Commit Graph

118 Commits

Author SHA1 Message Date
Martin Storsjö
424da30830 rtsp: Support decryption of SRTP signalled via RFC 4568 (SDES)
This only takes care of decrypting incoming packets; the outgoing
RTCP packets are not encrypted. This is enough for some use cases,
and signalling crypto keys for use with outgoing RTCP packets
doesn't fit as simply into the API. If the SDP demuxer is hooked
up with custom IO, the return packets can be encrypted e.g. via the
SRTP protocol.

If the SRTP keys aren't available within the SDP, the decryption
can be handled externally as well (when using custom IO).

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-15 11:54:40 +02:00
Martin Storsjö
86d9181cf4 rtpdec: Support sending RTCP feedback packets
This sends NACK for missed packets and PLI (picture loss indication)
if a depacketizer indicates that it needs a new keyframe, according
to RFC 4585.

This is only enabled if the SDP indicated that feedback is supported
(via the AVPF or SAVPF profile names).

The feedback packets are throttled to a certain maximum interval
(currently 250 ms) to make sure the feedback packets don't eat up
too much bandwidth (which might be counterproductive). The RFC
specifies a more elaborate feedback packet scheduling.

The feedback packets are currently sent independently from normal
RTCP RR packets, which is not totally spec compliant, but works
fine in the environments I've tested it in. (RFC 5506 allows this,
but requires a SDP attribute for enabling it.)

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-08 17:48:14 +02:00
Martin Storsjö
e96406eda4 rtsp: Add support for depacketizing RTP data via custom IO
To use this, set sdpflags=custom_io to the sdp demuxer. During
the avformat_open_input call, the SDP is read from the AVFormatContext
AVIOContext (ctx->pb) - after the avformat_open_input call,
during the av_read_frame() calls, the same ctx->pb is used for reading
packets (and sending back RTCP RR packets).

Normally, one would use this with a read-only AVIOContext for the
SDP during the avformat_open_input call, then close that one and
replace it with a read-write one for the packets after the
avformat_open_input call has returned.

This allows using the RTP depacketizers as "pure" demuxers, without
having them tied to the libavformat network IO.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-03 15:15:27 +02:00
Martin Storsjö
e0d5ac6ae3 rtsp: Update a comment to the current filename scheme
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-10-22 01:46:10 +03:00
Martin Storsjö
3f055f8f5f rtsp: Allow setting the reordering buffer size via an AVOption
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-10-18 23:10:48 +03:00
Martin Storsjö
1c37744963 rtsp: Vertically align a constant definition
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-10-18 23:10:42 +03:00
Martin Storsjö
1243c72251 rtsp: Support mpegts in raw udp packets
This is basically the same way as mpegts packets are parsed in
rtpdec.c.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-08-09 00:25:57 +03:00
Martin Storsjö
df8cf076c8 rtsp: Support receiving plain data over UDP without any RTP encapsulation
EvoStream Media Server can serve data in this format, and
VLC/live555 already supports it.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-08-09 00:25:15 +03:00
Jordi Ortiz
a8ad6ffafe rtsp: Add listen mode
This makes the RTSP demuxer act as a server, listening for an
incoming connection.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-10 22:00:28 +03:00
Jordi Ortiz
6e71c1202b rtsp: Make rtsp_open_transport_ctx() non-static
This is required for the upcoming RTSP listen mode.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-10 21:21:19 +03:00
Jordi Ortiz
45b068580b rtsp: Parse the mode=receive/record parameter in transport lines
We need to support the nonstandard mode=receive, for compatibility
with older libavformat clients.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-10 21:20:04 +03:00
Jordi Ortiz
fcd0298c05 rtsp: Add content-type message header parsing
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2012-05-08 10:18:35 -07:00
Martin Storsjö
dbb06b8c0d rtsp: Allow specifying the UDP port range via AVOptions
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-01-22 01:10:02 +02:00
Martin Storsjö
1f712e6a05 rtsp: Remove extern declarations for variables that don't exist
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-01-21 01:12:07 +02:00
Diego Biurrun
58c42af722 doxygen: misc consistency, spelling and wording fixes 2011-12-12 23:06:23 +01:00
John Brooks
f011fcd67e rtsp: add allowed_media_types option
Streams from RTSP or SDP that do not match an allowed type will
be skipped entirely, which allows video-only or audio-only
streaming from servers that provide both.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-02 21:37:46 +02:00
Martin Storsjö
9867aea524 rtsp: Remove the separate filter_source variable
Read it as a flag from the flags field instead.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-17 19:57:49 +03:00
Martin Storsjö
eca4850c6d rtsp: Accept options via private avoptions instead of URL options
Eventually, the old way of passing options by adding
stuff to the URL can be dropped.

This avoids having to tamper with the user-specified URL to
pass options on the transport mode. This also works better
with redirects, since the options don't need to be parsed out
from the URL.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-17 19:57:48 +03:00
Martin Storsjö
17fff881e7 rtsp: Merge the AVOption lists
This eases adding options that are common for both. The
AV_OPT_FLAG_EN/DECODING_PARAM still indicates whether they belong
to the muxer or demuxer.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-17 19:57:45 +03:00
Martin Storsjö
30eae32530 rtsp: Parse the x-Accept-Dynamic-Rate header
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-12 14:48:45 +03:00
Diego Biurrun
76e25dbca6 rtsp: remove disabled code 2011-07-18 18:22:02 +02:00
Diego Biurrun
96c1e6d40d doxygen: Make sure parameter names match between .c and .h files. 2011-07-14 04:09:49 +02:00
Diego Biurrun
f75e3da535 RTSP: Doxygen comment cleanup
Do not use Doxygen for comments that apply to specific implementation
details; merge some duplicated Doxygen comment blocks.
2011-07-03 22:33:22 +02:00
Martin Storsjö
e2e29c6247 rtspenc: Add RTP muxer options
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-06-10 10:52:22 +03:00
Anton Khirnov
4779f59378 rtspdec: add initial_pause private option.
Deprecate corresponding AVFormatParameters field.
2011-05-27 06:52:52 +02:00
Martin Storsjö
0b4949b518 rtsp: Only do keepalive using GET_PARAMETER if the server supports it
This is more like what VLC does. If the server doesn't mention
supporting GET_PARAMETER in response to an OPTIONS request,
VLC doesn't send any keepalive requests at all. After this patch,
libavformat will still send OPTIONS keepalives if GET_PARAMETER
isn't explicitly said to be supported.

Some RTSP cameras don't support GET_PARAMETER, and will
close the connection if this is sent as keepalive request
(but support OPTIONS just fine, but probably don't need any
keepalive at all). Some other cameras don't support using
OPTIONS as keepalive, but require GET_PARAMETER instead.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-05-11 10:42:34 +03:00
Mans Rullgard
2912e87a6c Replace FFmpeg with Libav in licence headers
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-03-19 13:33:20 +00:00
Anton Khirnov
ae628ec1fd avio: rename ByteIOContext to AVIOContext.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-20 08:37:15 -05:00
Luca Barbato
a8475bbdb6 os: replace select with poll
Select has limitations on the fd values it could accept and silently
breaks when it is reached.
2011-01-28 15:45:19 +01:00
Diego Elio Pettenò
57c4d01ec9 Make ff_rtsp_send_cmd_with_content_async static to rtsp.c.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-25 22:10:36 +01:00
Martin Storsjö
2762a7a28b rtspdec: Retry with TCP if UDP failed
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:49:36 +01:00
Martin Storsjo
93e7490ee0 rtsp: Split out a function undoing the setup made by ff_rtsp_make_setup_request
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:46:39 +01:00
Martin Storsjo
fef5649a82 rtsp: Make make_setup_request a nonstatic function
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:46:36 +01:00
Martin Storsjö
a92c30d76e rtsp: Allow requesting of filtering of source packets
If filtered, only packets from the right source address and port
are received.

To test, play back e.g. some mpeg4 video RTSP stream (where the
video stream is the first stream in the presentation) over UDP.
While receiving this stream, send another stream to the same port:
ffmpeg -re -i <whatever> -vcodec mpeg4 -an -f rtp
rtp://127.0.0.1:5000?localport=1234
Normally, the RTSP playback reports lots of errors at this point.

If the RTSP stream has the ?filter_src option enabled, these
interferring packets are ignored.

Originally committed as revision 26246 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-06 15:22:58 +00:00
Martin Storsjö
d2995eb910 rtsp: Store the Content-Base header value straight to the target
This avoids having a large temporary buffer in the struct used for
storing the rtsp reply headers.

Originally committed as revision 26192 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-02 10:11:12 +00:00
Martin Storsjö
77223c5388 rtsp: Pass the method name to ff_rtsp_parse_line
Originally committed as revision 26191 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-02 10:10:12 +00:00
Martin Storsjö
acc9ed1450 rtsp: Pass RTSPState to ff_rtsp_parse_line, instead of HTTPAuthState
This allows ff_rtsp_parse_line to do more changes directly in RTSPState
when parsing the reply, instead of having to store large amounts of
temporary data in RTSPMessageHeader.

Originally committed as revision 26190 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-02 10:07:56 +00:00
Martin Storsjö
3df54c6bf2 rtsp: Add a method parameter to ff_rtsp_read_reply
Originally committed as revision 26189 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-02 10:06:21 +00:00
Martin Storsjö
dd22cfb101 rtsp: Parse and use the Content-Base reply header, if present
This fixes playing RTSP urls with query parameters.

Originally committed as revision 25755 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-11-15 15:08:53 +00:00
Martin Storsjö
0526c6f7c7 rtsp: Split out the RTSP demuxer functions to a separate, new file
Originally committed as revision 25601 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-29 08:43:57 +00:00
Martin Storsjö
c2688f3ac8 rtsp: Move rtsp_setup_output_streams into rtspenc.c
Originally committed as revision 25600 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-29 08:41:49 +00:00
Aurelien Jacobs
a5cea13202 drop rtsp_default_protocols which is not part of public API and not used anymore
Originally committed as revision 25557 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-23 16:22:36 +00:00
Martin Storsjö
9e6acc7884 rtsp: Remove the start_time field from RTSPState, use AVFormatContext->start_time_realtime instead
Originally committed as revision 25408 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-08 08:50:29 +00:00
Martin Storsjö
96a7c9753e rtsp: Use a dynamically allocated receive buffer
Originally committed as revision 25288 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-01 17:41:31 +00:00
John Wimer
619298a84d Send NAT punching messages to the address specified in the Transport:
message, if available (RFC 2326, section 12.39), fixes issue 2212.

Patch by John Wimer <john at god vtic net>.

Originally committed as revision 25032 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-09-03 19:25:59 +00:00
Josh Allmann
b20359f51a rtsp: Return AVERROR_EOF when all streams have received an RTCP BYE packet
Patch by Josh Allmann, joshua dot allmann at gmail

Originally committed as revision 24965 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-29 10:25:16 +00:00
Martin Storsjö
7934b15d5a Handle IPv6 in the RTSP code
Originally committed as revision 24925 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-25 15:32:29 +00:00
Martin Storsjö
3fbd12d109 Handle IPv6 in the SDP demuxer
Originally committed as revision 24924 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-25 15:32:00 +00:00
Aurelien Jacobs
be73ba2fa4 get rid of MAX_STREAMS limit in RTSP
Originally committed as revision 24752 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-09 23:00:13 +00:00
Luca Barbato
d93fdcbf5c Preserve status reason
It is used to provide meaningful error messages.

Originally committed as revision 24714 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-06 10:26:30 +00:00