Parse session control attributes when no media control attribute is
present. Threat * control attributes as an empty string, just like the
spec says.
Fixes#646800
... by forcing a state changed to PLAYING, which should otherwise be a
no-op as elements should already be in that state.
In particular, jitterbuffer needs new base_time as soon as possible to perform
proper timing (e.g. eos timeout handling) and can't wait for the new base_time
that will be distributed when the whole pipeline returns to PLAYING.
See bug #646397.
In case server-side fails to perform seek, i.e. PLAY at non-zero requested
position, recovery so far would arrange for streaming to continue, albeit
having lost position tracking in the process. So, query position prior
to seek and use upon failed seek.
Use g_ascii_dtostr() and g_ascii_strtod() to serialise/deserialise
floating point numbers, instead of ugly hacks that switch locale
before and after calling libc functions (which is not a good idea
in a multi-threaded application).
In particular, when streaming interleaved, this arranges for setting a new
timestamp on outgoing buffer so downstream can appropriate reset
to a change in (rtp)time.
Extend the _push_event() function so that it can also send events to the udp
sources when asked.
Implement a custum send_event function that correctly dispatches the downstream
events in TCP mode. This fixes sending EOS to rtspsrc and have it push the EOS
downstream.
Don't reuse sockets but make the udpsrc element fail the state change when the
socket is already in use. If we don't prevent reuse, we might end up using the same
port for different streams in some cases.
Fixes#622017
To support setups with firewall/ipsec, it is useful for an rtsp client to be
able to set the range of ports that can be used for rtp/rtcp reception.
Allows this by adding a "port-range" property to the rtspsrc element.
Fixes#625153
The RTSP server rtspsrc is communicating with, sends a GET_PARAMETER request
periodically as a ping. The code in gst_rtspsrc_handle_request forms an OK
response and sends, but doesn't call gst_rtsp_message_unset to free the memory
after sending the response. This results in a constant slow memory leak.
Fixes#624770
Keep a global connection for aggregate control but also keep stream connections
for non-aggregate control.
Add some helper methods to connect/close/flush the connections.
Parse the connection info from the SDP.
When we need to configure the multicast destination, fall back to the SDP
connection info when the transport did not specify a destination and ttl.
Fixes#617537
