rtsp: Allow requesting of filtering of source packets

If filtered, only packets from the right source address and port
are received.

To test, play back e.g. some mpeg4 video RTSP stream (where the
video stream is the first stream in the presentation) over UDP.
While receiving this stream, send another stream to the same port:
ffmpeg -re -i <whatever> -vcodec mpeg4 -an -f rtp
rtp://127.0.0.1:5000?localport=1234
Normally, the RTSP playback reports lots of errors at this point.

If the RTSP stream has the ?filter_src option enabled, these
interferring packets are ignored.

Originally committed as revision 26246 to svn://svn.ffmpeg.org/ffmpeg/trunk

diff --git a/doc/protocols.texi b/doc/protocols.texi
index 600f470..29f5c90 100644 (file)
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -251,6 +251,9 @@ Use UDP multicast as lower transport protocol.
 @item http
 Use HTTP tunneling as lower transport protocol, which is useful for
 passing proxies.
+
+@item filter_src
+Accept packets only from negotiated peer address and port.
 @end table
 
 Multiple lower transport protocols may be specified, in that case they are

diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index a38cc42..b09ebd7 100644 (file)
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -1186,16 +1186,18 @@ static int make_setup_request(AVFormatContext *s, const char *host, int port,
             break;
 
         case RTSP_LOWER_TRANSPORT_UDP: {
-            char url[1024];
+            char url[1024], options[30] = "";
 
+            if (rt->filter_source)
+                av_strlcpy(options, "?connect=1", sizeof(options));
             /* Use source address if specified */
             if (reply->transports[0].source[0]) {
                 ff_url_join(url, sizeof(url), "rtp", NULL,
                             reply->transports[0].source,
-                            reply->transports[0].server_port_min, NULL);
+                            reply->transports[0].server_port_min, options);
             } else {
                 ff_url_join(url, sizeof(url), "rtp", NULL, host,
-                            reply->transports[0].server_port_min, NULL);
+                            reply->transports[0].server_port_min, options);
             }
             if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
                 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
@@ -1315,6 +1317,8 @@ redirect:
             } else if(!strcmp(option, "http")) {
                 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
                 rt->control_transport = RTSP_MODE_TUNNEL;
+            } else if (!strcmp(option, "filter_src")) {
+                rt->filter_source = 1;
             } else {
                 /* Write options back into the buffer, using memmove instead
                  * of strcpy since the strings may overlap. */

diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h
index 628ce37..eee07ac 100644 (file)
--- a/libavformat/rtsp.h
+++ b/libavformat/rtsp.h
@@ -309,6 +309,10 @@ typedef struct RTSPState {
 
     /** Reusable buffer for receiving packets */
     uint8_t* recvbuf;
+
+    /** Filter incoming UDP packets - receive packets only from the right
+     * source address and port. */
+    int filter_source;
 } RTSPState;
 
 /**