[FFmpeg-devel] [PATCH] rtsp.c: keep-alive

[Ronald S. Bultje]

Hi Luca,

On Fri, Mar 6, 2009 at 4:21 PM, Luca Barbato <lu_zero at gentoo.org> wrote:
> Ronald S. Bultje wrote:
>> On Fri, Mar 6, 2009 at 1:32 PM, Luca Barbato <lu_zero at gentoo.org> wrote:
>>> Shouldn't be enough sending rtcp RR as usual?
>>
>> RM doesn't use RTCP.
>
> Once you have it tested on wms and darwin (and feng if you like) you may
> commit it then.

MS doesn't like the OPTIONS and ignores it. MSDN [1] tells me to use
an empty GET_PARAMETER, and that indeed appears to work in my tests
(tried ~30min, worked fine). Attached patch works for WMS and Real
servers.

Feng / Darwin yet untested, any fancy test-sites you recommend for DSS?

Ronald

[1] http://msdn.microsoft.com/en-us/library/cc245366(PROT.10).aspx
-------------- next part --------------
Index: ffmpeg-svn/libavformat/rtsp.c
===================================================================
--- ffmpeg-svn.orig/libavformat/rtsp.c	2009-03-06 18:41:02.000000000 -0500
+++ ffmpeg-svn/libavformat/rtsp.c	2009-03-06 20:21:05.000000000 -0500
@@ -686,7 +686,12 @@
     /* NOTE: we do case independent match for broken servers */
     p = buf;
     if (av_stristart(p, "Session:", &p)) {
+        int t;
         get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
+        if (av_stristart(p, ";timeout=", &p) &&
+            (t = strtol(p, NULL, 10)) > 0) {
+            reply->timeout = t;
+        }
     } else if (av_stristart(p, "Content-Length:", &p)) {
         reply->content_length = strtol(p, NULL, 10);
     } else if (av_stristart(p, "Transport:", &p)) {
@@ -855,7 +860,7 @@
     return 0;
 }
 
-static void rtsp_send_cmd(AVFormatContext *s,
+static void rtsp_send_cmd_async (AVFormatContext *s,
                           const char *cmd, RTSPMessageHeader *reply,
                           unsigned char **content_ptr)
 {
@@ -875,6 +880,14 @@
     printf("Sending:\n%s--\n", buf);
 #endif
     url_write(rt->rtsp_hd, buf, strlen(buf));
+    rt->last_cmd_time = av_gettime();
+}
+
+static void rtsp_send_cmd (AVFormatContext *s,
+                           const char *cmd, RTSPMessageHeader *reply,
+                           unsigned char **content_ptr)
+{
+    rtsp_send_cmd_async(s, cmd, reply, content_ptr);
 
     rtsp_read_reply(s, reply, content_ptr, 0);
 }
@@ -957,6 +970,9 @@
     else
         trans_pref = "RTP/AVP";
 
+    /* default timeout: 1 minute */
+    rt->timeout = 60;
+
     /* for each stream, make the setup request */
     /* XXX: we assume the same server is used for the control of each
        RTSP stream */
@@ -1137,6 +1153,9 @@
             goto fail;
     }
 
+    if (reply->timeout > 0)
+        rt->timeout = reply->timeout;
+
     if (rt->server_type == RTSP_SERVER_REAL)
         rt->need_subscription = 1;
 
@@ -1419,12 +1438,12 @@
     RTSPStream *rtsp_st;
     int ret, len;
     uint8_t buf[10 * RTP_MAX_PACKET_LENGTH];
+    RTSPMessageHeader reply1, *reply = &reply1;
+    char cmd[1024];
 
     if (rt->server_type == RTSP_SERVER_REAL) {
         int i;
-        RTSPMessageHeader reply1, *reply = &reply1;
         enum AVDiscard cache[MAX_STREAMS];
-        char cmd[1024];
 
         for (i = 0; i < s->nb_streams; i++)
             cache[i] = s->streams[i]->discard;
@@ -1526,6 +1545,22 @@
         /* more packets may follow, so we save the RTP context */
         rt->cur_transport_priv = rtsp_st->transport_priv;
     }
+
+    /* send dummy request to keep TCP connection alive */
+    if ((rt->server_type == RTSP_SERVER_WMS ||
+         rt->server_type == RTSP_SERVER_REAL) &&
+        (av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
+        if (rt->server_type == RTSP_SERVER_WMS) {
+            snprintf(cmd, sizeof(cmd) - 1,
+                     "GET_PARAMETER %s RTSP/1.0\r\n",
+                     s->filename);
+            rtsp_send_cmd_async(s, cmd, reply, NULL);
+        } else {
+            rtsp_send_cmd_async(s, "OPTIONS * RTSP/1.0\r\n",
+                                reply, NULL);
+        }
+    }
+
     return 0;
 }
 
Index: ffmpeg-svn/libavformat/rtsp.h
===================================================================
--- ffmpeg-svn.orig/libavformat/rtsp.h	2009-03-06 18:41:02.000000000 -0500
+++ ffmpeg-svn/libavformat/rtsp.h	2009-03-06 18:41:57.000000000 -0500
@@ -137,6 +137,14 @@
      * http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
      * for a complete list of supported values. */
     int notice;
+
+    /** The "timeout" comes as part of the server response to the "SETUP"
+     * command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
+     * time, in seconds, that the server will go without traffic over the
+     * RTSP/TCP connection before it closes the connection. To prevent
+     * this, sent dummy requests (e.g. OPTIONS) with intervals smaller
+     * than this value. */
+    int timeout;
 } RTSPMessageHeader;
 
 /**
@@ -195,6 +203,16 @@
      * identifier that the client should re-transmit in each RTSP command */
     char session_id[512];
 
+    /** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
+     * the server will go without traffic on the RTSP/TCP line before it
+     * closes the connection. */
+    int timeout;
+
+    /** timestamp of the last RTSP command that we sent to the RTSP server.
+     * This is used to calculate when to send dummy commands to keep the
+     * connection alive, in conjunction with \p timeout. */
+    int64_t last_cmd_time;
+
     /** the negotiated data/packet transport protocol; e.g. RTP or RDT */
     enum RTSPTransport transport;