RTP/RTSP and MPEG4-AAC audio

  - preliminary support for mpeg4-aac rtp payload (no interleaving support)
  - use udp transport as default (makes more sense with rtp, doesn't it ?)
  - some code factorization, so adding support for new rtp payload will be easier
  (I hope ;-)
patch by (Romain DEGEZ: romain degez, smartjog com)

Originally committed as revision 4306 to svn://svn.ffmpeg.org/ffmpeg/trunk

4 files changed, 463 insertions, 182 deletions

diff --git a/libavformat/rtp.c b/libavformat/rtp.c
index 32711af405..d78a777bc6 100644
--- a/libavformat/rtp.c
+++ b/libavformat/rtp.c
@@ -18,6 +18,7 @@
  */
 #include "avformat.h"
 #include "mpegts.h"
+#include "bitstream.h"
 
 #include <unistd.h>
 #include <sys/types.h>
@@ -42,36 +43,146 @@
          'url_open_dyn_packet_buf') 
 */
 
-#define RTP_VERSION 2
-
-#define RTP_MAX_SDES 256   /* maximum text length for SDES */
-
-/* RTCP paquets use 0.5 % of the bandwidth */
-#define RTCP_TX_RATIO_NUM 5
-#define RTCP_TX_RATIO_DEN 1000
-
-typedef enum {
-  RTCP_SR   = 200,
-  RTCP_RR   = 201,
-  RTCP_SDES = 202,
-  RTCP_BYE  = 203,
-  RTCP_APP  = 204
-} rtcp_type_t;
-
-typedef enum {
-  RTCP_SDES_END    =  0,
-  RTCP_SDES_CNAME  =  1,
-  RTCP_SDES_NAME   =  2,
-  RTCP_SDES_EMAIL  =  3,
-  RTCP_SDES_PHONE  =  4,
-  RTCP_SDES_LOC    =  5,
-  RTCP_SDES_TOOL   =  6,
-  RTCP_SDES_NOTE   =  7,
-  RTCP_SDES_PRIV   =  8, 
-  RTCP_SDES_IMG    =  9,
-  RTCP_SDES_DOOR   = 10,
-  RTCP_SDES_SOURCE = 11
-} rtcp_sdes_type_t;
+/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
+AVRtpPayloadType_t AVRtpPayloadTypes[]=
+{
+  {0, "PCMU",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_MULAW, 8000, 1},
+  {1, "Reserved",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {2, "Reserved",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {3, "GSM",         CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
+  {4, "G723",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
+  {5, "DVI4",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
+  {6, "DVI4",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 16000, 1},
+  {7, "LPC",         CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
+  {8, "PCMA",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_ALAW, 8000, 1},
+  {9, "G722",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
+  {10, "L16",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_S16BE, 44100, 2},
+  {11, "L16",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_S16BE, 44100, 1},
+  {12, "QCELP",      CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
+  {13, "CN",         CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
+  {14, "MPA",        CODEC_TYPE_AUDIO,   CODEC_ID_MP2, 90000, -1},
+  {15, "G728",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
+  {16, "DVI4",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 11025, 1},
+  {17, "DVI4",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 22050, 1},
+  {18, "G729",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
+  {19, "reserved",   CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
+  {20, "unassigned", CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
+  {21, "unassigned", CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
+  {22, "unassigned", CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
+  {23, "unassigned", CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
+  {24, "unassigned", CODEC_TYPE_VIDEO,   CODEC_ID_NONE, -1, -1},
+  {25, "CelB",       CODEC_TYPE_VIDEO,   CODEC_ID_NONE, 90000, -1},
+  {26, "JPEG",       CODEC_TYPE_VIDEO,   CODEC_ID_MJPEG, 90000, -1},
+  {27, "unassigned", CODEC_TYPE_VIDEO,   CODEC_ID_NONE, -1, -1},
+  {28, "nv",         CODEC_TYPE_VIDEO,   CODEC_ID_NONE, 90000, -1},
+  {29, "unassigned", CODEC_TYPE_VIDEO,   CODEC_ID_NONE, -1, -1},
+  {30, "unassigned", CODEC_TYPE_VIDEO,   CODEC_ID_NONE, -1, -1},
+  {31, "H261",       CODEC_TYPE_VIDEO,   CODEC_ID_H261, 90000, -1},
+  {32, "MPV",        CODEC_TYPE_VIDEO,   CODEC_ID_MPEG1VIDEO, 90000, -1},
+  {33, "MP2T",       CODEC_TYPE_DATA,    CODEC_ID_MPEG2TS, 90000, -1},
+  {34, "H263",       CODEC_TYPE_VIDEO,   CODEC_ID_H263, 90000, -1},
+  {35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {96, "dynamic",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {97, "dynamic",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {98, "dynamic",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {99, "dynamic",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {100, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {101, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {102, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {103, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {104, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {105, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {106, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {107, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {108, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {109, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {110, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {111, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {112, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {113, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {114, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {115, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {116, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {117, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {118, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {119, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {120, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {121, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {122, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {123, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {124, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {125, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {126, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {127, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+  {-1, "",           CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}
+};
+
+AVRtpDynamicPayloadType_t AVRtpDynamicPayloadTypes[]=
+{
+    {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4},
+    {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_MPEG4AAC},
+    {"", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE}
+};
 
 struct RTPDemuxContext {
     AVFormatContext *ic;
@@ -83,7 +194,7 @@ struct RTPDemuxContext {
     uint32_t base_timestamp;
     uint32_t cur_timestamp;
     int max_payload_size;
-    MpegTSContext *ts; /* only used for RTP_PT_MPEG2TS payloads */
+    MpegTSContext *ts; /* only used for MP2T payloads */
     int read_buf_index;
     int read_buf_size;
     
@@ -99,94 +210,37 @@ struct RTPDemuxContext {
     /* buffer for output */
     uint8_t buf[RTP_MAX_PACKET_LENGTH];
     uint8_t *buf_ptr;
+    /* special infos for au headers parsing */
+    rtp_payload_data_t *rtp_payload_data;
 };
 
 int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
 {
-    switch(payload_type) {
-    case RTP_PT_ULAW:
-        codec->codec_type = CODEC_TYPE_AUDIO;
-        codec->codec_id = CODEC_ID_PCM_MULAW;
-        codec->channels = 1;
-        codec->sample_rate = 8000;
-        break;
-    case RTP_PT_ALAW:
-        codec->codec_type = CODEC_TYPE_AUDIO;
-        codec->codec_id = CODEC_ID_PCM_ALAW;
-        codec->channels = 1;
-        codec->sample_rate = 8000;
-        break;
-    case RTP_PT_S16BE_STEREO:
-        codec->codec_type = CODEC_TYPE_AUDIO;
-        codec->codec_id = CODEC_ID_PCM_S16BE;
-        codec->channels = 2;
-        codec->sample_rate = 44100;
-        break;
-    case RTP_PT_S16BE_MONO:
-        codec->codec_type = CODEC_TYPE_AUDIO;
-        codec->codec_id = CODEC_ID_PCM_S16BE;
-        codec->channels = 1;
-        codec->sample_rate = 44100;
-        break;
-    case RTP_PT_MPEGAUDIO:
-        codec->codec_type = CODEC_TYPE_AUDIO;
-        codec->codec_id = CODEC_ID_MP2;
-        break;
-    case RTP_PT_JPEG:
-        codec->codec_type = CODEC_TYPE_VIDEO;
-        codec->codec_id = CODEC_ID_MJPEG;
-        break;
-    case RTP_PT_MPEGVIDEO:
-        codec->codec_type = CODEC_TYPE_VIDEO;
-        codec->codec_id = CODEC_ID_MPEG1VIDEO;
-        break;
-    case RTP_PT_MPEG2TS:
-        codec->codec_type = CODEC_TYPE_DATA;
-        codec->codec_id = CODEC_ID_MPEG2TS;
-        break;
-    default:
-        return -1;
+    if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) {
+        codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type;
+        codec->codec_id = AVRtpPayloadTypes[payload_type].codec_type;
+        if (AVRtpPayloadTypes[payload_type].audio_channels > 0)
+            codec->channels = AVRtpPayloadTypes[payload_type].audio_channels;
+        if (AVRtpPayloadTypes[payload_type].clock_rate > 0)
+            codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate;
+        return 0;
     }
-    return 0;
+    return -1;
 }
 
 /* return < 0 if unknown payload type */
 int rtp_get_payload_type(AVCodecContext *codec)
 {
-    int payload_type;
+    int i, payload_type;
 
     /* compute the payload type */
-    payload_type = -1;
-    switch(codec->codec_id) {
-    case CODEC_ID_PCM_MULAW:
-        payload_type = RTP_PT_ULAW;
-        break;
-    case CODEC_ID_PCM_ALAW:
-        payload_type = RTP_PT_ALAW;
-        break;
-    case CODEC_ID_PCM_S16BE:
-        if (codec->channels == 1) {
-            payload_type = RTP_PT_S16BE_MONO;
-        } else if (codec->channels == 2) {
-            payload_type = RTP_PT_S16BE_STEREO;
+    for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
+        if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) {
+            if (codec->codec_id == CODEC_ID_PCM_S16BE)
+                if (codec->channels != AVRtpPayloadTypes[i].audio_channels)
+                    continue;
+            payload_type = AVRtpPayloadTypes[i].pt;
         }
-        break;
-    case CODEC_ID_MP2:
-    case CODEC_ID_MP3:
-        payload_type = RTP_PT_MPEGAUDIO;
-        break;
-    case CODEC_ID_MJPEG:
-        payload_type = RTP_PT_JPEG;
-        break;
-    case CODEC_ID_MPEG1VIDEO:
-        payload_type = RTP_PT_MPEGVIDEO;
-        break;
-    case CODEC_ID_MPEG2TS:
-        payload_type = RTP_PT_MPEG2TS;
-        break;
-    default:
-        break;
-    }
     return payload_type;
 }
 
@@ -216,7 +270,7 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l
  * MPEG2TS streams to indicate that they should be demuxed inside the
  * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) 
  */
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type)
+RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_t *rtp_payload_data)
 {
     RTPDemuxContext *s;
 
@@ -228,7 +282,8 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_t
     s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
     s->ic = s1;
     s->st = st;
-    if (payload_type == RTP_PT_MPEG2TS) {
+    s->rtp_payload_data = rtp_payload_data;
+    if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) {
         s->ts = mpegts_parse_open(s->ic);
         if (s->ts == NULL) {
             av_free(s);
@@ -250,6 +305,57 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_t
     return s;
 }
 
+static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
+{
+    AVCodecContext codec;
+    int au_headers_length, au_header_size, i;
+    GetBitContext getbitcontext;
+    rtp_payload_data_t *infos;
+
+    infos = s->rtp_payload_data;
+
+    if (infos == NULL)
+        return -1;
+
+    codec = s->st->codec;
+
+    /* decode the first 2 bytes where are stored the AUHeader sections
+       length in bits */
+    au_headers_length = BE_16(buf);
+
+    if (au_headers_length > RTP_MAX_PACKET_LENGTH)
+      return -1;
+
+    infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
+
+    /* skip AU headers length section (2 bytes) */
+    buf += 2;
+
+    init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
+
+    /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
+    au_header_size = infos->sizelength + infos->indexlength;
+    if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
+        return -1;
+
+    infos->nb_au_headers = au_headers_length / au_header_size;
+    infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
+
+    /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
+       In my test, the faad decoder doesnt behave correctly when sending each AU one by one
+       but does when sending the whole as one big packet...  */
+    infos->au_headers[0].size = 0;
+    infos->au_headers[0].index = 0;
+    for (i = 0; i < infos->nb_au_headers; ++i) {
+        infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
+        infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
+    }
+
+    infos->nb_au_headers = 1;
+
+    return 0;
+}
+
 /**
  * Parse an RTP or RTCP packet directly sent as a buffer. 
  * @param s RTP parse context.
@@ -304,8 +410,8 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
         av_log(&s->st->codec, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n", 
                payload_type, seq, ((s->seq + 1) & 0xffff));
     }
-    s->seq = seq;
 #endif
+    s->seq = seq;
     len -= 12;
     buf += 12;
 
@@ -370,6 +476,28 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
                 pkt->pts = addend + delta_timestamp;
             }
             break;
+        case CODEC_ID_MPEG4:
+            pkt->pts = timestamp;
+            break;
+        case CODEC_ID_MPEG4AAC:
+            if (rtp_parse_mp4_au(s, buf))
+              return -1;
+            rtp_payload_data_t *infos = s->rtp_payload_data;
+            if (infos == NULL)
+                return -1;
+            buf += infos->au_headers_length_bytes + 2;
+            len -= infos->au_headers_length_bytes + 2;
+
+            /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
+               one au_header */
+            av_new_packet(pkt, infos->au_headers[0].size);
+            memcpy(pkt->data, buf, infos->au_headers[0].size);
+            buf += infos->au_headers[0].size;
+            len -= infos->au_headers[0].size;
+            s->read_buf_size = len;
+            s->buf_ptr = (char *)buf;
+            pkt->stream_index = s->st->index;
+            return 0;
         default:
             /* no timestamp info yet */
             break;
@@ -381,7 +509,7 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
 
 void rtp_parse_close(RTPDemuxContext *s)
 {
-    if (s->payload_type == RTP_PT_MPEG2TS) {
+    if (!strcmp(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) {
         mpegts_parse_close(s->ts);
     }
     av_free(s);
diff --git a/libavformat/rtp.h b/libavformat/rtp.h
index 0d9869236e..b7f9e909ef 100644
--- a/libavformat/rtp.h
+++ b/libavformat/rtp.h
@@ -19,22 +19,6 @@
 #ifndef RTP_H
 #define RTP_H
 
-enum RTPPayloadType {
-    RTP_PT_ULAW = 0,
-    RTP_PT_GSM = 3,
-    RTP_PT_G723 = 4,
-    RTP_PT_ALAW = 8,
-    RTP_PT_S16BE_STEREO = 10,
-    RTP_PT_S16BE_MONO = 11,
-    RTP_PT_MPEGAUDIO = 14,
-    RTP_PT_JPEG = 26,
-    RTP_PT_H261 = 31,
-    RTP_PT_MPEGVIDEO = 32,
-    RTP_PT_MPEG2TS = 33,
-    RTP_PT_H263 = 34, /* old H263 encapsulation */
-    RTP_PT_PRIVATE = 96,
-};
-
 #define RTP_MIN_PACKET_LENGTH 12
 #define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
 
@@ -43,8 +27,8 @@ int rtp_get_codec_info(AVCodecContext *codec, int payload_type);
 int rtp_get_payload_type(AVCodecContext *codec);
 
 typedef struct RTPDemuxContext RTPDemuxContext;
-
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type);
+typedef struct rtp_payload_data_s rtp_payload_data_s;
+RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_s *rtp_payload_data);
 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, 
                      const uint8_t *buf, int len);
 void rtp_parse_close(RTPDemuxContext *s);
@@ -58,4 +42,82 @@ void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd);
 
 extern URLProtocol rtp_protocol;
 
+#define RTP_PT_PRIVATE 96
+#define RTP_VERSION 2
+#define RTP_MAX_SDES 256   /* maximum text length for SDES */
+
+/* RTCP paquets use 0.5 % of the bandwidth */
+#define RTCP_TX_RATIO_NUM 5
+#define RTCP_TX_RATIO_DEN 1000
+
+/* Structure listing usefull vars to parse RTP packet payload*/
+typedef struct rtp_payload_data_s
+{
+    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;
+} rtp_payload_data_t;
+
+typedef struct AVRtpPayloadType_s
+{
+    int pt;
+    const char enc_name[50]; /* XXX: why 50 ? */
+    enum CodecType codec_type;
+    enum CodecID codec_id;
+    int clock_rate;
+    int audio_channels;
+} AVRtpPayloadType_t;
+
+typedef struct AVRtpDynamicPayloadType_s /* payload type >= 96 */
+{
+    const char enc_name[50]; /* XXX: still why 50 ? ;-) */
+    enum CodecType codec_type;
+    enum CodecID codec_id;
+} AVRtpDynamicPayloadType_t;
+
+typedef enum {
+  RTCP_SR   = 200,
+  RTCP_RR   = 201,
+  RTCP_SDES = 202,
+  RTCP_BYE  = 203,
+  RTCP_APP  = 204
+} rtcp_type_t;
+
+typedef enum {
+  RTCP_SDES_END    =  0,
+  RTCP_SDES_CNAME  =  1,
+  RTCP_SDES_NAME   =  2,
+  RTCP_SDES_EMAIL  =  3,
+  RTCP_SDES_PHONE  =  4,
+  RTCP_SDES_LOC    =  5,
+  RTCP_SDES_TOOL   =  6,
+  RTCP_SDES_NOTE   =  7,
+  RTCP_SDES_PRIV   =  8,
+  RTCP_SDES_IMG    =  9,
+  RTCP_SDES_DOOR   = 10,
+  RTCP_SDES_SOURCE = 11
+} rtcp_sdes_type_t;
+
+extern AVRtpPayloadType_t AVRtpPayloadTypes[];
+extern AVRtpDynamicPayloadType_t AVRtpDynamicPayloadTypes[];
+
 #endif /* RTP_H */
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 3261cf3d23..67c23aa4bc 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -66,22 +66,15 @@ typedef struct RTSPStream {
     struct in_addr sdp_ip; /* IP address  (from SDP content - not used in RTSP) */
     int sdp_ttl;  /* IP TTL (from SDP content - not used in RTSP) */
     int sdp_payload_type; /* payload type - only used in SDP */
+    rtp_payload_data_t rtp_payload_data; /* rtp payload parsing infos from SDP */
 } RTSPStream;
 
 static int rtsp_read_play(AVFormatContext *s);
 
 /* XXX: currently, the only way to change the protocols consists in
    changing this variable */
-#if 0
-int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_TCP) | (1 << RTSP_PROTOCOL_RTP_UDP) | (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST);
-#else
-/* try it if a proxy is used */
-int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_TCP);
-#endif
 
-/* if non zero, then set a range for RTP ports */
-int rtsp_rtp_port_min = 0;
-int rtsp_rtp_port_max = 0;
+int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_UDP);
 
 FFRTSPCallback *ff_rtsp_callback = NULL;
 
@@ -113,6 +106,8 @@ static void get_word_sep(char *buf, int buf_size, const char *sep,
     char *q;
 
     p = *pp;
+    if (*p == '/')
+        p++;
     skip_spaces(&p);
     q = buf;
     while (!strchr(sep, *p) && *p != '\0') {
@@ -145,18 +140,67 @@ static void get_word(char *buf, int buf_size, const char **pp)
 
 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other
    params>] */
-static int sdp_parse_rtpmap(AVCodecContext *codec, const char *p)
+static int sdp_parse_rtpmap(AVCodecContext *codec, int payload_type, const char *p)
 {
     char buf[256];
+    int i;
+    AVCodec *c;
+    char *c_name;
 
-    /* codec name */
+    /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
+       see if we can handle this kind of payload */
     get_word_sep(buf, sizeof(buf), "/", &p);
-    if (!strcmp(buf, "MP4V-ES")) {
-        codec->codec_id = CODEC_ID_MPEG4;
-        return 0;
+    if (payload_type >= RTP_PT_PRIVATE) {
+        /* We are in dynmaic payload type case ... search into AVRtpDynamicPayloadTypes[] */
+        for (i = 0; AVRtpDynamicPayloadTypes[i].codec_id != CODEC_ID_NONE; ++i)
+            if (!strcmp(buf, AVRtpDynamicPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpDynamicPayloadTypes[i].codec_type)) {
+                codec->codec_id = AVRtpDynamicPayloadTypes[i].codec_id;
+                break;
+            }
     } else {
-        return -1;
+        /* We are in a standard case ( from http://www.iana.org/assignments/rtp-parameters) */
+        /* search into AVRtpPayloadTypes[] */
+        for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
+            if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpPayloadTypes[i].codec_type)){
+                codec->codec_id = AVRtpPayloadTypes[i].codec_id;
+                break;
+            }
     }
+
+    c = avcodec_find_decoder(codec->codec_id);
+    if (c && c->name)
+        c_name = (char *)c->name;
+    else
+        c_name = (char *)NULL;
+
+    if (c_name) {
+        get_word_sep(buf, sizeof(buf), "/", &p);
+        i = atoi(buf);
+        switch (codec->codec_type) {
+            case CODEC_TYPE_AUDIO:
+                av_log(codec, AV_LOG_DEBUG, " audio codec set to : %s\n", c_name);
+                codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
+                codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
+                if (i > 0) {
+                    codec->sample_rate = i;
+                    get_word_sep(buf, sizeof(buf), "/", &p);
+                    i = atoi(buf);
+                    if (i > 0)
+                        codec->channels = i;
+                }
+                av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate);
+                av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels);
+                break;
+            case CODEC_TYPE_VIDEO:
+                av_log(codec, AV_LOG_DEBUG, " video codec set to : %s\n", c_name);
+                break;
+            default:
+                break;
+        }
+        return 0;
+    }
+
+    return -1;
 }
 
 /* return the length and optionnaly the data */
@@ -188,11 +232,58 @@ static int hex_to_data(uint8_t *data, const char *p)
     return len;
 }
 
-static void sdp_parse_fmtp(AVCodecContext *codec, const char *p)
+static void sdp_parse_fmtp_config(AVCodecContext *codec, char *attr, char *value)
+{
+    switch (codec->codec_id) {
+        case CODEC_ID_MPEG4:
+        case CODEC_ID_MPEG4AAC:
+            if (!strcmp(attr, "config")) {
+                /* decode the hexa encoded parameter */
+                int len = hex_to_data(NULL, value);
+                codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
+                if (!codec->extradata)
+                    return;
+                codec->extradata_size = len;
+                hex_to_data(codec->extradata, value);
+            }
+            break;
+        default:
+            break;
+    }
+    return;
+}
+
+typedef struct attrname_map
+{
+    char *str;
+    uint16_t type;
+    uint32_t offset;
+} attrname_map_t;
+
+/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
+#define ATTR_NAME_TYPE_INT 0
+#define ATTR_NAME_TYPE_STR 1
+static attrname_map_t attr_names[]=
+{
+    {"SizeLength",       ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, sizelength)},
+    {"IndexLength",      ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexlength)},
+    {"IndexDeltaLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexdeltalength)},
+    {"profile-level-id", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, profile_level_id)},
+    {"StreamType",       ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, streamtype)},
+    {"mode",             ATTR_NAME_TYPE_STR, offsetof(rtp_payload_data_t, mode)},
+    {NULL, -1, -1},
+};
+
+/* parse a SDP line and save stream attributes */
+static void sdp_parse_fmtp(AVStream *st, const char *p)
 {
     char attr[256];
     char value[4096];
-    int len;
+    int i;
+
+    RTSPStream *rtsp_st = st->priv_data;
+    AVCodecContext *codec = &st->codec;
+    rtp_payload_data_t *rtp_payload_data = &rtsp_st->rtp_payload_data;
 
     /* loop on each attribute */
     for(;;) {
@@ -205,25 +296,17 @@ static void sdp_parse_fmtp(AVCodecContext *codec, const char *p)
         get_word_sep(value, sizeof(value), ";", &p);
         if (*p == ';')
             p++;
-        /* handle MPEG4 video */
-        switch(codec->codec_id) {
-        case CODEC_ID_MPEG4:
-            if (!strcmp(attr, "config")) {
-                /* decode the hexa encoded parameter */
-                len = hex_to_data(NULL, value);
-                codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
-                if (!codec->extradata)
-                    goto fail;
-                codec->extradata_size = len;
-                hex_to_data(codec->extradata, value);
-            }
-            break;
-        default:
-            /* ignore data for other codecs */
-            break;
+	/* grab the codec extra_data from the config parameter of the fmtp line */
+        sdp_parse_fmtp_config(codec, attr, value);
+        /* Looking for a known attribute */
+        for (i = 0; attr_names[i].str; ++i) {
+            if (!strcasecmp(attr, attr_names[i].str)) {
+                if (attr_names[i].type == ATTR_NAME_TYPE_INT)
+                    *(int *)((char *)rtp_payload_data + attr_names[i].offset) = atoi(value);
+                else if (attr_names[i].type == ATTR_NAME_TYPE_STR)
+                    *(char **)((char *)rtp_payload_data + attr_names[i].offset) = av_strdup(value);
+	    }
         }
-    fail: ;
-        //        printf("'%s' = '%s'\n", attr, value);
     }
 }
 
@@ -314,7 +397,7 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
         get_word(buf1, sizeof(buf1), &p); /* format list */
         rtsp_st->sdp_payload_type = atoi(buf1);
 
-        if (rtsp_st->sdp_payload_type == RTP_PT_MPEG2TS) {
+        if (!strcmp(AVRtpPayloadTypes[rtsp_st->sdp_payload_type].enc_name, "MP2T")) {
             /* no corresponding stream */
         } else {
             st = av_new_stream(s, 0);
@@ -323,7 +406,7 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
             st->priv_data = rtsp_st;
             rtsp_st->stream_index = st->index;
             st->codec.codec_type = codec_type;
-            if (rtsp_st->sdp_payload_type < 96) {
+            if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
                 /* if standard payload type, we can find the codec right now */
                 rtp_get_codec_info(&st->codec, rtsp_st->sdp_payload_type);
             }
@@ -355,7 +438,7 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
                 st = s->streams[i];
                 rtsp_st = st->priv_data;
                 if (rtsp_st->sdp_payload_type == payload_type) {
-                    sdp_parse_rtpmap(&st->codec, p);
+                    sdp_parse_rtpmap(&st->codec, payload_type, p);
                 }
             }
         } else if (strstart(p, "fmtp:", &p)) {
@@ -366,7 +449,7 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
                 st = s->streams[i];
                 rtsp_st = st->priv_data;
                 if (rtsp_st->sdp_payload_type == payload_type) {
-                    sdp_parse_fmtp(&st->codec, p);
+                    sdp_parse_fmtp(st, p);
                 }
             }
         }
@@ -715,7 +798,7 @@ static int rtsp_read_header(AVFormatContext *s,
     RTSPState *rt = s->priv_data;
     char host[1024], path[1024], tcpname[1024], cmd[2048];
     URLContext *rtsp_hd;
-    int port, i, ret, err;
+    int port, i, j, ret, err;
     RTSPHeader reply1, *reply = &reply1;
     unsigned char *content = NULL;
     RTSPStream *rtsp_st;
@@ -763,7 +846,8 @@ static int rtsp_read_header(AVFormatContext *s,
     /* for each stream, make the setup request */
     /* XXX: we assume the same server is used for the control of each
        RTSP stream */
-    for(i=0;i<rt->nb_rtsp_streams;i++) {
+
+    for(j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
         char transport[2048];
 
         rtsp_st = rt->rtsp_streams[i];
@@ -774,22 +858,24 @@ static int rtsp_read_header(AVFormatContext *s,
         /* RTP/UDP */
         if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP)) {
             char buf[256];
-            int j;
 
             /* first try in specified port range */
-            if (rtsp_rtp_port_min != 0) {
-                for(j=rtsp_rtp_port_min;j<=rtsp_rtp_port_max;j++) {
+            if (RTSP_RTP_PORT_MIN != 0) {
+                while(j <= RTSP_RTP_PORT_MAX) {
                     snprintf(buf, sizeof(buf), "rtp://?localport=%d", j);
-                    if (url_open(&rtsp_st->rtp_handle, buf, URL_RDONLY) == 0)
+                    if (url_open(&rtsp_st->rtp_handle, buf, URL_RDONLY) == 0) {
+                        j += 2; /* we will use two port by rtp stream (rtp and rtcp) */
                         goto rtp_opened;
+                    }
                 }
             }
 
-            /* then try on any port */
-            if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
-                err = AVERROR_INVALIDDATA;
-                goto fail;
-            }
+/*            then try on any port
+**            if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
+**                err = AVERROR_INVALIDDATA;
+**                goto fail;
+**            }
+*/
 
         rtp_opened:
             port = rtp_get_local_port(rtsp_st->rtp_handle);
@@ -801,14 +887,14 @@ static int rtsp_read_header(AVFormatContext *s,
         }
 
         /* RTP/TCP */
-        if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) {
+        else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) {
             if (transport[0] != '\0')
                 pstrcat(transport, sizeof(transport), ",");
             snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1,
                      "RTP/AVP/TCP");
         }
 
-        if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) {
+        else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) {
             if (transport[0] != '\0')
                 pstrcat(transport, sizeof(transport), ",");
             snprintf(transport + strlen(transport), 
@@ -887,7 +973,8 @@ static int rtsp_read_header(AVFormatContext *s,
             st = s->streams[rtsp_st->stream_index];
         if (!st)
             s->ctx_flags |= AVFMTCTX_NOHEADER;
-        rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type);
+        rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
+
         if (!rtsp_st->rtp_ctx) {
             err = AVERROR_NOMEM;
             goto fail;
@@ -1233,7 +1320,7 @@ static int sdp_read_header(AVFormatContext *s,
             st = s->streams[rtsp_st->stream_index];
         if (!st)
             s->ctx_flags |= AVFMTCTX_NOHEADER;
-        rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type);
+        rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
         if (!rtsp_st->rtp_ctx) {
             err = AVERROR_NOMEM;
             goto fail;
diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h
index 3a2713f655..6c2c5efd52 100644
--- a/libavformat/rtsp.h
+++ b/libavformat/rtsp.h
@@ -35,6 +35,10 @@ enum RTSPProtocol {
 #define RTSP_DEFAULT_PORT   554
 #define RTSP_MAX_TRANSPORTS 8
 #define RTSP_TCP_MAX_PACKET_SIZE 1472
+#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 2
+#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
+#define RTSP_RTP_PORT_MIN 5000
+#define RTSP_RTP_PORT_MAX 10000
 
 typedef struct RTSPTransportField {
     int interleaved_min, interleaved_max;  /* interleave ids, if TCP transport */