Commit 1e17aba3 authored by Kristofer's avatar Kristofer Committed by Tim-Philipp Müller

rtsp-client: RTP Info exists conditionally in PLAY

If RTP Info is missing and it is not a receiver only, eg. audio
backchannel. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR.
In rfc2326 it says RTP-info is req. but in RFC7826 it is conditional.

Since 1.14 there is audio backchannel support. Thus RTP-info is
conditional now. When audio backchannel only mode, there is no RTP-info.

Fixes #82
parent 491631a6
Pipeline #68507 failed with stages
in 26 minutes and 5 seconds
......@@ -1741,7 +1741,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
GstRTSPResult res;
GstRTSPState rtspstate;
GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
gchar *path, *rtpinfo;
gchar *path, *rtpinfo = NULL;
gint matched;
gchar *seek_style = NULL;
GstRTSPStatusCode sig_result;
......@@ -1830,7 +1830,8 @@ handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
}
/* grab RTPInfo from the media now */
if (!(rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia)))
if (!gst_rtsp_media_is_receive_only (media) &&
!(rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia)))
goto rtp_info_error;
/* construct the response now */
......@@ -1839,7 +1840,9 @@ handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
gst_rtsp_status_as_text (code), ctx->request);
/* add the RTP-Info header */
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO, rtpinfo);
if (rtpinfo)
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
rtpinfo);
if (seek_style)
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_SEEK_STYLE,
seek_style);
......
......@@ -54,6 +54,51 @@ test_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
return TRUE;
}
static gboolean
test_response_play_200 (GstRTSPClient * client, GstRTSPMessage * response,
gboolean close, gpointer user_data)
{
GstRTSPStatusCode code;
const gchar *reason;
GstRTSPVersion version;
gchar *str;
gchar **session_hdr_params;
gchar *pattern;
fail_unless_equals_int (gst_rtsp_message_get_type (response),
GST_RTSP_MESSAGE_RESPONSE);
fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
&version)
== GST_RTSP_OK);
fail_unless_equals_int (code, GST_RTSP_STS_OK);
fail_unless_equals_string (reason, "OK");
fail_unless_equals_int (version, GST_RTSP_VERSION_1_0);
/* Verify mandatory headers according to RFC 2326 */
/* verify mandatory CSeq header */
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
0) == GST_RTSP_OK);
fail_unless (atoi (str) == cseq++);
/* verify mandatory Session header */
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION,
&str, 0) == GST_RTSP_OK);
session_hdr_params = g_strsplit (str, ";", -1);
fail_unless (session_hdr_params[0] != NULL);
g_strfreev (session_hdr_params);
/* verify mandatory RTP-Info header */
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_RTP_INFO,
&str, 0) == GST_RTSP_OK);
pattern = g_strdup_printf ("^url=rtsp://.+;seq=[0-9]+;rtptime=[0-9]+");
fail_unless (g_regex_match_simple (pattern, str, 0, 0),
"GST_RTSP_HDR_RTP_INFO '%s' doesn't match pattern '%s'", str, pattern);
g_free (pattern);
return TRUE;
}
static gboolean
test_response_400 (GstRTSPClient * client, GstRTSPMessage * response,
gboolean close, gpointer user_data)
......@@ -1707,6 +1752,56 @@ GST_START_TEST (test_client_multicast_invalid_ttl)
GST_END_TEST;
GST_START_TEST (test_client_play)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
gchar *str;
GstRTSPContext ctx = { NULL };
client = setup_multicast_client (1);
ctx.client = client;
ctx.auth = gst_rtsp_auth_new ();
ctx.token =
gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
"user", NULL);
gst_rtsp_context_push_current (&ctx);
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
"RTP/AVP;multicast");
/* destination is from adress pool */
expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=.*;mode=\"PLAY\"";
gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
expected_transport = NULL;
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
gst_rtsp_client_set_send_func (client, test_response_play_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
send_teardown (client);
teardown_client (client);
g_object_unref (ctx.auth);
gst_rtsp_token_unref (ctx.token);
gst_rtsp_context_pop_current (&ctx);
}
GST_END_TEST;
static Suite *
rtspclient_suite (void)
{
......@@ -1759,6 +1854,7 @@ rtspclient_suite (void)
tcase_add_test (tc, test_client_multicast_max_ttl_first_client);
tcase_add_test (tc, test_client_multicast_max_ttl_second_client);
tcase_add_test (tc, test_client_multicast_invalid_ttl);
tcase_add_test (tc, test_client_play);
return s;
}
......
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