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.
