We validate the header extensions length of an RTP buffer by comparing
it against the block size. Since we multiply the length in words by 4 to
get the length in bytes, a suitably large length could cause a wrapround
of the uint16, giving a lower length which erroneously passes the check
and allows the buffer to be mapped.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/864>
The wordlen ("length") MUST represent the total "number of 32-bit words
in the extension, excluding the four-octet extension header" (rfc3550).
There are cases where already existent padding is reused for adding
the new extension. So the new wordlen should be updated if the new
added extension makes it to increase.
According to RFC3611, the extended report blocks in XR packet can
have variable length. To visit each block, the iterator should look
into block header. Once XR type is extracted, users can parse the
detailed information by given functions.
Loss/Duplicate RLE
The Loss RLE and the Duplicate RLE have same format so
they can share parsers. For unit test, randomly generated
pseudo packet is used.
Packet Receipt Times
The packet receipt times report block has a list of receipt
times which are in [begin_seq, end_seq).
Receiver Reference Time paser for XR packet
The receiver reference time has ntptime which is 64 bit type.
DLRR
The DLRR report block consists of sub-blocks which has ssrc, last RR,
and delay since last RR. The number of sub-blocks should be calculated
from block length.
Statistics Summary
The Statistics Summary report block provides fixed length
information.
VoIP Metrics
VoIP Metrics consists of several metrics even though they are in
a report block. Data retrieving functions are added per metrics.
https://bugzilla.gnome.org/show_bug.cgi?id=789822
If timestamp goes forwards more than allowed, we consider that the
timestamp belongs to the previous counting, so the extended timestamp
is unwrapped.
https://bugzilla.gnome.org/show_bug.cgi?id=783443
Encrypted RTP buffers may contain encrypted padding, hence it's
necessary to have an option to relax the validation in order to
successfully map the buffer.
When the flag GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is set
gst_rtp_buffer_map() will map the buffer like if padding is not
present.
https://bugzilla.gnome.org/show_bug.cgi?id=752705
The padding (if any) is included in the length of the last packet, see
RFC 3550.
Section 6.4.1:
padding (P): 1 bit
If the padding bit is set, this individual RTCP packet contains
some additional padding octets at the end which are not part of
the control information but are included in the length field. The
last octet of the padding is a count of how many padding octets
should be ignored, including itself (it will be a multiple of
four).
Section A.2:
* The padding bit (P) should be zero for the first packet of a
compound RTCP packet because padding should only be applied, if it
is needed, to the last packet.
* The length fields of the individual RTCP packets must add up to
the overall length of the compound RTCP packet as received.
https://bugzilla.gnome.org/show_bug.cgi?id=751883
According to this section of the rfc.
https://tools.ietf.org/html/rfc5506#section-3.4.2
The validation should be updated to accept more types of RTCP
packages, with this mask change feedback packages will be also
accepted.
Change-Id: If5ead59e03c7c60bbe45a9b09f3ff680e7fa4868
The function gst_rtp_buffer_get_payload can not be used in Python
because it lacks necessary length parameter. This patch adds a new
function, gst_rtp_buffer_get_payload_bytes, to use from Python
bindings. The new function has the advisory "Rename to:" annotation
so it can replace the gst_rtp_buffer_get_payload whan creating
bindings.
The function gst_rtp_buffer_get_extension_bytes is also added. It wraps
gst_rtp_buffer_get_extension_data which doesn't work in Python due to
incomplete annotation and because it returns the length as number of
32-bit words.
https://bugzilla.gnome.org/show_bug.cgi?id=698562
Allocate header, payload and padding in separate memory blocks in
gst_rtp_buffer_allocate().
don't use part of the payload data as storage for the extension data but store
it in a separate memory block that can be enlarged when needed.
Rework the one and two-byte header extension to make it reserve space for the
extra extension first.
Fix RTP unit test. Don't map the complete buffer or make assumptions on the
memory layout of the underlaying implementation. We can now always add extension
data because we have a separate memory block for it.
Add support RTP buffers with multiple memory blocks. We allow one block for the
header, one for the extension data, N for data and one memory block for the
padding.
Remove the validate function, we validate now when we map because we need to
parse things in order to map multiple memory blocks.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_event):
Remove erroneous gst_buffer_ref().
* tests/check/libs/rtp.c: (GST_START_TEST):
Don't forget to unref the buffer once you're done with it.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester ca>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement gst_rtcp_packet_remove(). Fixes#563174.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add unit test for some RTCP functions.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain):
Validate the RTP packet before further processing it. It's just too
dangerous to accept random packets and people are not forced to use a
jitterbuffer or session manager to filter out the bad packets.
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data),
(gst_rtp_buffer_get_payload_subbuffer):
Small cleanups.
When setting extension data in a buffer that is too small, we fail and
we should not set the extension bit.
Change GST_WARNINGS into g_warning because they really are
programming errors.
* tests/check/libs/rtp.c: (GST_START_TEST):
Catch the g_warnings now in the unit tests and that fact that failing to
set extension data left the extension bit untouched.
Original commit message from CVS:
Patch by: Bernard B <b-gnome at largestprime dot net>
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
Fix seqnum compare function for bordercase values and fix the docs
again. Fixes#533075.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add a testcase for seqnum compare function.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data):
* gst-libs/gst/rtp/gstrtpbuffer.h:
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add gst_rtp_buffer_set_extension_data()
Add a unit test for this addition. Fixes#511478.
API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
Original commit message from CVS:
* tests/check/Makefile.am:
Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
* tests/check/elements/audiorate.c: (do_perfect_stream_test):
* tests/check/elements/playbin.c:
* tests/check/libs/mixer.c: (test_element_interface_supported),
(gst_implements_interface_init):
* tests/check/libs/rtp.c: (GST_START_TEST):
Fix various assignment type mismatches.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* tests/check/libs/rtp.c:
Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
Original commit message from CVS:
Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
Fix up GstRTPHeader helper struct so that compilers will not under
any circumstances add padding in between our fields, as currently
happens with MSVC on win32, because that would lead to us sending
out RTP payloads with broken RTP headers (#471194).
Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
* tests/check/Makefile.am:
* tests/check/libs/.cvsignore:
* tests/check/libs/rtp.c:
Add some simple unit tests for GstRTPBuffer. Some are disabled
because the code tested still needs fixing (set_csrc() does not work).
