[FFmpeg-cvslog] rtsp: Support tls-encapsulated RTSP

[Luca Barbato]

ffmpeg | branch: master | Luca Barbato <lu_zero at gentoo.org> | Thu Oct  9 19:35:34 2014 +0200| [8b2e9636c57b22582143467a8a06b509b47b92f9] | committer: Luca Barbato

rtsp: Support tls-encapsulated RTSP

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8b2e9636c57b22582143467a8a06b509b47b92f9
---

 libavformat/rtsp.c |   24 ++++++++++++++++++------
 libavformat/rtsp.h |    1 +
 2 files changed, 19 insertions(+), 6 deletions(-)

diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 646712e..d574797 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -1589,10 +1589,13 @@ void ff_rtsp_close_connections(AVFormatContext *s)
 int ff_rtsp_connect(AVFormatContext *s)
 {
     RTSPState *rt = s->priv_data;
-    char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
+    char proto[128], host[1024], path[1024];
+    char tcpname[1024], cmd[2048], auth[128];
+    const char *lower_rtsp_proto = "tcp";
     int port, err, tcp_fd;
     RTSPMessageHeader reply1 = {0}, *reply = &reply1;
     int lower_transport_mask = 0;
+    int default_port = RTSP_DEFAULT_PORT;
     char real_challenge[64] = "";
     struct sockaddr_storage peer;
     socklen_t peer_len = sizeof(peer);
@@ -1619,15 +1622,23 @@ int ff_rtsp_connect(AVFormatContext *s)
     rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
 
 redirect:
-    lower_transport_mask = rt->lower_transport_mask;
     /* extract hostname and port */
-    av_url_split(NULL, 0, auth, sizeof(auth),
+    av_url_split(proto, sizeof(proto), auth, sizeof(auth),
                  host, sizeof(host), &port, path, sizeof(path), s->filename);
+
+    if (!strcmp(proto, "rtsps")) {
+        lower_rtsp_proto         = "tls";
+        default_port             = RTSPS_DEFAULT_PORT;
+        rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
+    }
+
     if (*auth) {
         av_strlcpy(rt->auth, auth, sizeof(rt->auth));
     }
     if (port < 0)
-        port = RTSP_DEFAULT_PORT;
+        port = default_port;
+
+    lower_transport_mask = rt->lower_transport_mask;
 
     if (!lower_transport_mask)
         lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
@@ -1647,7 +1658,7 @@ redirect:
     /* Construct the URI used in request; this is similar to s->filename,
      * but with authentication credentials removed and RTSP specific options
      * stripped out. */
-    ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
+    ff_url_join(rt->control_uri, sizeof(rt->control_uri), proto, NULL,
                 host, port, "%s", path);
 
     if (rt->control_transport == RTSP_MODE_TUNNEL) {
@@ -1726,7 +1737,8 @@ redirect:
         }
     } else {
         /* open the tcp connection */
-        ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
+        ff_url_join(tcpname, sizeof(tcpname), lower_rtsp_proto, NULL,
+                    host, port, NULL);
         if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,
                        &s->interrupt_callback, NULL) < 0) {
             err = AVERROR(EIO);
diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h
index 7a910b0..e83ad54 100644
--- a/libavformat/rtsp.h
+++ b/libavformat/rtsp.h
@@ -70,6 +70,7 @@ enum RTSPControlTransport {
 };
 
 #define RTSP_DEFAULT_PORT   554
+#define RTSPS_DEFAULT_PORT  322
 #define RTSP_MAX_TRANSPORTS 8
 #define RTSP_TCP_MAX_PACKET_SIZE 1472
 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1