Flag the media via stream params, when it's
trickle or non-trickle offer/answer, and,
unflag it always only when it's non-trickle.
Change-Id: Ia85ce5792684a4121e224aff1a8e941e061fe5a8
The OTHER type attributes are not used outside of sdp.c. Move the
definitions into the appropriate scope.
Rename fields to be consistent with other attributes.
No-op.
Change-Id: Id92f2df2f475db92ee5ae1f3474191266d6d196d
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
Use ngbuf's free function to destroy the JSON parser instead of the
callback for the bencode_buffer object.
Change-Id: I7eccf7284f55b34ef1a4800017ea1a4519f42bbc
Obsolete str_init(), rename STR_INIT() to just STR(), and replace all
instances of str_init() with STR().
no-op
Change-Id: I981529063ad2ea26089add467f7a84b638dbf423
Two version of the origin replace exist from now on:
- `origin-replace` - replaces only the origin address
- `origin-replace-full` - replaces all the values,
so name, id, version and IP family with address.
Values for replacing are taken from the rtpengine instance,
so local values provided by the daemon.
Additionally: documentation updated accordingly.
Additionally: revert changes in pub-sub test.
Change-Id: I4d068944726d1ab82683ca5aa641a954890aefcf
Instead of using separate char arrays storing
parts of the SDP origin (in monologue), just
use the corresponding structure.
Deprecate unused parts used before.
Additionally add logic:
don't set `->session_sdp_orig` for monologues
with empty tags. This leads to setting origin line
to those monologues, which will later skip updating it
with its own (so real one).
This is fixes the case with the offer/answer model,
where offer always sets its origin for the other side,
meanwhile the other side hasn't given the answer yet.
Then later this answer related monologue never gets
its real origin.
For this sake also adopt logic of `sdp_version_check()`
which serves 'SDP-version' and 'force-increment-sdp-ver'
flags.
Change-Id: I17f3ff67e1b3dafca797c5feb876ebb238dceaa2
Instead of using separate data members to
bring data with the `call_monologue` structure,
just use the whole `sdp_origin` object type (structure)
as a pointer and keep it aling with real SDP origin
of according monologue's side.
Refactor the code accordingly for `sdp_create()`
users (firstly only here).
Additionally introduce functions to alloc/de-alloc
`sdp_orig` object:
- `sdp_orig_dup()` returns a pointer to copied object
- `sdp_orig_free()` deallocates it
Change-Id: Iff6a777e4867e78c73ca79c73fdb73ff8e9f22eb
For the sake of simplicity in usage and also
visibility for rest of file implementers
via the types.h, turn `strct sdp_origin` into
the typedef `sdp_origin`.
Change-Id: I13e71b9bbc944cf2931afc4fbc2c3f465eea815c
Move the `call_subscribe_request_ng()` fully to a usage
of the `sdp_create()` only.
Carry the origin IP and net family via flags
to the monologue, so can be reused later when creating SDP.
Always use given SDP session origin IP address and family
for the SDP being prepared, unless sdp origin replacement
is required via given flags (in this case just used
an advertised IP of rtpengine).
Additionally: fix unit tests for subscribe cases accordingly
to the policy.
Change-Id: Ib7697876ce45e01597edd27764d4147d12f738c8
Update the bandwidth in the media object only on the side that has
received the SDP. Then when printing the SDP, look up the peer's media
via the subscriptions and use that one's bandwidth values.
Change-Id: I53c99b3628f53b2469f4cd73eb486c0110d989ba
Instead of carrying along an extra entry for the remote address, use the
existing advertised_endpoint, and look up the appropriate peer via media
subscriptions when printing the SDP.
Change-Id: I4c79053ba0fe072562ad71eb62ece3c527008936
Don't carry parameters required for processing
in the `sdp_create()` via the `stream_params`,
but rahter handle them like:
- parsing in `sdp_parse()`
- `sdp_media` -> `stream_params` in sdp_streams()
- `stream_params` -> `call_media` in ` __media_init_from_flags()`
Additionally: update the test "subscribe_request AMR asymmetric".
This is because we seem to never actually take into account
presence of bandwidth data in offer/answer model preceding
the subscribe request.
Change-Id: I5b4b19ae244c6bbf961d5ea7c18b6747519144db
Always ensure to set the actual sdp session timing
if given, when creating SDP using `sdp_create()`.
Otherwise just use default value "0 0".
A new dedicated func `sdp_out_add_timing()` introduced
for that.
Change-Id: Ic02e1a1f55e21b85e50793e1608978ca0951c49d
Always ensure to set the actual media session level
bandwidth, if given, when creating SDP using `sdp_create()`:
- AS
- RR
- RS
All of them can be presented simultaneously.
Adjust existing function `sdp_out_add_bandwidth()` to comply
with media session level demands.
Change-Id: I9599df051109ec05c4549ae79fae906fb5980dad
Always ensure to set the actual sdp session level
bandwidth, if given, when creating SDP using `sdp_create()`.
A new dedicated func `sdp_out_add_bandwidth()` introduced
for that.
Additionally: explicitely set b=RR/b=RS to -1 when creating
a brand new monologue (e.g. for subscriber requests case)
so it's not considered as 0 inadvertently.
Change-Id: I5bfa236ceeb326785feadadf7f22393814505d3f
For typed hash tables, enforce the correct type in the arguments to the
hashing and equality functions.
Adapt existing affected callback functions and change their arguments
from void* to the respective types.
Add reverse casts to GHashFunc and GEqualFunc in instances where these
functions are used in non-typed hash tables (that should be converted at
a later point).
Add convenience macro to create typed wrapper functions for hash tables
that use "direct" hashing (i.e. the pointer value).
Add wrappers for existing GLib functions that have generic arguments so
that they can be used in typed hash tables.
Change-Id: I43bb32969208f4aae49584d95c0df8353df6e2a0
In all the cases (apart PUBLISH) ensure to set the actual
media connection when creating SDP using `sdp_create()`.
A new dedicated func `sdp_out_add_media_connection()`
introduced for that.
Change-Id: I26e9b123aad95e2d335aef903d441ecc2cae2605
Donat Zenichev
Richard Fuchs
Peter Kelly