Untested, due to lack of rtp stream with CSRCs, but the code as
is does not work with multiple CSRCs
Reviewed-by: Luca Abeni <lucabe72@email.it>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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>
Without this, we'd signal a huge loss rate (due to unsigned
wraparound) if we had received one packet more than expected (that
is, one seq number sent twice). The code has a check for lost_interval
<= 0, but that doesn't do what was intended as long as the variable is
unsigned.
Signed-off-by: Martin Storsjö <martin@martin.st>
The code below the comment does not at all relate to statistics,
and even if moved to the right place, the comment adds little
value.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously, we always signalled a zero time since the last RTCP
SR, which is dubious.
The code also suggested that this would be the difference in
RTP NTP time units (32.32 fixed point), while it actually is
in in 1/65536 second units. (RFC 3550 section 6.4.1)
Signed-off-by: Martin Storsjö <martin@martin.st>
This brings back some code that was added originally in 4a6cc061
but never was used, and was removed as unused in 4cc843fa. The
code is updated to actually work and is tested to return sane
values.
Signed-off-by: Martin Storsjö <martin@martin.st>
The base_seq variable is set to first_seq - 1 (in
rtp_init_sequence), so no + 1 is needed here.
This avoids reporting 1 lost packet from the start.
Signed-off-by: Martin Storsjö <martin@martin.st>
The question can be answered: No, we do not know the initial sequence
number from the SDP. In certain cases, it can be known from the
RTP-Info response header in RTSP though. (In that case, we use it as
timestamp origin, but not for rtp receiver statistics.)
Signed-off-by: Martin Storsjö <martin@martin.st>
This is required by RFC 3550 (section 6.5):
The list of items in each chunk MUST be terminated by one or more
null octets, the first of which is interpreted as an item type of
zero to denote the end of the list.
This was implicitly added as padding before, unless the host name
length matched up so no padding was added.
This makes wireshark parse the packets properly if other RTCP items
are appended to the same packet.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
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>
This comment was added in e309128f, in 2002, and has been brought
along since then more or less unmodified.
The first point of the todo was implemented in dbf30963 in 2006,
the second one is not relevant to rtpdec.c (brought along from
rtp.c in 8eb793c4 in 2008) but would be more relevant to the
rtp muxer, although it isn't a good idea anyway.
Signed-off-by: Martin Storsjö <martin@martin.st>
Mainly clean up the RTP statistics code, plus a few other obviously
misindentend lines.
Remove some useless comments, de-doxygenize some comments,
add spacing around operators and fix a typo.
Signed-off-by: Martin Storsjö <martin@martin.st>
RTPDynamicProtocolHandler for speex is added. Initial support for
speex depacketization from RTP stream comes with it.
Currently, only codec audio rate can be applied based on sdp:
* Narrowband ( 8K)
* Wideband (16K)
* Ultrawideband (32K)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
unistd.h used to be required for gethostname. On windows, gethostname
is provided by winsock2.h. Now network.h includes both unistd.h and
winsock2.h if they exist.
Signed-off-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
unistd.h used to be required for gethostname. On windows, gethostname
is provided by winsock2.h. Now network.h includes both unistd.h and
winsock2.h if they exist.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids exposing a dummy AVStream which won't get any data
and which will make avformat_find_stream_info wait for info about
this stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is different from the "modern" RTP payload formats for H263
as defined by RFC 4629, 2429 and 3555. According to the newer RFCs,
this old one is to be considered deprecated and only be used for
interoperating with legacy systems.
Signed-off-by: Martin Storsjö <martin@martin.st>
The s->ssrc field is the sender's SSRC, we use ssrc + 1 to get
a collision free "unique" SSRC for ourselves in the RR part.
The SDES block in the RTCP packet should describe ourselves,
not the sender.
This was fixed for the RR part in 952139a322, but wasn't
fixed for the SDES part until now.
This could cause some Axis cameras to send RTCP BYE packets
to us due to the SSRC collision.
Signed-off-by: Martin Storsjö <martin@martin.st>
This requires using a separate init function, since there
isn't necessarily any fmtp lines for this codec, so
parse_sdp_a_line won't be called. Incorporating it with the
alloc function wouldn't do either, since it is called before
the full rtpmap line is parsed (where the sample rate is
extracted).
Signed-off-by: Martin Storsjö <martin@martin.st>
RTCP timestamps are only necessary to synchronize time between
multiple streams. For a single stream, the RTP packet timestamp
provides more reliable timing. As a result, single-stream RTP
sessions should now have accurate and monotonic PTS.
Signed-off-by: Martin Storsjö <martin@martin.st>
Martin Storsjö
Michael Niedermayer
Anton Khirnov
Dmitry Samonenko
Samuel Pitoiset
Ronald S. Bultje
Piotr Bandurski
Justin Ruggles
Miroslav Slugeň
John Brooks