RTCP packets were not sent because the same tr_cache_cookie was used for
both RTP and RTCP. So only one of the tr_cache lists were populated
depending on which one was sent first. If the tr_cache list is not
populated then no packets can be sent. Most often this happened to be
RTCP. Now seperate RTCP and RTP transport cache cookies are added which
resulted in both the tr_cache_lists to be populated regardless of which
one was sent first.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
RFC4566 Section 5.2 says that it should make the username, session id,
nettype, addrtype and unicast address tuple globally unique. Always using
1188340656180883 is not going to guarantee that: https://xkcd.com/221/
Instead let's create a 64 bit random number, which at least brings us
closer to the goal of global uniqueness.
https://tools.ietf.org/html/rfc4566#section-5.2
We add a trailing \0 in GstRTSPConnection to make parsing of
string message bodies easier (e.g. the SDP from DESCRIBE) but
for actual data this means we have to drop it or otherwise
create invalid data.
Fixes crash when two threads access handle_new_sample() at the same
time, one for RTP, one for RTCP.
Otherwise, when iterating over the transports cache, it might be modified by
another thread at the same time if the transports cookie has changed.
https://bugzilla.gnome.org/show_bug.cgi?id=742954
This reverts commit 935e8f852d.
RFC 2326 states that session IDs may consist of alphanumeric as well as
the safe characters $-_.+ -- N.B. the percent character is not allowed.
Previously the session ID was URI-escaped, this meant that any character
which was not alphanumeric or any of the characters +-._~ would be
percent encoded. While the RFC (surprisingly) mentions that linear white
space in session IDs should be URI-escaped, it does not say anything
about other characters. Moreover no white space is allowed in the
session ID. Finally the percent character which is the result of
URI-escaping is not allowed in a session ID.
So there is no reason to do any URI-escaping, and now it is removed.
https://bugzilla.gnome.org/show_bug.cgi?id=742869
rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
^
rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
^
The default implementation of configure_client_transport() in
rtsp-client uses the session media when it chooses channels for
interleaved traffic.
https://bugzilla.gnome.org/show_bug.cgi?id=739112
If the media has been managed by a session media, it should not be
cached in the client any longer. The GstRTSPSessionMedia object is now
responsible for unpreparing the GstRTSPMedia object using
gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
session media.
https://bugzilla.gnome.org/show_bug.cgi?id=739112
We need to set session medias to NULL without the client lock otherwise
we can end up in a deadlock if another thread is waiting for the lock
and media unprepare is also waiting for that thread to end.
https://bugzilla.gnome.org/show_bug.cgi?id=737690
If the backlog limit is kept two cases of deadlocks may be
encountered when streaming over TCP. Without the backlog
limit this deadlocks can not happen, at the expence of
memory usage.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
As long as gst-rtsp-server can successfully send RTP/RTCP data to
clients then the client must be reading. This change makes the server
timeout the connection if the client stops reading.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
Allow the send backlog in the RTSP watch to grow to unlimited size while
attempting to bring the media pipeline to NULL due to a session
expiring. Without this change the appsink element cannot change state
because it is blocked while rendering data in the new_sample callback.
This callback will block until it has successfully put the data into the
send backlog. There is a chance that the send backlog is full at this
point which means that the callback may block for a long time, possibly
forever. Therefore the media pipeline may also be prevented from
changing state for a long time.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
Just in case that guint8 doesn't fit in a pointer. Just in case ...
We need to raise the backlog limits before pausing the pipeline or else
the appsink might be blocking in the render method in wait_backlog() and
we would deadlock waiting for paused.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
link/unlink of the transport in a session was done to keep track of all
TCP transports and to send RTP/RTCP data to the streams. We can simplify
that by putting all the TCP transports in a hashtable indexed with the
channel number.
We also don't need to link/unlink the transports when we pause/resume
the streams. The same effect is already achieved when we pause/play the
media. Indeed, when we pause the media, the transport is removed from
the media and the callbacks will not be called anymore.
See https://bugzilla.gnome.org/show_bug.cgi?id=736041
Make a method to handle the data received on a channel. It sends the
data to the stream of the transport on the RTP or RTCP pads based on
the channel number.
The sequence number is not monotonic for RTP packets after pause. The
reason is basepayloader generates a randon sequence number when the
pipeline goes from ready to pause. With this fix generation of sequence
number will be monotonic when going from pause to play request.
https://bugzilla.gnome.org/show_bug.cgi?id=736017
Fixes a crash when close() is called while merging clients
in handle_tunnel(). In that case close() would destroy the
watch while it is still being used in handle_tunnel().
https://bugzilla.gnome.org/show_bug.cgi?id=735570
The RTCP parts, in specific the RTCP udpsinks, are not flushed when
seeking and will always continue counting the time. This leads to
the NPT after a backwards seek to be something completely different
to the actual seek position.
https://bugzilla.gnome.org/show_bug.cgi?id=732644
When a UDP multicast transport is used it is expected that the server listens
for RTP and RTCP packets on the multicast group with the corresponding port.
Without this we will never get RTCP packets from clients in multicast mode.
https://bugzilla.gnome.org/show_bug.cgi?id=732238
Release the object lock before calling the filter functions. We need to
keep a cookie to detect when the list changed during the filter
callback. We also keep a hashtable to make sure we only call the filter
function once for each object in case of concurrent modification.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
This extra ref will be dropped when all client sessions have been
removed. A session is removed when a client sends teardown, closes its
endpoint of the TCP connection or the sessions expires.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
Once we manage a media in a session, we can't unmanage it anymore
without destroying it. Therefore, first check everything before we
manage the media, otherwise if something is wrong we have no way to
unmanage the media.
If we created a new session and something went wrong, remove the session
again. Fixes a leak in the unit test.
Don't just keep a weak ref to the session objects but use a hard ref. We
will be notified when a session is removed from the pool (expired) with
the new session-removed signal.
Don't automatically close the RTSP connection when all the sessions of
a client are removed, a client can continue to operate and it can create
a new session if it wants. If you want to remove the client from the
server, you have to use gst_rtsp_server_client_filter() now.
Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
See https://bugzilla.gnome.org/show_bug.cgi?id=732226
Add encryption and authentication key length parameters to MIKEY. For
the encoders, the key lengths are obtained from the cipher and auth
algorithms set in the caps. For the decoders, they are obtained while
parsing the key management from the client.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
Coverity was moaning about unreachable code, and I think it was just
confused by { being before the label. We'll see if it pops up again.
Coverity 1197705
When we have too many messages queued for a client (currently hardcoded
to 100) we overflow and drop the messages. Add a drop-backlog property
to control this behaviour. Setting this property to FALSE will retry
to send the messages to the client by waiting for more room in the
backlog.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
Set our state to UNPREPARING and release the state-lock before
setting the pipeline to the NULL state. This way, any pad-added
callback will be able to take the state-lock and check that we are now
unpreparing instead of deadlocking.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
If we have multiple profiles (AVP or AVPF) for a stream, make one m=
line in the SDP for each profile. The client is then supposed to pick
one of the profiles in the SETUP request. Because the m= lines have the
same pt, the client also knows that only 1 option is possible.
Pass the multicast address of the stream in the connection info in the
SDP so that clients try a multicast connection first.
Only allow multicast connections in the test-multicast example. Also
increase the TTL a little.
If client supplied a range check if the range specifies the start point.
If not, then do an accurate seek to the current position. If a start
point was specified do do a key unit seek to make sure the streaming
starts with decodeable frames.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
in gst_rtsp_media_prepare() the thread is not used if media is already
prepared (e.g. media shared) so we want to stop the thread. otherwise, a
leak occurs.
https://bugzilla.gnome.org/show_bug.cgi?id=724182
Ignore streams that can't generate RTP-Info instead of failing.
Don't return the empty string when all streams are unconfigured but
return NULL so that we don't generate and empty RTP-Info header.
Improve docs a little.
Only set the keyframe flag when we modify the start position. The
keyframe flag should probably be ignored when no change is requested but
until we can claim this is all documented properly and all demuxer
implement this, avoid setting the flag.
See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
Add method to create the RTPInfo for a stream-transport.
Add method to create the RTPInfo for all stream-transports in a
session-media.
Use the session-media RTPInfo code in client. This allows us to refactor
another method to link the TCP callbacks.
Implement a vmethod that can be used to configure the media and the
streams based on the current context. Handle the blocksize handling in
the default handler.
See https://bugzilla.gnome.org/show_bug.cgi?id=720667
Add support for different suspend modes. The stream is suspended right after
producing the SDP and after PAUSE. Different suspend modes are available that
affect the state of the pipeline. NONE leaves the pipeline state unchanged and
is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
state and RESET will bring the pipeline to the NULL state.
A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
this means that the pipeline needs to be prerolled again.
Base on patches by Ognyan Tonchev <ognyan@axis.com>
See https://bugzilla.gnome.org/show_bug.cgi?id=711257
Start live streams in the blocked state and make them preroll using the
messages. This ensure that no data is played by the sink until we explicitly
unblock the stream right before going to PLAYING.
See https://bugzilla.gnome.org/show_bug.cgi?id=711257
* gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
trying to change to GST_STATE_NULL and media is in error status, we
remove all transports.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
* gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
* gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
element and create the stream with this one instead of the dynpay%d
element.
https://bugzilla.gnome.org/show_bug.cgi?id=712396
* rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
* rtsp-auth: Refer to part of constant name as text
* rtsp-auth/-permissions/-token: Refer to Permissions not Permission
* rtsp-session-media: Fix GstRTSPSessionMedia typo
* rtsp-stream: Fix typo when refering to GstBin
https://bugzilla.gnome.org/show_bug.cgi?id=714988
Use a guint instead of guint8 to increment the address. It's still not
completely correct because a guint might not be able to hold the complete
address range, but that's an enhacement for later.
Add unit test to test improved behaviour.
https://bugzilla.gnome.org/show_bug.cgi?id=708237
If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
don't actually stop the mainloop ever. Solve this race by adding an idle source
to the mainloop that calls the _quit. This way we immediately exit the mainloop
if quit was called before we started it.
Previously a role that was removed was unreffed twice, and when
replacing an existing role the replaced role was freed while still being
referenced. Both bugs are now fixed.
See https://bugzilla.gnome.org/show_bug.cgi?id=710202
If the SETUP url contains a query it must be appended to the control
path so that it matches any already created stream in the media. The
query will also be appended to the session media path.
We should not depend on whether or not the pipeline state change
returned NO_PREROLL or not. A media could dynamically change its
element and switch from seekable to non seekable so it's best to test
the seekable nature of the pipeline dynamically when we try to do a seek.
Only map the request url to a path in the DESCRIBE method. The SDP then
contains the base and control urls that should be used to SETUP/PAUSE/
PLAY/TEARDOWN the media.
This reverts commit e3fded2cec.
This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
contains the base and control urls which are used in the SETUP, PLAY,
PAUSE and TEARDOWN requests.
Make a vmethod to transform an url into a path. The path is then used to lookup
the factory. This makes it possible to also use other bits of the url, such as
the query parameters, to locate the factory.
Also fix race condition if two threads are asking for the first
thread from the thread pool at once. This would case two internal
GThreadPools to be created.
https://bugzilla.gnome.org/show_bug.cgi?id=707753
When no auth module is specified, use our table of defaults to look up the
default value of the check instead of always allowing everything. This was
we can disallow client settings by default.
If we try to reuse a thread right after we made it stop, we end up using a
stopped thread. Catch this case and only reuse threads that are not stopping.
Don't authorize on methods anymore but on the resources that we
try to access, this is more flexible.
Move the authorization checks to where they are needed and let the
check return the response on error.
Avoid passing GstStructure in the add_role method, use varargs instead
to construct the structure behind the scenes. We can then also use the
structure name as the role and simplify some more logic.
Move handling of the unauthorized response to the auth module, it can add
the appropriate headers to request authorization for the required method
much better than the client.
Start the media pipeline in the provided context (or our default one
when NULL). This makes sure that we run the bus thread in this context and that
all media threads are children of this context.
Add a simply miniobject that contains the authorizations. The object contains a
GstStructure that hold all authorization fields. When a user is authenticated,
the auth module will create a Token for the user. The token is then used to
check what operations the user is allowed to do and various other configuration
values.
Remove the auth object from media and factory. We want to have the RTSPClient
authenticate and authorize resources, there is no need to place another auth
manager on the media/factory.
Make it possible to add multiple basic authorisation tokens to one authorization
object. Associate with each token an authorization group that will define what
capabilities are allowed.
Cache the media in DESCRIBE based on the longest matching path with the uri
that we can find in the mount points.
Rework the setup request a little to get the media from the session or from
the longest matching path, this way we can derive the control string as
everything after the path instead of hardcoding it.
Find the stream based on the control string and only open a session when all
this can be done.
Making lookups in the mount points should not be done with a URL, if there is a
mapping to be done from URL to mount points, we'll need to do it somewhere
else.
Use a GSequence to keep track of the mount points.
Match a URL to the longest matching registered mount point. This should be the
URL to perform aggreagate control and the remainder is the stream specific
control part.
Add some unit tests for this.
Rename _get_address to _get_multicast_address in GstRTSPStream to
make it clear that this function only deals with multicast.
Make it possible to have both an IPv4 and IPv6 multicast address on
a stream. Give the client an IPv4 or IPv6 address depending on the
address it used to connect to the server.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
can't allocate any family at all. Also keep track of what port families we
allocated.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
Remove the get_uri() method on the client. A client has no uri, the uri
property is an internal property to manage the last cached media for
the client.
This patch makes configure_client_transport virtual. The functionality is
needed to handle some weird clients sending multicast transport settings as url
options.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
get_range_times worked for handling UTC ranges for seeks, but we also
need to convert back from NPT to the requested unit in
get_range_string. convert_range is now used for both.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
Make it possible to override the conversion from GstRTSPTimeRange to
GstClockTimes, that is done before seeking on the media
pipeline. Overriding can be useful for UTC ranges, where the default
conversion gives nanoseconds since 1900.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
This reverts commit 5fd034ff1a.
We already have a way to place extra attributes in the SDP by using a string
property with prefix x- or a- in the caps.
This reverts commit d6a4dee036.
We already have a way to place extra attributes in the SDP, just make a string
property in the payloader with a- or x- prefix.
application/x-rtp has properties with a- and x- prefixes that should be
placed as attributes in the SDP for the media instead of being added to the
fmtp.
Add methods to set and get a TLS certificate.
Add vmethod to configure a new connection. By default, configure the TLS
certificate in a new connection if needed.
Let the server accept the socket connection and construct a GstRTSPConnection
from it. Remove the code from the client and let the client only deal with
a fully configure GstRTSPConnection object.
We will need this later when the server will configure the connection for
TLS.
Listen to the pad-removed signal and remove the stream associated with the
removed pad.
Add signal to be notified of the removed pad.
Remove the fakesink in unprepare()
Fix signatures of the signal methods
The advantage is that in the signal handler you get direct access to
information about what streams are about to get torn down (in the
GstRTSPClientState).
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
Together with a shared clock, this base-time could eventually be sent to
the client so that it can reconstruct the exact running-time of the clock
on the server.
Add a property to let the media provide a GstNetTimeProvider for its clock.
Make methods to get the clock and nettimeprovider
Add a x-gst-clock property to the SDP with the IP and port number of the nettime
provider and also the current time of the clock. This should make it possible
for (GStreamer) clients to slave their clock to the server clock.
Protect the send_func with a lock. This allows us to wait for sending
to complete before changing the send_func and user_data. We add an
extra ref to the watch to make sure that it remains valid during
sending.
When closing the connection, set the send_func to NULL
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
This reverts commit ba5b78ff2f.
We can't use the refcount to trigger unprepare because it is the unprepare call
that removes the last refcount after all messages are consumed. What we should
probably do is make a prepared refcount and only unprepare when the refcount
reaches 0.
Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
isn't being used anymore.
Keep the factory in the state object only for authorization checks and make
sure we unref it on failure. Also don't keep invalid objects in the state
object.
We should use 454 when a session can't be found because there was no session
pool configured in the server. This is not a server configuration problem
because the server on which the request is done might not be the same one that
will keep the sessions for us and so it does not need to support sessions.
