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A question about RTSP on gst-plugins-base-0.10.22

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Yang Felix
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A question about RTSP on gst-plugins-base-0.10.22

Yang Felix
Dear All:
     I have port the rtsp test program on the Fedora 11,. The following is the code and it is modified from the test.c in the previous version of gst-plugins-base. I use live555 media server to do the rtsp server.
When I built and run it, it will stop when the "PLAY" finishs. I use the sniffer and I am sure the server start to sending the data. After a while, the gst_rtsp_connection_receive return -11. Do someone help me? Thanks!
 
 
gstream version:0.10.24-1.fc11
gst-plugin-base version:0.10.23-3.fc11
 
//The console display
 
[felix@Felix-NB test_rtsp]$ ./run
parsing url "rtsp://192.168.11.167/sd.ts"...
  url host: 192.168.11.167
  url port: 0
  url path: /sd.ts
creating connection...
opening connection...
RTSP request message 0xbfe22800
 request line:
   method: 'DESCRIBE'
   uri:    'rtsp://192.168.11.167/sd.ts'
   version: '1.0'
 headers:
   key: 'Accept', value: 'application/sdp'
 body:
RTSP response message 0xbfe227e4
 status line:
   code:   '200'
   reason: 'OK'
   version: '1.0'
 headers:
   key: 'CSeq', value: '1'
   key: 'Date', value: 'Thu, Aug 13 2009 09:38:43 GMT'
   key: 'Content-Base', value: 'rtsp://192.168.11.167/sd.ts/'
   key: 'Content-Type', value: 'application/sdp'
   key: 'Content-Length', value: '376'
 body: length 377
00000000 (0x87608e0): 76 3d 30 0d 0a 6f 3d 2d 20 31 32 35 30 31 34 37  v=0..o=- 1250147
00000010 (0x87608f0): 39 39 39 36 39 38 31 34 32 20 31 20 49 4e 20 49  999698142 1 IN I
00000020 (0x8760900): 50 34 20 31 30 2e 31 34 34 2e 32 34 2e 38 0d 0a  P4 10.144.24.8..
00000030 (0x8760910): 73 3d 4d 50 45 47 20 54 72 61 6e 73 70 6f 72 74  s=MPEG Transport
00000040 (0x8760920): 20 53 74 72 65 61 6d 2c 20 73 74 72 65 61 6d 65   Stream, streame
00000050 (0x8760930): 64 20 62 79 20 74 68 65 20 4c 49 56 45 35 35 35  d by the LIVE555
00000060 (0x8760940): 20 4d 65 64 69 61 20 53 65 72 76 65 72 0d 0a 69   Media Server..i
00000070 (0x8760950): 3d 73 64 2e 74 73 0d 0a 74 3d 30 20 30 0d 0a 61  =sd.ts..t=0 0..a
00000080 (0x8760960): 3d 74 6f 6f 6c 3a 4c 49 56 45 35 35 35 20 53 74  =tool:LIVE555 St
00000090 (0x8760970): 72 65 61 6d 69 6e 67 20 4d 65 64 69 61 20 76 32  reaming Media v2
000000a0 (0x8760980): 30 30 39 2e 30 37 2e 30 39 0d 0a 61 3d 74 79 70  009.07.09..a=typ
000000b0 (0x8760990): 65 3a 62 72 6f 61 64 63 61 73 74 0d 0a 61 3d 63  e:broadcast..a=c
000000c0 (0x87609a0): 6f 6e 74 72 6f 6c 3a 2a 0d 0a 61 3d 72 61 6e 67  ontrol:*..a=rang
000000d0 (0x87609b0): 65 3a 6e 70 74 3d 30 2d 0d 0a 61 3d 78 2d 71 74  e:npt=0-..a=x-qt
000000e0 (0x87609c0): 2d 74 65 78 74 2d 6e 61 6d 3a 4d 50 45 47 20 54  -text-nam:MPEG T
000000f0 (0x87609d0): 72 61 6e 73 70 6f 72 74 20 53 74 72 65 61 6d 2c  ransport Stream,
00000100 (0x87609e0): 20 73 74 72 65 61 6d 65 64 20 62 79 20 74 68 65   streamed by the
00000110 (0x87609f0): 20 4c 49 56 45 35 35 35 20 4d 65 64 69 61 20 53   LIVE555 Media S
00000120 (0x8760a00): 65 72 76 65 72 0d 0a 61 3d 78 2d 71 74 2d 74 65  erver..a=x-qt-te
00000130 (0x8760a10): 78 74 2d 69 6e 66 3a 73 64 2e 74 73 0d 0a 6d 3d  xt-inf:sd.ts..m=
00000140 (0x8760a20): 76 69 64 65 6f 20 30 20 52 54 50 2f 41 56 50 20  video 0 RTP/AVP
00000150 (0x8760a30): 33 33 0d 0a 63 3d 49 4e 20 49 50 34 20 30 2e 30  33..c=IN IP4 0.0
00000160 (0x8760a40): 2e 30 2e 30 0d 0a 61 3d 63 6f 6e 74 72 6f 6c 3a  .0.0..a=control:
00000170 (0x8760a50): 74 72 61 63 6b 31 0d 0a 00                       track1...
sdp packet 0xbfe22784:
 version:       '0'
 origin:
  username:     '-'
  sess_id:      '1250147999698142'
  sess_version: '1'
  nettype:      'IN'
  addrtype:     'IP4'
  addr:         '10.144.24.8'
 session_name:  'MPEG Transport Stream, streamed by the LIVE555 Media Server'
 information:   'sd.ts'
 uri:           '(NULL)'
 connection:
  nettype:      '(NULL)'
  addrtype:     '(NULL)'
  address:      '(NULL)'
  ttl:          '0'
  addr_number:  '0'
 key:
  type:         '(NULL)'
  data:         '(NULL)'
 attributes:
  attribute 'tool' : 'LIVE555 Streaming Media v2009.07.09'
  attribute 'type' : 'broadcast'
  attribute 'control' : '*'
  attribute 'range' : 'npt=0-'
  attribute 'x-qt-text-nam' : 'MPEG Transport Stream, streamed by the LIVE555 Media Server'
  attribute 'x-qt-text-inf' : 'sd.ts'
 medias:
  media 0:
   media:       'video'
   port:        '0'
   num_ports:   '4294967295'
   proto:       'RTP/AVP'
   formats:
    format  '33'
   information: '(NULL)'
   connections:
    nettype:      'IN'
    addrtype:     'IP4'
    address:      '0.0.0.0'
    ttl:          '0'
    addr_number:  '0'
   key:
    type:       '(NULL)'
    data:       '(NULL)'
   attributes:
    attribute 'control' : 'track1'
setup media 0
setup rtsp://192.168.11.167/sd.ts/track1
RTSP request message 0xbfe22800
 request line:
   method: 'SETUP'
   uri:    'rtsp://192.168.11.167/sd.ts/track1'
   version: '1.0'
 headers:
   key: 'Transport', value: 'RTP/AVP/UDP;unicast;client_port=30000-30001,RTP/AVP/UDP;multicast,RTP/AVP/TCP'
 body:
RTSP response message 0xbfe227e4
 status line:
   code:   '200'
   reason: 'OK'
   version: '1.0'
 headers:
   key: 'CSeq', value: '2'
   key: 'Date', value: 'Thu, Aug 13 2009 09:38:43 GMT'
   key: 'Transport', value: 'RTP/AVP;unicast;destination=192.168.11.168;source=192.168.11.167;client_port=30000-30001;server_port=6972-6973'
   key: 'Session', value: '52'
 body: length 0
RTSP request message 0xbfe22800
 request line:
   method: 'PLAY'
   uri:    'rtsp://192.168.11.167/sd.ts'
   version: '1.0'
 headers:
   key: 'Accept', value: 'npt=0-'
 body:
RTSP response message 0xbfe227e4
 status line:
   code:   '200'
   reason: 'OK'
   version: '1.0'
 headers:
   key: 'CSeq', value: '3'
   key: 'Date', value: 'Thu, Aug 13 2009 09:38:43 GMT'
   key: 'Session', value: '52'
   key: 'RTP-Info', value: 'url=rtsp://192.168.11.167/sd.ts/track1;seq=23957;rtptime=2147510730'
 body: length 0
error receiving response -11
[felix@Felix-NB test_rtsp]$
 
 
 
 
 
///The code
#include <gst/gst.h>
#include <gst/rtsp/gstrtspconnection.h>
#include <gst/rtsp/gstrtspmessage.h>
#include <gst/rtsp/gstrtspurl.h>
#include <gst/sdp/gstsdpmessage.h>
#include <gst/rtp/gstrtppayloads.h>
#include <gst/rtsp/gstrtsprange.h>
#include <gst/check/gstcheck.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include <string.h>
int main(int argc,char *argv[])
{
 GstRTSPUrl     *pURL;
 gchar       *pURLStr;
 GMainLoop        *g_loop;
 GstRTSPConnection *pConn;
 GstRTSPResult   nRes; 
 GstRTSPMessage   Request_t  = { 0 };
  GstRTSPMessage   Response_t  = { 0 };
  GstSDPMessage   SDP_t    = { 0 };
 
 
 gst_init(&argc,&argv);
 g_loop = g_main_loop_new (NULL, FALSE);
 
 pURLStr="rtsp://192.168.11.167/sd.ts";
 //pURLStr="rtsp://thread:5454/south-rtsp.mp3"
 
 /*create url */ 
  g_print ("parsing url \"%s\"...\n", pURLStr);
   
  nRes = gst_rtsp_url_parse (pURLStr, &pURL);
  if (nRes <0) {
    g_print ("error parsing url \"%s\"\n", pURLStr);
    return (-1);
  }
  g_print ("  url host: %s\n", pURL->host);
  g_print ("  url port: %d\n", pURL->port);
  g_print ("  url path: %s\n", pURL->abspath);
 
 
  /* create and open connection */
  g_print ("creating connection...\n");
  nRes = gst_rtsp_connection_create (pURL, &pConn);
  if (nRes != GST_RTSP_OK) {
   g_print ("error creating connection to \"%s\"\n", pURLStr);
    return (-1);
  }
 
  /* open connection */
  g_print ("opening connection...\n");
  nRes = gst_rtsp_connection_connect (pConn, NULL);
  if (nRes != GST_RTSP_OK) {
   g_print ("error opening connection to \"%s\"\n", pURLStr);
    return (-1);
  }
   /* do describe */
  {
    nRes = gst_rtsp_message_init_request (&Request_t, GST_RTSP_DESCRIBE, pURLStr);
    if (nRes != GST_RTSP_OK) {
      g_print ("error creating request\n");
      return (-1);
    }
    gst_rtsp_message_add_header (&Request_t, GST_RTSP_HDR_ACCEPT, "application/sdp");
    gst_rtsp_message_dump (&Request_t);
    nRes = gst_rtsp_connection_send (pConn, &Request_t, NULL);
    if (nRes != GST_RTSP_OK) {
      g_print ("error sending request\n");
      return (-1);
    }
    nRes = gst_rtsp_connection_receive (pConn, &Response_t, NULL);
    if (nRes != GST_RTSP_OK) {
      g_print ("error receiving response\n");
      return (-1);
    }
    gst_rtsp_message_dump (&Response_t);
  }
  /* parse SDP */
  {
    guint8 *data;
    guint size;
    gst_rtsp_message_get_body (&Response_t, &data, &size);
    gst_sdp_message_init (&SDP_t);
    gst_sdp_message_parse_buffer (data, size, &SDP_t);
    gst_sdp_message_dump (&SDP_t);
  }
 
 
    /* do setup */
  {
    gint i;
    for (i = 0; i < gst_sdp_message_medias_len (&SDP_t); i++) {
      GstSDPMedia *media;
      gchar *pSetupURL;
      gchar *pControlURL;
      media = gst_sdp_message_get_media (&SDP_t, i);
      g_print ("setup media %d\n", i);
      pControlURL = gst_sdp_media_get_attribute_val (media, "control");
      pSetupURL = g_strdup_printf ("%s/%s", pURLStr, pControlURL);
      g_print ("setup %s\n", pSetupURL);
      nRes = gst_rtsp_message_init_request (&Request_t, GST_RTSP_SETUP, pSetupURL);
      if (nRes != GST_RTSP_OK) {
        g_print ("error creating request\n");
        return (-1);
      }
      gst_rtsp_message_add_header (&Request_t, GST_RTSP_HDR_TRANSPORT,
          "RTP/AVP/UDP;unicast;client_port=30000-30001,RTP/AVP/UDP;multicast,RTP/AVP/TCP");
          //"RTP/AVP;unicast;client_port=5000-5001");
         // "RTP/AVP/TCP");
      gst_rtsp_message_dump (&Request_t);
      nRes = gst_rtsp_connection_send (pConn, &Request_t, NULL);
      if (nRes != GST_RTSP_OK) {
        g_print ("error sending request\n");
        return (-1);
     }
     nRes = gst_rtsp_connection_receive (pConn, &Response_t, NULL);
     if (nRes != GST_RTSP_OK) {
      g_print ("error receiving response\n");
       return (-1);
     }
     gst_rtsp_message_dump (&Response_t);
    }       
  }
    /* do play */
  {
    nRes = gst_rtsp_message_init_request (&Request_t, GST_RTSP_PLAY, pURLStr);
    if (nRes != GST_RTSP_OK) {
      g_print ("error creating request\n");
      return (-1);
    }
   
    gst_rtsp_message_add_header (&Request_t, GST_RTSP_HDR_ACCEPT, "npt=0-");
   
    gst_rtsp_message_dump (&Request_t);
    nRes = gst_rtsp_connection_send (pConn, &Request_t, NULL);
    if (nRes != GST_RTSP_OK) {
      g_print ("error sending request\n");
      return (-1);
    }
    nRes = gst_rtsp_connection_receive (pConn, &Response_t, NULL);
    if (nRes != GST_RTSP_OK) {
      g_print ("error receiving response\n");
      return (-1);
    }
    gst_rtsp_message_dump (&Response_t);
  }

  while (TRUE) {
    nRes = gst_rtsp_connection_receive (pConn, &Response_t, NULL);
  g_print("%d\n",nRes);
    if (nRes != GST_RTSP_OK) {
      g_print ("error receiving response %d\n",nRes);
      return (-1);
    }
    gst_rtsp_message_dump (&Response_t);
  }
  /* close connection */
  g_print ("closing connection...\n");
  nRes = gst_rtsp_connection_close (pConn);
  if (nRes != GST_RTSP_OK) {
    g_print ("error closing connection to \"%s\"\n", pURLStr);
    return (-1);
  }
 return 0;
}

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