The initial pipeline does not contain specific transport
elements. The receiver and the sender parts are added
after PLAY.
If the media is shared, the streams are dynamically
reconfigured after each PLAY.
https://bugzilla.gnome.org/show_bug.cgi?id=788340
If no sinks have been added yet, obtain the current and
the stop position of the stream from the send_src pad.
Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
https://bugzilla.gnome.org/show_bug.cgi?id=788340
According to the documentation, a timeout of value 0 means
that the session never timeouts. This adds handling of that.
If timeout=0 we just return with a -1 from
gst_rtsp_session_next_timeout_usec ().
https://bugzilla.gnome.org/show_bug.cgi?id=785058
This adds basic support for new 2.0 features, though the protocol is
subposdely backward incompatible, most semantics are the sames.
This commit adds:
- features:
* version negotiation
* pipelined requests support
* Media-Properties support
* Accept-Ranges support
- APIs:
* gst_rtsp_media_seekable
The RTSP methods that have been removed when using 2.0 now return
BAD_REQUEST.
https://bugzilla.gnome.org/show_bug.cgi?id=781446
Commit 852cc09f54 assumed that
multiudpsink's last-sample always comes from the payloader. Which
is wrong if auxiliary streams are multiplexed in the same stream.
So check the buffer's ssrc against the caps'ssrc before to use its
seqnum. If not the same ssrc just use the payloader as done prior
the commit above or when there is no last-sample yet.
https://bugzilla.gnome.org/show_bug.cgi?id=784094
Calling function gst_rtsp_stream_get_server_port() results in
segmenation fault in the RTP/RTSP/TCP case.
Port that the server will use to receive RTCP makes only
sense in the UDP case, however the function should handle
the TCP case in a nicer way.
https://bugzilla.gnome.org/show_bug.cgi?id=776345
85c52e194b introduced a more correct
detection of the srtp rollover counter to add to the SDP.
Unfortunately, it was incomplete for live pipelines where the logic
blocks the source bin before creating the SDP and thus would never have
the necessary informaiton to create a correct SDP with srtp encryption.
Move the pad blocks to rtpbin's output pads instead so that the
necessary information can be created before we need the information for
the SDP.
https://bugzilla.gnome.org/show_bug.cgi?id=770239
The RTSP server will not timeout an idle RTSP connection
(note this is different from doing timeout on a RTSP
session).
At least for Apache this is a problem when running RTSP over
HTTPS since it uses one of the threads (there is a rather
limited number) that are available for handling requests.
https://bugzilla.gnome.org/show_bug.cgi?id=771830
With this RTSP server can use the sockets independent on the udpsrc
state.
When the udp src is finalized it will unref socket and when g_socket
is finalized the socket will be closed.
https://bugzilla.gnome.org/show_bug.cgi?id=765673
These signals let the application validate the requests, configure the
media/stream in a certain way and also generate error status code in
case of error or bad request.
https://bugzilla.gnome.org/show_bug.cgi?id=758062
Call session filter with filter_session_media as paramer in
client_unwatch_session if using drop_backlog = FALSE.
In client_unwatch_session its allowed to grow the watchs backlog.
If using drop_backlog = FALSE and the backlog is full it will cause
a deadlock when setting session media state to NULL
if the backlog is not allowed to grow.
https://bugzilla.gnome.org/show_bug.cgi?id=771983
When using dynamic elements, gst_rtsp_stream_join_bin() is called from
"pad-added" signal. In that case priv->srcpad could already have its caps,
and they'll be sent to priv->send_src[0] pad. That means that when it
connects "notify::caps" signal, that pad could already have received its
caps and the signal won't be emitted anymore.
In that case priv->caps stay to NULL and when building the SDP that stream
gets ignored. Leading to missing video or audio when playing in client side.
https://bugzilla.gnome.org/show_bug.cgi?id=772478
Adding them later will cause deadlocks due to
1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
2) adding the multicast sink
3) waiting for it to get data to preroll again
3) never happens because the queues after the tee are full.
This is basically reverting changes introduced in commit f62a9a7,
because it was introducing various regressions:
- It introduces a leak of udpsrc elements that got wrongly fixed by adding
an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
- If a mcast client connects, it creates a new socket in SETUP to try to respect
the destination/port given by the client in the transport, and overrides the
socket already set on the udpsink element. That means that if we already had a
client connected, the source address on the udp packets it receives suddenly
changes.
- If a 2nd mcast client connects, the destination/port in its transport is
ignored but its transport wasn't updated.
What this patch does:
- Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
- Always have a tee+queue when udp is enabled. This could be optimized
again in a later patch, but is more complicated. If no unicast clients
connects then those elements are useless, this could be also optimized
in a later patch.
- When mcast transport is added, it creates a new set of udpsrc/udpsink,
seperated from those for unicast clients. Since we already support only
one mcast address, we also create only one set of elements.
https://bugzilla.gnome.org/show_bug.cgi?id=766612
We add different crypto sessions in MIKEY, one for each sender
SSRC. Currently, all of them will have the same security policy, 0.
The rollover counters are obtained from the srtpenc element using the
"stats" property.
https://bugzilla.gnome.org/show_bug.cgi?id=730539
It's what introspection.mak does as well. Should
fix spurious build failures on gnome-continuous
(caused by g-ir-scanner getting compiler details
via python which is broken in some environments
so passing the compiler details bypasses that).
- Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
- Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
- Create unit test for shared media.
https://bugzilla.gnome.org/show_bug.cgi?id=764744
For NTP and PTP clocks we signal the actual clock that is used and signal
the direct media clock offset.
For all other clocks we at least signal that it's the local sender clock.
This allows receivers to know which clock was used to generate the media and
its RTP timestamps. Receivers can then implement network synchronization,
either absolute or at least relative by getting the sender clock rate directly
via NTP/PTP instead of estimating it from RTP timestamps and packet receive
times.
https://bugzilla.gnome.org/show_bug.cgi?id=760005
Without this, RECORD pipelines are broken because
a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
added later. Previously it was there earlier and due to NO_PREROLL caused the
pipeline to preroll immediately
b) the udpsrc for the pipeline is added later and never set to PLAYING state,
as the corresponding code previously was only for PLAY pipelines.
https://bugzilla.gnome.org/show_bug.cgi?id=763281
On Windows this is a receiver-side setting, on Linux a sender-side setting. As
we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
loopback setting on the socket... while udpsink does which unfortunately has
no effect here on Windows but on Linux.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
On Linux it is still needed to bind to the multicast address
to filter out random other packets, while on Windows binding
to multicast addresses just fails.
Otherwise we fail to allocate UDP ports if the pool only contains multicast
addresses, which is something that used to work before. For unicast addresses
if the pool contains none, we just allocate them as if there is no pool at
all.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
Postpone the allocation of the UDP sockets until we know
what transport has been chosen by the client.
Both unicast and multicast UDP sources are created in one
function.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
Code refactoring: allocate the UDP ports after the sender and
the reciver parts have been created.
We postpone the creation of the UDP sources until the UDP
ports have been allocated.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
Add a boolean to indicate that the rtsp-stream is running on the
'client' side of an RTSP connection, for sending streams via
RECORD. In that case, the roles of the client/server ports
in transport setup are swapped.
https://bugzilla.gnome.org/show_bug.cgi?id=758180
When RTSP server trying update transport during multicast, it throws an
assert. The assert is thrown because it is trying to get the parent of
an non-existing funnel element.
https://bugzilla.gnome.org/show_bug.cgi?id=760150
