I am writing my own network plugin that should transfer packets over SCTP.
I am passing the data into the SCTP socket from rtph264pay and on the receiver from the SCTP socket into rtph264depay but the stream cannot be decoded.
I checked hex dumps of the data and the incoming packets (from rtppay) are equivalent to the once outgoing (into rtpdepay).
The one message that really bothers me and I couldn't find any solution is:
I don't have all the debug output enabled:
sender:
GST_DEBUG="4,basesink:4,sctpsink:6,gstutils:5,basesrc:4,pushsrc:5,sctpsrc:5,rtph264pay:5,rtpbasepayload:5" ./build/rtpsctp /rtpsctpsend
send.txtreceiver:
GST_DEBUG="4,basesink:4,sctpsink:6,gstutils:5,basesrc:4,pushsrc:5,sctpsrc:5,rtph264depay:8,rtpbasedepayload:8"
recv.txtI am also dumpung the first 16bytes form the packet as a hex dump on every packet..
It seems like the RTP header starts at byte 3 -> what are the first 2 bytes?
I create the pipelines like this:
sender:
GstElement *pipeline, *source, *sink, *encoder, *rtppay;
/* create pipeline */
pipeline = gst_pipeline_new ("pipeline");
/* create element */
source = gst_element_factory_make("videotestsrc", "source");
if (!source) {
/* g_print ("Failed to create element of type 'videotestsrc'\n"); */
g_print ("Failed to create element of type 'videotestsrc'\n");
/* return -1; */
}
g_object_set(G_OBJECT(source),
/* "pattern", GstVideoTestSrcPattern.GST_VIDEO_TEST_SRC_SNOW, */
"is-live", TRUE,
NULL);
gst_util_set_object_arg(G_OBJECT(source), "pattern", "snow");
GstCaps *src_caps = gst_caps_new_simple ("video/x-raw",
"width", G_TYPE_INT, 1280,
"height", G_TYPE_INT, 720,
"framerate", GST_TYPE_FRACTION, 25, 1,
NULL);
encoder = gst_element_factory_make("x264enc", "encoder");
if (!encoder) {
g_print ("failed to create element of type 'x264enc'\n");
/* return -1; */
}
gst_util_set_object_arg(G_OBJECT(encoder), "tune", "zerolatency");
/* g_object_set(encoder, "tune", (GstX264EncTune)"zerolatency", NULL); */
rtppay = gst_element_factory_make("rtph264pay", "rtppay");
if (!rtppay) {
g_print ("failed to create element of type 'rtph264pay'\n");
/* return -1; */
}
receiver:
GstElement *pipeline = gst_pipeline_new ("pipeline");
/* create element */
GstElement *source = gst_element_factory_make("sctpsrc", "source");
if (!source) {
g_critical ("Failed to create element of type 'sctpsrc'\n");
/* return -1; */
}
g_object_set(G_OBJECT(source),
"port", 1117,
NULL);
GstCaps *src_caps = gst_caps_new_simple("application/x-rtp",
"media", G_TYPE_STRING, "video",
"clock-rate", G_TYPE_INT, 90000,
"encoding-name", G_TYPE_STRING, "H264",
"a-framerate", G_TYPE_STRING, "25",
"packetization-mode", G_TYPE_STRING, "1",
"payload", G_TYPE_INT, 96,
NULL);
GstElement *rtpdepay = gst_element_factory_make("rtph264depay", "rtpdepay");
if (!rtpdepay) {
g_critical ("failed to create element of type 'rtph264depay'\n");
/* return -1; */
}
GstElement *decoder = gst_element_factory_make("avdec_h264", "decoder");
if (!decoder) {
g_critical ("failed to create element of type 'avdec_h264'\n");
/* return -1; */
}
GstCaps *decode_caps = gst_caps_new_simple("video/x-h264",
"stream-format", G_TYPE_STRING, "avc",
"alignment", G_TYPE_STRING, "au",
"level", G_TYPE_STRING, "3.1",
"width", G_TYPE_INT, 1280,
"height", G_TYPE_INT, 720,
"framerate", GST_TYPE_FRACTION, 25, 1,
"pixel-aspect-ratio", GST_TYPE_FRACTION, 1, 1,
NULL);
GstElement *videoconvert = gst_element_factory_make("videoconvert", "videoconvert");
if (!videoconvert) {
g_critical ("failed to create element of type 'videoconvert'\n");
/* return -1; */
}
GstElement *videosink = gst_element_factory_make("ximagesink", "videsink");
if (!videosink) {
g_critical ("Failed to create element of type 'ximagesink'\n");
/* return -1; */
}
/* add to pipeline */
gst_bin_add_many(GST_BIN(pipeline), source, rtpdepay, decoder, videoconvert, videosink, NULL);
/* link */
/* rtpdepay, decoder, */
if (!gst_element_link_filtered(source, rtpdepay, src_caps)) {
g_warning ("Failed to link filterd source and rtpdepay!\n");
}
gst_caps_unref(src_caps);
if (!gst_element_link_filtered(rtpdepay, decoder, decode_caps)) {
g_critical ("Failed to link source and rtpdepay'\n");
}
gst_caps_unref(decode_caps);
if (!gst_element_link(decoder, videoconvert)) {
g_critical ("Failed to link source and rtpdepay'\n");
}
if (!gst_element_link(videoconvert, videosink)) {
g_critical ("Failed to link source and rtpdepay'\n");
}
I've implemented the .create() methos of PushSrc where I retrieve the data from the socket and store it into the give buffer:
static GstFlowReturn gst_sctpsrc_create(GstPushSrc *src, GstBuffer **buf) {
GstSctpSrc *sctpsrc = GST_SCTPSRC(src);
ssize_t n;
int flags;
char name[INET6_ADDRSTRLEN];
/* char *buffer = g_malloc(BUFFER_SIZE); */
/* buf = g_malloc(BUFFER_SIZE); */
/* [BUFFER_SIZE]; // FIXME: change to a malloc! */
socklen_t infolen;
struct sctp_rcvinfo rcv_info;
unsigned int infotype;
struct sockaddr_in6 from;
socklen_t from_len;
// allocate the buffer
GstMapInfo map;
*buf = gst_buffer_new_and_alloc (BUFFER_SIZE);
gst_buffer_map (*buf, &map, GST_MAP_READWRITE);
from_len = (socklen_t)sizeof(struct sockaddr_in6);
flags = 0;
infolen = (socklen_t)sizeof(struct sctp_rcvinfo);
n = usrsctp_recvv(sctpsrc->sock, (void *)map.data, BUFFER_SIZE, (struct sockaddr *)&from,
&from_len, (void *)&rcv_info, &infolen, &infotype, &flags);
if (n > 0) {
if (flags & MSG_NOTIFICATION) {
union sctp_notification *sn=(union sctp_notification*)map.data;
/* switch(sn->sn_header.sn_type) { */
/* case SCTP_COMM_UP: { */
GST_DEBUG_OBJECT(sctpsrc, "Notificatjion of type %u length %llu received.",
sn->sn_header.sn_type, (unsigned long long)n);
} else {
if (infotype == SCTP_RECVV_RCVINFO) {
GST_DEBUG_OBJECT(sctpsrc, "Msg len %4llu from [%s]:%u SID %d "
"SSN %u TSN %u PPID %u contxt %u compl %d",
(unsigned long long)n, inet_ntop(AF_INET6, &from.sin6_addr, name, INET6_ADDRSTRLEN),
ntohs(from.sin6_port), rcv_info.rcv_sid, rcv_info.rcv_ssn, rcv_info.rcv_tsn,
ntohl(rcv_info.rcv_ppid), rcv_info.rcv_context, (flags & MSG_EOR) ? 1 : 0);
} else {
GST_DEBUG_OBJECT(sctpsrc, "Msg of length %llu received from %s:%u, complete %d.",
(unsigned long long)n,
inet_ntop(AF_INET6, &from.sin6_addr, name, INET6_ADDRSTRLEN),
ntohs(from.sin6_port), (flags & MSG_EOR) ? 1 : 0);
}
print_rtp_header(sctpsrc, map.data+2);
hexDump(NULL, map.data, MIN(16, n));
/* hexDump(NULL, map.data, n); */
}
} else {
GST_WARNING_OBJECT(sctpsrc, "GST_FLOW_EOS");
return GST_FLOW_EOS;
}
gst_buffer_unmap (*buf, &map);
/* *buf = buffer; */
return GST_FLOW_OK;
}
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