Use a BIO WRITE callback instead of BIO_read'ing from the BIO after each
operation. This is a more direct way to intercept data that needs to be
sent out.
Implement MTU-related BIO callbacks.
Deduct the assumed IP MTU overhead from the configured MTU during
startup.
Unlike the previous code, this does not necessarily send DTLS from the
same socket that received a message, nor to the same address that sent
one, and instead always uses the selected_sfd and ->endpoint. This may
or may not be a regression.
Closes#1806
Change-Id: I4d4456df3f378d00782cbfa64afdb2a038217e6c
Move the SDP offer/answer model to the sdp_create
approach instead of using the sdp_replace one.
This assumes the SDP body including session level
attributes (s=, o=, t= etc.), as well as the media
attributes, are formatted using currently given
session context.
In other words, rtpengine collects all possible
information during all of the offer/answer exchanges
within the dialog, which in its turn affects each
monologue's context, from which each new SDP message
will be built up.
This approach replaces the older one, which instead
used to go through the currenty processed SDP and
just replace those attributes, that rtpengine
is required to affect, leaving the rest untouched.
Additionally: all of the existing offer/asnwer model
tests were fixed to comply with the currect change.
Most of it is just an offset of attributes, which are
stored a bit higher/lower within the same media or global
SDP session.
Also a good part of unit test fixes targets a move of
`c=` line from the session level to the media one (so
each media has from now on its own `c=` attribute). This
does discontinue a support for session level connection
information.
Change-Id: Iecb4739683d23c4f9341e8a34b71f8ca2070956c
Add `call_ng_process_flags()` based parsing
as for other opmodes, like offer and answer.
This keeps the backwards compatibility with
the older "flags" parsing approach on the
module side, as well as adds the possibility
to parse rtpp-flags on the daemon side.
As an advantage, there is no need to use
specific local parsing for things like
to/from tags, call-id, delete-delay etc.
Additionally:
- this commit introduces flags-flags parsing
for the "fatal" flag.
However, as before is only taken into account
by the `call_delete_ng()` processing,
so no functional change.
- this commit introduces main-flags parsing
for the "delete-delay" flag, which is also
only taken into account by the `call_delete_ng()`
processing, so no functional change.
- this commit adds To-tag options flag prasing
into the `call_ng_flags_flags()` function,
and is used by `call_delete_ng()` specificially,
for cases when more specific identification of
monologues to be deleted is used.
Change-Id: Ia992e5375a2f86318d9ad193a7857dd589038eed
For trickle ICE updates that need to be queued up, this requires storing
the unparsed SDP in the fragment object, and then doing the parsing when
actually processing the fragment.
This allows the call's memory arena to be used for parsing.
Change-Id: I28ed192c4443cedfa3095007cc8a555e3aa7a17a
Instead of having to explicitly pass the call object to each invocation,
keep one thread-local reference to a call, implicitly set by setting the
logging context.
Add helper functions to set and release the respective reference.
Change-Id: Ic0d82eeaa403467d50dae867e33fdf9b9dd7cec5
Support session level group attribute
for the sdp_create approach.
Additionally: fix the `testVideoroomWebRTCVideo` test,
because as it turned out it used to lose the `a=group:`
attribute before.
Change-Id: I799dfe0dcc5e0708864150cd0e0262a61dd4cc05
Added support of the CT (conference total) bandwidth
attribute for the SDP session level. See RFC8866.
Will be required later for the SDP formatting in `sdp_create()`.
Change-Id: Ifc64f68d7acee8ce253882f4fa480bbf7ad7c0bd
... so that the desired wav channel can be controlled when producing a
mixed audio file
When a mixed wav file is created, the channels in the wav container are
currently allocated in the same order as each SSRC is received, meaning
it is impossible to know which channels have been allocated to the offer
or answer side of the call. Furthermore if there is a reinvite or media
file played, these are also allocated in the order that SSRC is received
- so an "answer" could end up sharing a channel with an "offer" with no
way of knowing this.
This patch allows you to specify how many channel slots should be
allocated within the mixer, and allows you to then specify which slot is
assigned to each media in the call (this will usually be 2 slots in
total, slot 1 for answer, slot 2 for offer or vice versa).
Ported from https://github.com/sipwise/rtpengine/pull/1852Closes#1857Closes#1852
Change-Id: I010208427cabc3a48d6ef7bd3a84e9a5bdcfd492
Added support of the AS bandwidth attribute for the
SDP session level. Will be required later for the SDP
formatting in `sdp_create()`.
Change-Id: I1bde4659679de6e60bdad12c0578ced2c1983300
Same as ptime but for the maxptime.
Also add replication of it.
This is required later to be used for the sdp_create() handling.
Additionally: fix tests, because maxptime now takes another
place within the media session, which doesn't affect functionality.
Change-Id: I058e35323849679976c60b2e9fb2555fd0168e67
Richard Fuchs
Donat Zenichev
Peter Kelly