For libsrtp 1, add defines that translate the new namespaced identifiers
to the old unnamespaced ones. Also move the code for setting and getting
a stream's ROC into two compat functions that match libsrtp2's API.
It seems that libsrtp2 properly supports changing the ROC without having
to touch the sequence numbers afterwards, given that srtp_set_stream_roc
sets a pending_roc field, so the entire roc_changed dance should not be
needed anymore. The compat functions for libsrtp 1 just contain our
preexisting hacks, however, so it's still needed there.
libsrtp2 has no means of discovering the streams in the session, so to
create the stats structure we need to iterate over our own set of SSRCs.
For this we also need to re-add the previously removed ssrcs_set to the
encoder.
https://bugzilla.gnome.org/show_bug.cgi?id=776901
Currently, gst_srtp_dec_sink_setcaps is happy if the "roc" field is not
provided in the caps. If it is not provided the stream will be properly
inserted in the hash table with a default "roc". Then, when the first
buffer arrives validate_buffer will find an existing stream in the hash
table and will not signal request-key, not allowing the user to provide
a "roc".
This patch expects "roc" in gst_srtp_dec_sink_setcaps, if not found a
request-key will be signaled and the user will be able to provide all
the srtp fields, including "roc".
https://bugzilla.gnome.org/show_bug.cgi?id=765079
This reverts commit ff11a1a8a0.
It can't be assumed that all buffers in a buffer list have the same SSRC or
are RTP or RTCP only. It has to be checked for every single buffer, and one
basically has to do the processing that is done by the default chain_list
implementation.
Upstream might not give us a caps event (dtlssrtpdec) because it might be an
RTP/RTCP mixed stream, but we split the two streams anyway and should report
proper caps downstream if possible.
Fixes "sticky event misordering" warnings with dtlssrtpdec.
We add a new signal, get-rollover-counter, to the SRTP encoder. Given a
ssrc the signal will return the currently internal SRTP rollover counter
for the given stream.
For the SRTP decoder we have a new SRTP caps parameter "roc" that needs
to be set when a new SRTP stream is created for a given SSRC.
https://bugzilla.gnome.org/show_bug.cgi?id=726861
* ext/srtp/gstsrtp.[ch]: added GST_SRTP_CIPHER_AES_256_ICM to
GstSrtpCipherType and new function cipher_key_size.
* ext/srtp/gstsrtpenc.c: maximum key size is now 46 characters (14 for
the salt plus the key). If different ciphers are chosen for RTP and
RTCP the maximum needed key size is expected.
* ext/srtp/gstsrtpdec.c: minor documentation updates.
https://bugzilla.gnome.org/show_bug.cgi?id=720434
