<html> <head> <style> .hmmessage P { margin:0px; padding:0px } body.hmmessage { font-size: 9pt; font-family:新細明體 } </style> </head> <body class='hmmessage'> 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; } <hr />聰明搜尋和瀏覽網路的免費工具列 — <a href='http://download.live.com/toolbar' target='_new'>Windows Live 工具列</a></body> </html>