Redirect on PLAY wasn't doing the necessary session cleanup. Fixed by
removing code from gst_rtspsrc_send that changed the state varable upon
encountering a redirect. Better to let the redirect handlers in
gst_rtspsrc_retrieve_sdp and gst_rtspsrc_play do their own
state-dependent cleanup.
https://bugzilla.gnome.org/show_bug.cgi?id=775543
Improve RFC2326 - chapter C.3 compatibility:
In case just a single stream is specified in SDP and the control attribute
is missing do not drop the stream but rather assume "a=control:*"
https://bugzilla.gnome.org/show_bug.cgi?id=770568
This solves a hanging mainloop in following scenario:
* connect to source
* network/server drops
* pipeline set to NULL (and connection to flushing as part)
* pipeline set to PAUSED/PLAYING (connection to non-flushing, but not recorded)
* [connecting still not possible]
* pipeline set to NULL => mainloop hangs (since no actual flushing is done)
Some servers add properties like charset, e.g.
application/sdp; charset=utf8
Ideally we should also parse the charset and do conversion of all messages,
but that's for a later time.
If we consider the RTSP state, what can happen is that it is PLAYING but the
element already asynchronously tried to PAUSE and it just did not happen yet.
We would then override this setting to PAUSED (while the element actually is
in PAUSED) and set the RTSP state to PLAYING again. This would then cause us
to produce packets while the sinks are all PAUSED, piling up thousands of
packets in the rtpjitterbuffer and other elements and finally failing.
The Session Data Protocol doesn't allow specifying a cipher for the
SRTCP, so it will use the SRTP one. In the "srtpenc" element the cipher
"aes-128-icm" is the default for SRTP and SRTCP, but if we want to have
an SRTCP with the "aes-256-icm" cipher then we also need to set the SRTP
cipher to "aes-256-icm", otherwise "aes-128-icm" will be used instead.
https://bugzilla.gnome.org/show_bug.cgi?id=767799
We only changed them for UDP so far, which caused the wrong seqnum-base and
other information to be passed to rtpjitterbuffer/etc when seeking. This
usually wasn't that much of a problem as the code there is robust enough, but
every now and then it causes us to drop up to 32756 packets before we
continue doing anything meaningful.
https://bugzilla.gnome.org/show_bug.cgi?id=765689
The server can send multiple crypto sessions, one for each SSRC with its
own rollover counter. We parse this information and pass it to the SRTP
decoder via the "request-key" signal.
https://bugzilla.gnome.org/show_bug.cgi?id=730540
Leverage response from gst_rtsp_connection_connect_with_response to
determine if the connection should be retried using authentication. If
so, add the appropriate authentication headers based upon the response
and retry the connection.
https://bugzilla.gnome.org/show_bug.cgi?id=749596
The string could exist but with a wrong format, in that case we still want
to reset the values of client_port_range.min and max like we do if there is
no string.
CID 1139593
In parse_keymgmt(), don't mutate the input string that's been passed
as const, especially since we might need the original value again if
the same key info applies to multiple streams (RTX, for example).
When a resource is 404, and we have auth info - retry with the auth
info the same as if we had receive unauthorised, in case the resource
isn't even visible until credentials are supplied.
Fix a memory leak handling Mikey data.
When generating a random keystring, don't overrun the 30 byte
buffer by generating 32 bytes into it.
Sometimes it is useful to know this information on the
server side. Other popular implementations (vlc, ffmpeg, ...)
also send this header on every message.
This includes a new "user-agent" property that the user
can set to use a custom User-Agent string. The default
is "GStreamer/<version>"
https://bugzilla.gnome.org/show_bug.cgi?id=750101
In media to caps function, reserved_keys array is being used for variable i,
leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
changed it to variable j
https://bugzilla.gnome.org/show_bug.cgi?id=753009
Skip keys from the fmtp, which we already use ourselves for the
caps. Some software is adding random things like clock-rate into
the fmtp, and we would otherwise here set a string-typed clock-rate
in the caps... and thus fail to create valid RTP caps
https://bugzilla.gnome.org/show_bug.cgi?id=753009
It is expected that buffers are time-stamped with running time. Set
a segment accordingly. In this case we pick 0,-1 as this is what udpsrc
would do. Depayloaders will update the segment to reflect the playback
position.
https://bugzilla.gnome.org/show_bug.cgi?id=635701
The segment start/stop in the query is meant to represent the seekable
portion of the stream. It does not match the segment start/stop. Instead
export 0 to duration.
We need to set up the transport in any case, not just if we have a container
stream or a non-interleaved stream. Only if we have an interleaved stream and
are retrying, we should not set up the stream again.
https://bugzilla.gnome.org/show_bug.cgi?id=745599
