diff --git a/ffserver.c b/ffserver.c
index 8bfea66c2a..11088601fe 100644
--- a/ffserver.c
+++ b/ffserver.c
@@ -32,7 +32,7 @@
 #include "libavformat/avformat.h"
 #include "libavformat/network.h"
 #include "libavformat/os_support.h"
-#include "libavformat/rtp.h"
+#include "libavformat/rtpdec.h"
 #include "libavformat/rtsp.h"
 #include "libavutil/avstring.h"
 #include "libavutil/random.h"
diff --git a/libavformat/rdt.c b/libavformat/rdt.c
index ce8903dc4c..aeb35c7135 100644
--- a/libavformat/rdt.c
+++ b/libavformat/rdt.c
@@ -27,7 +27,7 @@
 
 #include "avformat.h"
 #include "libavutil/avstring.h"
-#include "rtp.h"
+#include "rtpdec.h"
 #include "rdt.h"
 #include "libavutil/base64.h"
 #include "libavutil/md5.h"
diff --git a/libavformat/rdt.h b/libavformat/rdt.h
index e24a0d516b..1592c2f02d 100644
--- a/libavformat/rdt.h
+++ b/libavformat/rdt.h
@@ -24,7 +24,7 @@
 
 #include <stdint.h>
 #include "avformat.h"
-#include "rtp.h"
+#include "rtpdec.h"
 
 typedef struct RDTDemuxContext RDTDemuxContext;
 
diff --git a/libavformat/rtp.h b/libavformat/rtp.h
index c66a0c7d71..834fb1b0d3 100644
--- a/libavformat/rtp.h
+++ b/libavformat/rtp.h
@@ -1,7 +1,6 @@
 /*
  * RTP definitions
  * Copyright (c) 2002 Fabrice Bellard
- * Copyright (c) 2006 Ryan Martell <rdm4@martellventures.com>
  *
  * This file is part of FFmpeg.
  *
@@ -23,65 +22,12 @@
 #define AVFORMAT_RTP_H
 
 #include "libavcodec/avcodec.h"
-#include "avformat.h"
 
-/** Structure listing useful vars to parse RTP packet payload*/
-typedef struct rtp_payload_data
-{
-    int sizelength;
-    int indexlength;
-    int indexdeltalength;
-    int profile_level_id;
-    int streamtype;
-    int objecttype;
-    char *mode;
-
-    /** mpeg 4 AU headers */
-    struct AUHeaders {
-        int size;
-        int index;
-        int cts_flag;
-        int cts;
-        int dts_flag;
-        int dts;
-        int rap_flag;
-        int streamstate;
-    } *au_headers;
-    int nb_au_headers;
-    int au_headers_length_bytes;
-    int cur_au_index;
-} RTPPayloadData;
-
-typedef struct PayloadContext PayloadContext;
-typedef struct RTPDynamicProtocolHandler_s RTPDynamicProtocolHandler;
-
-#define RTP_MIN_PACKET_LENGTH 12
 #define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
 
-int rtp_get_codec_info(AVCodecContext *codec, int payload_type);
-
 /** return < 0 if unknown payload type */
 int rtp_get_payload_type(AVCodecContext *codec);
 
-typedef struct RTPDemuxContext RTPDemuxContext;
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data);
-void rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
-                                    RTPDynamicProtocolHandler *handler);
-int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
-                     const uint8_t *buf, int len);
-void rtp_parse_close(RTPDemuxContext *s);
-
-int rtp_get_local_port(URLContext *h);
-int rtp_set_remote_url(URLContext *h, const char *uri);
-void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd);
-
-/**
- * some rtp servers assume client is dead if they don't hear from them...
- * so we send a Receiver Report to the provided ByteIO context
- * (we don't have access to the rtcp handle from here)
- */
-int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count);
-
 #define RTP_PT_PRIVATE 96
 #define RTP_VERSION 2
 #define RTP_MAX_SDES 256   /**< maximum text length for SDES */
@@ -90,111 +36,4 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count);
 #define RTCP_TX_RATIO_NUM 5
 #define RTCP_TX_RATIO_DEN 1000
 
-// these statistics are used for rtcp receiver reports...
-typedef struct {
-    uint16_t max_seq;           ///< highest sequence number seen
-    uint32_t cycles;            ///< shifted count of sequence number cycles
-    uint32_t base_seq;          ///< base sequence number
-    uint32_t bad_seq;           ///< last bad sequence number + 1
-    int probation;              ///< sequence packets till source is valid
-    int received;               ///< packets received
-    int expected_prior;         ///< packets expected in last interval
-    int received_prior;         ///< packets received in last interval
-    uint32_t transit;           ///< relative transit time for previous packet
-    uint32_t jitter;            ///< estimated jitter.
-} RTPStatistics;
-
-/**
- * Packet parsing for "private" payloads in the RTP specs.
- *
- * @param ctx RTSP demuxer context
- * @param s stream context
- * @param st stream that this packet belongs to
- * @param pkt packet in which to write the parsed data
- * @param timestamp pointer in which to write the timestamp of this RTP packet
- * @param buf pointer to raw RTP packet data
- * @param len length of buf
- * @param flags flags from the RTP packet header (PKT_FLAG_*)
- */
-typedef int (*DynamicPayloadPacketHandlerProc) (AVFormatContext *ctx,
-                                                PayloadContext *s,
-                                                AVStream *st,
-                                                AVPacket * pkt,
-                                                uint32_t *timestamp,
-                                                const uint8_t * buf,
-                                                int len, int flags);
-
-struct RTPDynamicProtocolHandler_s {
-    // fields from AVRtpDynamicPayloadType_s
-    const char enc_name[50];    /* XXX: still why 50 ? ;-) */
-    enum CodecType codec_type;
-    enum CodecID codec_id;
-
-    // may be null
-    int (*parse_sdp_a_line) (AVFormatContext *s,
-                             int st_index,
-                             PayloadContext *priv_data,
-                             const char *line); ///< Parse the a= line from the sdp field
-    PayloadContext *(*open) (); ///< allocate any data needed by the rtp parsing for this dynamic data.
-    void (*close)(PayloadContext *protocol_data); ///< free any data needed by the rtp parsing for this dynamic data.
-    DynamicPayloadPacketHandlerProc parse_packet; ///< parse handler for this dynamic packet.
-
-    struct RTPDynamicProtocolHandler_s *next;
-};
-
-// moved out of rtp.c, because the h264 decoder needs to know about this structure..
-struct RTPDemuxContext {
-    AVFormatContext *ic;
-    AVStream *st;
-    int payload_type;
-    uint32_t ssrc;
-    uint16_t seq;
-    uint32_t timestamp;
-    uint32_t base_timestamp;
-    uint32_t cur_timestamp;
-    int max_payload_size;
-    struct MpegTSContext *ts;   /* only used for MP2T payloads */
-    int read_buf_index;
-    int read_buf_size;
-    int num_frames;
-    /* used to send back RTCP RR */
-    URLContext *rtp_ctx;
-    char hostname[256];
-
-    RTPStatistics statistics; ///< Statistics for this stream (used by RTCP receiver reports)
-
-    /* rtcp sender statistics receive */
-    int64_t last_rtcp_ntp_time;    // TODO: move into statistics
-    int64_t first_rtcp_ntp_time;   // TODO: move into statistics
-    uint32_t last_rtcp_timestamp;  // TODO: move into statistics
-
-    /* rtcp sender statistics */
-    unsigned int packet_count;     // TODO: move into statistics (outgoing)
-    unsigned int octet_count;      // TODO: move into statistics (outgoing)
-    unsigned int last_octet_count; // TODO: move into statistics (outgoing)
-    int first_packet;
-    /* buffer for output */
-    uint8_t buf[RTP_MAX_PACKET_LENGTH];
-    uint8_t *buf_ptr;
-
-    /* special infos for au headers parsing */
-    RTPPayloadData *rtp_payload_data; // TODO: Move into dynamic payload handlers
-
-    /* dynamic payload stuff */
-    DynamicPayloadPacketHandlerProc parse_packet;     ///< This is also copied from the dynamic protocol handler structure
-    PayloadContext *dynamic_protocol_context;        ///< This is a copy from the values setup from the sdp parsing, in rtsp.c don't free me.
-    int max_frames_per_packet;
-};
-
-extern RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler;
-void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler);
-
-int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size); ///< from rtsp.c, but used by rtp dynamic protocol handlers.
-
-void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m);
-const char *ff_rtp_enc_name(int payload_type);
-enum CodecID ff_rtp_codec_id(const char *buf, enum CodecType codec_type);
-
-void av_register_rtp_dynamic_payload_handlers(void);
-
 #endif /* AVFORMAT_RTP_H */
diff --git a/libavformat/rtp_aac.c b/libavformat/rtp_aac.c
index 71ca9c6617..60097f125b 100644
--- a/libavformat/rtp_aac.c
+++ b/libavformat/rtp_aac.c
@@ -20,14 +20,14 @@
 
 #include "avformat.h"
 #include "rtp_aac.h"
-#include "rtp.h"
+#include "rtpenc.h"
 
 #define MAX_FRAMES_PER_PACKET (s->max_frames_per_packet ? s->max_frames_per_packet : 5)
 #define MAX_AU_HEADERS_SIZE (2 + 2 * MAX_FRAMES_PER_PACKET)
 
 void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
 {
-    RTPDemuxContext *s = s1->priv_data;
+    RTPMuxContext *s = s1->priv_data;
     int len, max_packet_size;
     uint8_t *p;
 
diff --git a/libavformat/rtp_h264.c b/libavformat/rtp_h264.c
index 4b89ea61af..85133be2f8 100644
--- a/libavformat/rtp_h264.c
+++ b/libavformat/rtp_h264.c
@@ -46,7 +46,7 @@
 #include "network.h"
 #include <assert.h>
 
-#include "rtp.h"
+#include "rtpdec.h"
 #include "rtp_h264.h"
 
 /**
diff --git a/libavformat/rtp_h264.h b/libavformat/rtp_h264.h
index 94eac47aea..31fd40ba66 100644
--- a/libavformat/rtp_h264.h
+++ b/libavformat/rtp_h264.h
@@ -22,7 +22,7 @@
 #ifndef AVFORMAT_RTP_H264_H
 #define AVFORMAT_RTP_H264_H
 
-#include "rtp.h"
+#include "rtpdec.h"
 
 extern RTPDynamicProtocolHandler ff_h264_dynamic_handler;
 void ff_rtp_send_h264(AVFormatContext *s1, const uint8_t *buf1, int size);
diff --git a/libavformat/rtp_mpv.c b/libavformat/rtp_mpv.c
index 2221dd2489..f3f4501367 100644
--- a/libavformat/rtp_mpv.c
+++ b/libavformat/rtp_mpv.c
@@ -22,13 +22,13 @@
 
 #include "libavcodec/mpegvideo.h"
 #include "avformat.h"
-#include "rtp.h"
+#include "rtpenc.h"
 
 /* NOTE: a single frame must be passed with sequence header if
    needed. XXX: use slices. */
 void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size)
 {
-    RTPDemuxContext *s = s1->priv_data;
+    RTPMuxContext *s = s1->priv_data;
     int len, h, max_packet_size;
     uint8_t *q;
     int begin_of_slice, end_of_slice, frame_type, temporal_reference;
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 8297b1eb34..5b7d63db12 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -29,7 +29,7 @@
 #include <unistd.h>
 #include "network.h"
 
-#include "rtp.h"
+#include "rtpdec.h"
 #include "rtp_h264.h"
 
 //#define DEBUG
diff --git a/libavformat/rtpdec.h b/libavformat/rtpdec.h
new file mode 100644
index 0000000000..1eeb0ba968
--- /dev/null
+++ b/libavformat/rtpdec.h
@@ -0,0 +1,187 @@
+/*
+ * RTP demuxer definitions
+ * Copyright (c) 2002 Fabrice Bellard
+ * Copyright (c) 2006 Ryan Martell <rdm4@martellventures.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+#ifndef AVFORMAT_RTPDEC_H
+#define AVFORMAT_RTPDEC_H
+
+#include "libavcodec/avcodec.h"
+#include "avformat.h"
+#include "rtp.h"
+
+/** Structure listing useful vars to parse RTP packet payload*/
+typedef struct rtp_payload_data
+{
+    int sizelength;
+    int indexlength;
+    int indexdeltalength;
+    int profile_level_id;
+    int streamtype;
+    int objecttype;
+    char *mode;
+
+    /** mpeg 4 AU headers */
+    struct AUHeaders {
+        int size;
+        int index;
+        int cts_flag;
+        int cts;
+        int dts_flag;
+        int dts;
+        int rap_flag;
+        int streamstate;
+    } *au_headers;
+    int nb_au_headers;
+    int au_headers_length_bytes;
+    int cur_au_index;
+} RTPPayloadData;
+
+typedef struct PayloadContext PayloadContext;
+typedef struct RTPDynamicProtocolHandler_s RTPDynamicProtocolHandler;
+
+#define RTP_MIN_PACKET_LENGTH 12
+
+int rtp_get_codec_info(AVCodecContext *codec, int payload_type);
+
+typedef struct RTPDemuxContext RTPDemuxContext;
+RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data);
+void rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
+                                    RTPDynamicProtocolHandler *handler);
+int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
+                     const uint8_t *buf, int len);
+void rtp_parse_close(RTPDemuxContext *s);
+
+int rtp_get_local_port(URLContext *h);
+int rtp_set_remote_url(URLContext *h, const char *uri);
+void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd);
+
+/**
+ * some rtp servers assume client is dead if they don't hear from them...
+ * so we send a Receiver Report to the provided ByteIO context
+ * (we don't have access to the rtcp handle from here)
+ */
+int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count);
+
+// these statistics are used for rtcp receiver reports...
+typedef struct {
+    uint16_t max_seq;           ///< highest sequence number seen
+    uint32_t cycles;            ///< shifted count of sequence number cycles
+    uint32_t base_seq;          ///< base sequence number
+    uint32_t bad_seq;           ///< last bad sequence number + 1
+    int probation;              ///< sequence packets till source is valid
+    int received;               ///< packets received
+    int expected_prior;         ///< packets expected in last interval
+    int received_prior;         ///< packets received in last interval
+    uint32_t transit;           ///< relative transit time for previous packet
+    uint32_t jitter;            ///< estimated jitter.
+} RTPStatistics;
+
+/**
+ * Packet parsing for "private" payloads in the RTP specs.
+ *
+ * @param ctx RTSP demuxer context
+ * @param s stream context
+ * @param st stream that this packet belongs to
+ * @param pkt packet in which to write the parsed data
+ * @param timestamp pointer in which to write the timestamp of this RTP packet
+ * @param buf pointer to raw RTP packet data
+ * @param len length of buf
+ * @param flags flags from the RTP packet header (PKT_FLAG_*)
+ */
+typedef int (*DynamicPayloadPacketHandlerProc) (AVFormatContext *ctx,
+                                                PayloadContext *s,
+                                                AVStream *st,
+                                                AVPacket * pkt,
+                                                uint32_t *timestamp,
+                                                const uint8_t * buf,
+                                                int len, int flags);
+
+struct RTPDynamicProtocolHandler_s {
+    // fields from AVRtpDynamicPayloadType_s
+    const char enc_name[50];    /* XXX: still why 50 ? ;-) */
+    enum CodecType codec_type;
+    enum CodecID codec_id;
+
+    // may be null
+    int (*parse_sdp_a_line) (AVFormatContext *s,
+                             int st_index,
+                             PayloadContext *priv_data,
+                             const char *line); ///< Parse the a= line from the sdp field
+    PayloadContext *(*open) (); ///< allocate any data needed by the rtp parsing for this dynamic data.
+    void (*close)(PayloadContext *protocol_data); ///< free any data needed by the rtp parsing for this dynamic data.
+    DynamicPayloadPacketHandlerProc parse_packet; ///< parse handler for this dynamic packet.
+
+    struct RTPDynamicProtocolHandler_s *next;
+};
+
+// moved out of rtp.c, because the h264 decoder needs to know about this structure..
+struct RTPDemuxContext {
+    AVFormatContext *ic;
+    AVStream *st;
+    int payload_type;
+    uint32_t ssrc;
+    uint16_t seq;
+    uint32_t timestamp;
+    uint32_t base_timestamp;
+    uint32_t cur_timestamp;
+    int max_payload_size;
+    struct MpegTSContext *ts;   /* only used for MP2T payloads */
+    int read_buf_index;
+    int read_buf_size;
+    /* used to send back RTCP RR */
+    URLContext *rtp_ctx;
+    char hostname[256];
+
+    RTPStatistics statistics; ///< Statistics for this stream (used by RTCP receiver reports)
+
+    /* rtcp sender statistics receive */
+    int64_t last_rtcp_ntp_time;    // TODO: move into statistics
+    int64_t first_rtcp_ntp_time;   // TODO: move into statistics
+    uint32_t last_rtcp_timestamp;  // TODO: move into statistics
+
+    /* rtcp sender statistics */
+    unsigned int packet_count;     // TODO: move into statistics (outgoing)
+    unsigned int octet_count;      // TODO: move into statistics (outgoing)
+    unsigned int last_octet_count; // TODO: move into statistics (outgoing)
+    int first_packet;
+    /* buffer for output */
+    uint8_t buf[RTP_MAX_PACKET_LENGTH];
+    uint8_t *buf_ptr;
+
+    /* special infos for au headers parsing */
+    RTPPayloadData *rtp_payload_data; // TODO: Move into dynamic payload handlers
+
+    /* dynamic payload stuff */
+    DynamicPayloadPacketHandlerProc parse_packet;     ///< This is also copied from the dynamic protocol handler structure
+    PayloadContext *dynamic_protocol_context;        ///< This is a copy from the values setup from the sdp parsing, in rtsp.c don't free me.
+    int max_frames_per_packet;
+};
+
+extern RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler;
+void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler);
+
+int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size); ///< from rtsp.c, but used by rtp dynamic protocol handlers.
+
+const char *ff_rtp_enc_name(int payload_type);
+enum CodecID ff_rtp_codec_id(const char *buf, enum CodecType codec_type);
+
+void av_register_rtp_dynamic_payload_handlers(void);
+
+#endif /* AVFORMAT_RTPDEC_H */
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index 32b160e08f..62ee905d19 100644
--- a/libavformat/rtpenc.c
+++ b/libavformat/rtpenc.c
@@ -26,7 +26,7 @@
 #include <unistd.h>
 #include "network.h"
 
-#include "rtp.h"
+#include "rtpenc.h"
 #include "rtp_mpv.h"
 #include "rtp_aac.h"
 #include "rtp_h264.h"
@@ -44,7 +44,7 @@ static uint64_t ntp_time(void)
 
 static int rtp_write_header(AVFormatContext *s1)
 {
-    RTPDemuxContext *s = s1->priv_data;
+    RTPMuxContext *s = s1->priv_data;
     int payload_type, max_packet_size, n;
     AVStream *st;
 
@@ -117,7 +117,7 @@ static int rtp_write_header(AVFormatContext *s1)
 /* send an rtcp sender report packet */
 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
 {
-    RTPDemuxContext *s = s1->priv_data;
+    RTPMuxContext *s = s1->priv_data;
     uint32_t rtp_ts;
 
     dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
@@ -142,7 +142,7 @@ static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
    must update the timestamp itself */
 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
 {
-    RTPDemuxContext *s = s1->priv_data;
+    RTPMuxContext *s = s1->priv_data;
 
     dprintf(s1, "rtp_send_data size=%d\n", len);
 
@@ -166,7 +166,7 @@ void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
 static void rtp_send_samples(AVFormatContext *s1,
                              const uint8_t *buf1, int size, int sample_size)
 {
-    RTPDemuxContext *s = s1->priv_data;
+    RTPMuxContext *s = s1->priv_data;
     int len, max_packet_size, n;
 
     max_packet_size = (s->max_payload_size / sample_size) * sample_size;
@@ -194,7 +194,7 @@ static void rtp_send_samples(AVFormatContext *s1,
 static void rtp_send_mpegaudio(AVFormatContext *s1,
                                const uint8_t *buf1, int size)
 {
-    RTPDemuxContext *s = s1->priv_data;
+    RTPMuxContext *s = s1->priv_data;
     int len, count, max_packet_size;
 
     max_packet_size = s->max_payload_size;
@@ -246,7 +246,7 @@ static void rtp_send_mpegaudio(AVFormatContext *s1,
 static void rtp_send_raw(AVFormatContext *s1,
                          const uint8_t *buf1, int size)
 {
-    RTPDemuxContext *s = s1->priv_data;
+    RTPMuxContext *s = s1->priv_data;
     int len, max_packet_size;
 
     max_packet_size = s->max_payload_size;
@@ -268,7 +268,7 @@ static void rtp_send_raw(AVFormatContext *s1,
 static void rtp_send_mpegts_raw(AVFormatContext *s1,
                                 const uint8_t *buf1, int size)
 {
-    RTPDemuxContext *s = s1->priv_data;
+    RTPMuxContext *s = s1->priv_data;
     int len, out_len;
 
     while (size >= TS_PACKET_SIZE) {
@@ -291,7 +291,7 @@ static void rtp_send_mpegts_raw(AVFormatContext *s1,
 /* write an RTP packet. 'buf1' must contain a single specific frame. */
 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
 {
-    RTPDemuxContext *s = s1->priv_data;
+    RTPMuxContext *s = s1->priv_data;
     AVStream *st = s1->streams[0];
     int rtcp_bytes;
     int size= pkt->size;
@@ -352,7 +352,7 @@ AVOutputFormat rtp_muxer = {
     NULL_IF_CONFIG_SMALL("RTP output format"),
     NULL,
     NULL,
-    sizeof(RTPDemuxContext),
+    sizeof(RTPMuxContext),
     CODEC_ID_PCM_MULAW,
     CODEC_ID_NONE,
     rtp_write_header,
diff --git a/libavformat/rtpenc.h b/libavformat/rtpenc.h
new file mode 100644
index 0000000000..d3d029fb93
--- /dev/null
+++ b/libavformat/rtpenc.h
@@ -0,0 +1,61 @@
+/*
+ * RTP muxer definitions
+ * Copyright (c) 2002 Fabrice Bellard
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+#ifndef AVFORMAT_RTPENC_H
+#define AVFORMAT_RTPENC_H
+
+#include "avformat.h"
+#include "rtp.h"
+
+struct RTPMuxContext {
+    AVFormatContext *ic;
+    AVStream *st;
+    int payload_type;
+    uint32_t ssrc;
+    uint16_t seq;
+    uint32_t timestamp;
+    uint32_t base_timestamp;
+    uint32_t cur_timestamp;
+    int max_payload_size;
+    int num_frames;
+
+    /* rtcp sender statistics receive */
+    int64_t last_rtcp_ntp_time;    // TODO: move into statistics
+    int64_t first_rtcp_ntp_time;   // TODO: move into statistics
+
+    /* rtcp sender statistics */
+    unsigned int packet_count;     // TODO: move into statistics (outgoing)
+    unsigned int octet_count;      // TODO: move into statistics (outgoing)
+    unsigned int last_octet_count; // TODO: move into statistics (outgoing)
+    int first_packet;
+    /* buffer for output */
+    uint8_t buf[RTP_MAX_PACKET_LENGTH];
+    uint8_t *buf_ptr;
+
+    int max_frames_per_packet;
+};
+
+typedef struct RTPMuxContext RTPMuxContext;
+
+void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m);
+
+void ff_rtp_send_h264(AVFormatContext *s1, const uint8_t *buf1, int size);
+
+#endif /* AVFORMAT_RTPENC_H */
diff --git a/libavformat/rtpenc_h264.c b/libavformat/rtpenc_h264.c
index fabc5584dc..26bd4a96a9 100644
--- a/libavformat/rtpenc_h264.c
+++ b/libavformat/rtpenc_h264.c
@@ -27,11 +27,11 @@
 
 #include "avformat.h"
 #include "avc.h"
-#include "rtp_h264.h"
+#include "rtpenc.h"
 
 static void nal_send(AVFormatContext *s1, const uint8_t *buf, int size, int last)
 {
-    RTPDemuxContext *s = s1->priv_data;
+    RTPMuxContext *s = s1->priv_data;
 
     av_log(s1, AV_LOG_DEBUG, "Sending NAL %x of len %d M=%d\n", buf[0] & 0x1F, size, last);
     if (size <= s->max_payload_size) {
@@ -63,7 +63,7 @@ static void nal_send(AVFormatContext *s1, const uint8_t *buf, int size, int last
 void ff_rtp_send_h264(AVFormatContext *s1, const uint8_t *buf1, int size)
 {
     const uint8_t *r;
-    RTPDemuxContext *s = s1->priv_data;
+    RTPMuxContext *s = s1->priv_data;
 
     s->timestamp = s->cur_timestamp;
     r = ff_avc_find_startcode(buf1, buf1 + size);
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 2d2c3130e8..34f0924d7e 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -34,7 +34,7 @@
 #include "network.h"
 #include "rtsp.h"
 
-#include "rtp.h"
+#include "rtpdec.h"
 #include "rdt.h"
 
 //#define DEBUG
diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h
index 628f0e1808..ca1e53f26b 100644
--- a/libavformat/rtsp.h
+++ b/libavformat/rtsp.h
@@ -24,7 +24,7 @@
 #include <stdint.h>
 #include "avformat.h"
 #include "rtspcodes.h"
-#include "rtp.h"
+#include "rtpdec.h"
 #include "network.h"
 
 enum RTSPLowerTransport {