for ossrs#133, show more info about rtp.

trunk/src/app/srs_app_rtsp.cpp

@@ -128,15 +128,19 @@ int SrsRtspConn::do_cycle()
             }
         } else if (req->is_announce()) {
             srs_assert(req->sdp);
-            video_id = req->sdp->video_stream_id;
-            audio_id = req->sdp->audio_stream_id;
+            video_id = ::atoi(req->sdp->video_stream_id.c_str());
+            audio_id = ::atoi(req->sdp->audio_stream_id.c_str());
+            video_codec = req->sdp->video_codec;
+            audio_codec = req->sdp->audio_codec;
+            audio_sample_rate = ::atoi(req->sdp->audio_sample_rate.c_str());
+            audio_channel = ::atoi(req->sdp->audio_channel.c_str());
             sps = req->sdp->video_sps;
             pps = req->sdp->video_pps;
             asc = req->sdp->audio_sh;
-            srs_trace("rtsp: video(#%s, %s), audio(#%s, %s, %sHZ %schannels)", 
-                req->sdp->video_stream_id.c_str(), req->sdp->video_codec.c_str(),
-                req->sdp->audio_stream_id.c_str(), req->sdp->audio_codec.c_str(), 
-                req->sdp->audio_sample_rate.c_str(), req->sdp->audio_channel.c_str()
+            srs_trace("rtsp: video(#%d, %s, %s/%s), audio(#%d, %s, %s/%s, %dHZ %dchannels)", 
+                video_id, video_codec.c_str(), req->sdp->video_protocol.c_str(), req->sdp->video_transport_format.c_str(), 
+                audio_id, audio_codec.c_str(), req->sdp->audio_protocol.c_str(), req->sdp->audio_transport_format.c_str(),
+                audio_sample_rate, audio_channel
             );
 
             SrsRtspResponse* res = new SrsRtspResponse(req->seq);
@@ -167,7 +171,12 @@ int SrsRtspConn::do_cycle()
                 srs_error("rtsp: rtp listen at port=%d failed. ret=%d", lpm, ret);
                 return ret;
             }
-            srs_trace("rtsp: rtp listen at port=%d ok.", lpm);
+            srs_trace("rtsp: #%d %s over %s/%s/%s %s client-port=%d-%d, server-port=%d-%d", 
+                req->stream_id, (req->stream_id == video_id)? "Video":"Audio", 
+                req->transport->transport.c_str(), req->transport->profile.c_str(), req->transport->lower_transport.c_str(), 
+                req->transport->cast_type.c_str(), req->transport->client_port_min, req->transport->client_port_max, 
+                lpm, lpm + 1
+            );
 
             // create session.
             if (session.empty()) {

trunk/src/app/srs_app_rtsp.hpp

@@ -76,10 +76,14 @@ class SrsRtspConn : public ISrsThreadHandler
 private:
     std::string session;
     // video stream.
-    std::string video_id;
+    int video_id;
+    std::string video_codec;
     SrsRtpConn* video_rtp;
     // audio stream.
-    std::string audio_id;
+    int audio_id;
+    std::string audio_codec;
+    int audio_sample_rate;
+    int audio_channel;
     SrsRtpConn* audio_rtp;
     // video sequence header.
     std::string sps;

trunk/src/protocol/srs_rtsp_stack.cpp

@@ -461,6 +461,7 @@ SrsRtspRequest::SrsRtspRequest()
 {
     seq = 0;
     content_length = 0;
+    stream_id = 0;
     sdp = NULL;
     transport = NULL;
 }
@@ -766,13 +767,15 @@ int SrsRtspStack::do_recv_message(SrsRtspRequest* req)
     // for setup, parse the stream id from uri.
     if (req->is_setup()) {
         size_t pos = string::npos;
+        std::string stream_id;
         if ((pos = req->uri.rfind("/")) != string::npos) {
-            req->stream_id = req->uri.substr(pos + 1);
+            stream_id = req->uri.substr(pos + 1);
         }
-        if ((pos = req->stream_id.find("=")) != string::npos) {
-            req->stream_id = req->stream_id.substr(pos + 1);
+        if ((pos = stream_id.find("=")) != string::npos) {
+            stream_id = stream_id.substr(pos + 1);
         }
-        srs_info("rtsp: setup stream id=%s", req->stream_id.c_str());
+        req->stream_id = ::atoi(stream_id.c_str());
+        srs_info("rtsp: setup stream id=%d", req->stream_id);
     }
 
     // parse rdp body.

trunk/src/protocol/srs_rtsp_stack.hpp

@@ -48,7 +48,7 @@ class ISrsProtocolReaderWriter;
 // SP             = <US-ASCII SP, space (32)>
 #define __SRS_RTSP_SP ' ' // 0x20
 
-// 4 RTSP Message
+// 4 RTSP Message, @see rtsp-rfc2326-1998.pdf, page 37
 // Lines are terminated by CRLF, but
 // receivers should be prepared to also interpret CR and LF by
 // themselves as line terminators.
@@ -100,7 +100,7 @@ enum SrsRtspSdpState
 };
 
 /**
-* 10 Method Definitions
+* 10 Method Definitions, @see rtsp-rfc2326-1998.pdf, page 57
 * The method token indicates the method to be performed on the resource
 * identified by the Request-URI. The method is case-sensitive. New
 * methods may be defined in the future. Method names may not start with
@@ -147,7 +147,7 @@ enum SrsRtspTokenState
 };
 
 /**
-* the sdp in announce.
+* the sdp in announce, @see rtsp-rfc2326-1998.pdf, page 159
 * Appendix C: Use of SDP for RTSP Session Descriptions
 * The Session Description Protocol (SDP, RFC 2327 [6]) may be used to
 * describe streams or presentations in RTSP.
@@ -241,7 +241,7 @@ class SrsRtspSdp
 
 /**
 * the rtsp transport.
-* 12.39 Transport
+* 12.39 Transport, @see rtsp-rfc2326-1998.pdf, page 115
 * This request header indicates which transport protocol is to be used
 * and configures its parameters such as destination address,
 * compression, multicast time-to-live and destination port for a single
@@ -288,7 +288,7 @@ class SrsRtspTransport
 
 /**
 * the rtsp request message.
-* 6 Request
+* 6 Request, @see rtsp-rfc2326-1998.pdf, page 39
 * A request message from a client to a server or vice versa includes,
 * within the first line of that message, the method to be applied to
 * the resource, the identifier of the resource, and the protocol
@@ -322,14 +322,14 @@ class SrsRtspRequest
     */
     long seq;
     /**
-    * 12.16 Content-Type
+    * 12.16 Content-Type, @see rtsp-rfc2326-1998.pdf, page 99
     * See [H14.18]. Note that the content types suitable for RTSP are
     * likely to be restricted in practice to presentation descriptions and
     * parameter-value types.
     */
     std::string content_type;
     /**
-    * 12.14 Content-Length
+    * 12.14 Content-Length, @see rtsp-rfc2326-1998.pdf, page 99
     * This field contains the length of the content of the method (i.e.
     * after the double CRLF following the last header). Unlike HTTP, it
     * MUST be included in all messages that carry content beyond the header
@@ -353,7 +353,7 @@ class SrsRtspRequest
     /**
     * for setup message, parse the stream id from uri.
     */
-    std::string stream_id;
+    int stream_id;
 public:
     SrsRtspRequest();
     virtual ~SrsRtspRequest();
@@ -366,7 +366,7 @@ class SrsRtspRequest
 
 /**
 * the rtsp response message.
-* 7 Response
+* 7 Response, @see rtsp-rfc2326-1998.pdf, page 43
 * [H6] applies except that HTTP-Version is replaced by RTSP-Version.
 * Also, RTSP defines additional status codes and does not define some
 * HTTP codes. The valid response codes and the methods they can be used
@@ -396,7 +396,7 @@ class SrsRtspResponse
     // @see about the status of rtsp, see SRS_CONSTS_RTSP_OK
     int status;
     /**
-    * 12.17 CSeq
+    * 12.17 CSeq, @see rtsp-rfc2326-1998.pdf, page 99
     * The CSeq field specifies the sequence number for an RTSP requestresponse
     * pair. This field MUST be present in all requests and
     * responses. For every RTSP request containing the given sequence
@@ -426,7 +426,7 @@ class SrsRtspResponse
 };
 
 /**
-* 10.1 OPTIONS
+* 10.1 OPTIONS, @see rtsp-rfc2326-1998.pdf, page 59
 * The behavior is equivalent to that described in [H9.2]. An OPTIONS
 * request may be issued at any time, e.g., if the client is about to
 * try a nonstandard request. It does not influence server state.
@@ -446,7 +446,7 @@ class SrsRtspOptionsResponse : public SrsRtspResponse
 };
 
 /**
-* 10.4 SETUP
+* 10.4 SETUP, @see rtsp-rfc2326-1998.pdf, page 65
 * The SETUP request for a URI specifies the transport mechanism to be
 * used for the streamed media. A client can issue a SETUP request for a
 * stream that is already playing to change transport parameters, which