get_payload_buffer ();
+			[Version (since = "1.2")]
+			public GLib.Bytes get_payload_bytes ();
+			public uint get_payload_len ();
+			public Gst.Buffer get_payload_subbuffer (uint offset, uint len);
+			public uint8 get_payload_type ();
+			[CCode (cname = "gst_buffer_get_rtp_source_meta")]
+			[Version (since = "1.16")]
+			public static unowned Gst.RTP.SourceMeta? get_rtp_source_meta (Gst.Buffer buffer);
+			public uint16 get_seq ();
+			public uint32 get_ssrc ();
+			public uint32 get_timestamp ();
+			public uint8 get_version ();
+			public static bool map (Gst.Buffer buffer, Gst.MapFlags flags, out Gst.RTP.Buffer rtp);
+			public static Gst.Buffer new_allocate (uint payload_len, uint8 pad_len, uint8 csrc_count);
+			public static Gst.Buffer new_allocate_len (uint packet_len, uint8 pad_len, uint8 csrc_count);
+			public static Gst.Buffer new_copy_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "gsize")] uint8[] data);
+			public static Gst.Buffer new_take_data ([CCode (array_length_cname = "len", array_length_pos = 1.1, array_length_type = "gsize")] owned uint8[] data);
+			public void pad_to (uint len);
+			public void set_csrc (uint8 idx, uint32 csrc);
+			public void set_extension (bool extension);
+			public bool set_extension_data (uint16 bits, uint16 length);
+			public void set_marker (bool marker);
+			public void set_packet_len (uint len);
+			public void set_padding (bool padding);
+			public void set_payload_type (uint8 payload_type);
+			public void set_seq (uint16 seq);
+			public void set_ssrc (uint32 ssrc);
+			public void set_timestamp (uint32 timestamp);
+			public void set_version (uint8 version);
+			public void unmap ();
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+		[GIR (name = "RTPPayloadInfo")]
+		public struct PayloadInfo {
+			public uint8 payload_type;
+			public weak string media;
+			public weak string encoding_name;
+			public uint clock_rate;
+			public weak string encoding_parameters;
+			public uint bitrate;
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", has_type_id = false)]
+		[GIR (name = "RTPSourceMeta")]
+		[Version (since = "1.16")]
+		public struct SourceMeta {
+			public Gst.Meta meta;
+			public uint32 ssrc;
+			public bool ssrc_valid;
+			[CCode (array_length = false)]
+			public weak uint32 csrc[15];
+			public uint csrc_count;
+			public bool append_csrc ([CCode (array_length_cname = "csrc_count", array_length_pos = 1.1, array_length_type = "guint", type = "const guint32*")] uint32[] csrc);
+			public uint get_source_count ();
+			public bool set_ssrc (uint32? ssrc);
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_BUFFER_FLAG_", type_id = "gst_rtp_buffer_flags_get_type ()")]
+		[Flags]
+		[GIR (name = "RTPBufferFlags")]
+		[Version (since = "1.10")]
+		public enum BufferFlags {
+			RETRANSMISSION,
+			REDUNDANT,
+			LAST
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_BUFFER_MAP_FLAG_", type_id = "gst_rtp_buffer_map_flags_get_type ()")]
+		[Flags]
+		[GIR (name = "RTPBufferMapFlags")]
+		[Version (since = "1.6.1")]
+		public enum BufferMapFlags {
+			SKIP_PADDING,
+			LAST
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_HEADER_EXTENSION_", type_id = "gst_rtp_header_extension_flags_get_type ()")]
+		[Flags]
+		[GIR (name = "RTPHeaderExtensionFlags")]
+		[Version (since = "1.20")]
+		public enum HeaderExtensionFlags {
+			ONE_BYTE,
+			TWO_BYTE
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_PAYLOAD_", type_id = "gst_rtp_payload_get_type ()")]
+		[GIR (name = "RTPPayload")]
+		public enum Payload {
+			PCMU,
+			@1016,
+			G721,
+			GSM,
+			G723,
+			DVI4_8000,
+			DVI4_16000,
+			LPC,
+			PCMA,
+			G722,
+			L16_STEREO,
+			L16_MONO,
+			QCELP,
+			CN,
+			MPA,
+			G728,
+			DVI4_11025,
+			DVI4_22050,
+			G729,
+			CELLB,
+			JPEG,
+			NV,
+			H261,
+			MPV,
+			MP2T,
+			H263;
+			public const string @1016_STRING;
+			public const string CELLB_STRING;
+			public const string CN_STRING;
+			public const string DVI4_11025_STRING;
+			public const string DVI4_16000_STRING;
+			public const string DVI4_22050_STRING;
+			public const string DVI4_8000_STRING;
+			public const string DYNAMIC_STRING;
+			public const string G721_STRING;
+			public const string G722_STRING;
+			public const int G723_53;
+			public const string G723_53_STRING;
+			public const int G723_63;
+			public const string G723_63_STRING;
+			public const string G723_STRING;
+			public const string G728_STRING;
+			public const string G729_STRING;
+			public const string GSM_STRING;
+			public const string H261_STRING;
+			public const string H263_STRING;
+			public const string JPEG_STRING;
+			public const string L16_MONO_STRING;
+			public const string L16_STEREO_STRING;
+			public const string LPC_STRING;
+			public const string MP2T_STRING;
+			public const string MPA_STRING;
+			public const string MPV_STRING;
+			public const string NV_STRING;
+			public const string PCMA_STRING;
+			public const string PCMU_STRING;
+			public const string QCELP_STRING;
+			public const int TS41;
+			public const string TS41_STRING;
+			public const int TS48;
+			public const string TS48_STRING;
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cprefix = "GST_RTP_PROFILE_", type_id = "gst_rtp_profile_get_type ()")]
+		[GIR (name = "RTPProfile")]
+		[Version (since = "1.6")]
+		public enum Profile {
+			UNKNOWN,
+			AVP,
+			SAVP,
+			AVPF,
+			SAVPF
+		}
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_BASE")]
+		public const string HDREXT_BASE;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_ELEMENT_CLASS")]
+		[Version (since = "1.20")]
+		public const string HDREXT_ELEMENT_CLASS;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_NTP_56")]
+		public const string HDREXT_NTP_56;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_NTP_56_SIZE")]
+		public const int HDREXT_NTP_56_SIZE;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_NTP_64")]
+		public const string HDREXT_NTP_64;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HDREXT_NTP_64_SIZE")]
+		public const int HDREXT_NTP_64_SIZE;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_HEADER_EXTENSION_URI_METADATA_KEY")]
+		[Version (since = "1.20")]
+		public const string HEADER_EXTENSION_URI_METADATA_KEY;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_SOURCE_META_MAX_CSRC_COUNT")]
+		public const int SOURCE_META_MAX_CSRC_COUNT;
+		[CCode (cheader_filename = "gst/rtp/rtp.h", cname = "GST_RTP_VERSION")]
+		public const int VERSION;
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		[Version (since = "1.20")]
+		public static GLib.List<Gst.RTP.HeaderExtension> get_header_extension_list ();
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static bool hdrext_get_ntp_56 ([CCode (array_length_cname = "size", array_length_pos = 1.5, array_length_type = "guint")] uint8[] data, out uint64 ntptime);
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static bool hdrext_get_ntp_64 ([CCode (array_length_cname = "size", array_length_pos = 1.5, array_length_type = "guint")] uint8[] data, out uint64 ntptime);
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static bool hdrext_set_ntp_56 (void* data, uint size, uint64 ntptime);
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static bool hdrext_set_ntp_64 (void* data, uint size, uint64 ntptime);
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static unowned Gst.RTP.PayloadInfo? payload_info_for_name (string media, string encoding_name);
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static unowned Gst.RTP.PayloadInfo? payload_info_for_pt (uint8 payload_type);
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static GLib.Type source_meta_api_get_type ();
+		[CCode (cheader_filename = "gst/rtp/rtp.h")]
+		public static unowned Gst.MetaInfo? source_meta_get_info ();
+	}
+}