Make the `sdp_replace()` compliant with `sdp_create()` in terms
of the replace `origin-full` flag support.
Also introduce according unit tests.
Change-Id: Ib950a957d5ee3820dbc8a831fe0a7ea570b4c59b
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 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
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
Run the SDES reordering routine also when the opposite side is plain
RTP. Add a matching test case.
Ref: https://groups.google.com/g/rtpengine/c/pG6NB24G-BY
Change-Id: I7844290e85e44c1bba2c7665202b6259aead1028
Split the code that handles reordering offerer SDES suites based on the
respective option into its own function. Rework it slightly so that it
reorders the list in place for simplicity. Remove the part that honours
SDES-no and SDES-only as these should affect the opposite (outgoing)
direction. This requires changing one of the test cases, which seems
more correct now.
Change-Id: Ie284d052d72031fad64c94767fa95c74639ae331
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
To prepare the deprecation of session level connection
usage (so the `c=` attribute of the session level),
deprecate first the `replace-session-connection` flag
support, because it will have essentially no use.
Change-Id: I9f9cb8f06d03d1c107a5c928054a79409642bb5b
Rework the `sdp_out_add_origin()` func so, that
it's compatible with offer/answer model.
Additionally it uses the previously existing logic
for the PUBLISH operations always, regardless any
conditions.
Additionally: fix auto-daemon-tests-websocket accordingly.
Change-Id: I0bddc67f4ebe47a24063ccc82428929aeb6cc37a
Always ensure to set the actual sdp session name when
creating SDP using `sdp_create()`.
A new dedicated func `sdp_out_add_session_name()` introduced
for that.
Additionally: fix auto-daemon-tests-websocket and
auto-daemon-tests-pubsub accordingly.
Change-Id: Ie62573149ef9ae226dc8c955dcb2cfaaa4d3ae87
Always ensure to set the origin name properly when
creating SDP using `sdp_create()`.
A new dedicated func `sdp_out_add_oirigin()` introduced
for that.
Additionally: fix auto-daemon-tests-websocket accordingly.
Change-Id: I671f7b54f8bed9c4b78d6532c0072836d77173e6
There isn't any immediate benefit to this, but it prepares the code for
use of shared memory for statistics.
Use the opportunity to switch accesses to these to relaxed memory order.
Change-Id: I585fef7579202179fbbcbc1b843d3bbe440a723b
Switch all memory buffers used for RTP I/O from generic stack or heap
allocated memory to the bufferpool implementation. Use a per-thread
bufferpool to minimise lock contention.
This commit is just a one-for-one swap and doesn't use the bufferpool's
reference counting semantics yet.
Change-Id: I9cba4ec97bd0afcd374bf6c0be2b608a46e73e57
now that the returned last_event_ts is always that of the previous
DTMF, we can ensure that the next one isn't transmitted until that
time plus the required pause.
Like the num_samples calculation, the actual time needs to be
increased by 1 packets worth of samples so tha pause lasts the
full duration required
Change-Id: I6da1dd7cbcf49f7f0431a5123df2cdc382fe3dba
this function is used to determine if a pause is needed on a new
injected DTMF's start ts to ensure a gap between the events. However,
if an inject request comes in after the end of the previous event
but before it would have been offset due to pause, no pause is added
This change returns the ts value from dtmf_state if the queue is
empty as that will always be the ts of the last DTMF transmitted
Change-Id: I4f3cf5115d1a8e26c0ca1bc9570c46e29391e0d0
the num_samples was added to the start_pts, which is the first event
packet timestamp, which has already increased its ts by its event
duration. so, the total duration of events ends up being one packet
more than intended.
Change-Id: I423bb222a81c5bd78e570ff2026c72dd4dd1b100
Make sure we increase the output RTP sequence
number for each generated packet in the case
of packets that need to be duplicated or sent
repeatedly (DTMF end event)
Change-Id: Ia16ffefc0791d01575248ac5d8025eb30ccaec67
For the period of time while we are using
a new approach for the parsing of option flags
on the daemon side in parallel with the parsing
on the kamailio module side, and,
the previous approach with the module isn't
deprecated yet, it's good to have some basic tests
covering a new way of prasing to ensure it's working fine.
Change-Id: I8b28310b9973878530688780b6fcf366d239629d
Add support of rtpp_flags parsing for the daemon.
From now on, it's possible to parse option flags
on the daemon side instead of the kamailio module.
It's identical to what the module does, but the
difference is:
- module sends general call identification such as:
call-id, From/To tags, viabranch using bencode
- meanwhile all generic/non-generic option flags
are added to the `rtpp_flags` bencode member as str
- parsing of that is done as usually using the
`call_ng_main_flags()` / `call_ng_flags_flags()`
and additionally using new parser `parse_rtpp_flags()`
New file implementation and header introduced:
- control_ng_flags_parser.c
- control_ng_flags_parser.h
Otherwise no functional changes, and the parsing itself
remains working following the same algorithm.
Change-Id: I59e47fa1947e2aeaa0bbf3930a0f21d9a6d669ad
If multiple pollers are in use, use a single poller per call instead of
assigning pollers round-robin to each socket used in the call.
Change-Id: Iec49bd9d2fbd75d947d6232bcccfdfe87c4c6d7c
The poller-per-thread feature was broken with a division by zero. Take
the opportunity to rework it and eliminate the poller_map object. Use a
simple array of pollers for media sockets, plus one global poller for
control sockets. In the regular case only one poller is created and
everything points to that poller. In the poller-per-thread case, one
poller per thread is created, plus one poller (also with its own single
thread) for control connections. All control sockets use the single
control poller, while all media sockets get assigned one poller from the
pool in a round-robin fashion.
closes#1801
Change-Id: Iae91a3e10b7206455c6df33b1a472254c700ce21
For convenience we provide extra HTTP and WS endpoints that accept a
cookie-less NG or JSON message string. Not all commands are sensitive to
retransmits and this makes it easier to query call status etc.
Change-Id: Iffbc4ef9a5fdf916a374dfdd4042c61b437d18c9
Consider call_media SDP attributes as ones that were received instead of
the ones to be sent out. Use media subscriptions to look up the source
media and print that one's attributes when creating an outgoing SDP.
Change-Id: Ibdf3a77a6f8a61654e0fc7c14aae16dfc6eabf14
Fix up tests to match reordered attributes.
Remove 10-year-old a=ssrc parsing code completely as it's unused.
Change-Id: I78064b1b2f27a442fe8bf4b448c7174c5458d3b1
Some of these are actually wrong at this point, and will be fixed up in
later commits when the outputs are correct
Change-Id: I12d264a73f44e5356912b34bd22d7075067df190
Graphite isn't able to deal with spaces in the metric names delivered to
it. Introduce special version of the command strings with spaces
replaced by underscores.
closes#1780
Change-Id: Ie8bcec5ca4f2d427e92901f6fa76b985df6e459e
Instead of using transcoding flag `_TRANSCODING`
on the monologue level, we have to use that on the media
level in order to properly reflect the level on which
transocding is being used, and also to be able to selectively
set this for specific media sessions.
Change-Id: I9a25dc7be24f80b2b6ada816448a67933c762d86
During an offer, we update the codecs from the given list not only on
the side of the offerer, but also on the answerer's side, in order to
perform the codec answer routine during the answer phase. While doing
this, we empty out the existing list of codecs (on both sides) and
repopulate it fresh from the given list.
This can cause problems during a reverse re-invite, when the list of
codecs on the answerer's side already contained the codecs that had been
offered before. When setting up the new re-invite offer, we want to
retain codecs (and their payload types and format parameters) that were
already in place, instead of recreating a new list from scratch.
Improve this by adding a `merge_cs` option to the populating functions,
which points back to the stream_params codec_store. Codecs that would
have been removed from the codec_store during the repopulation are then
moved back into the stream_params codec_store instead. This then allows
the functions adding new codecs to the list (offer/transcode) to
reference these codecs that were previously in place, and so they can be
added back with the same options as they had existed before, instead of
recreating them from scratch.
Change-Id: I53e7ab10e9144a308a5c36be5ebfddd73c212f06
Add handling of SDP session level attributes for:
- `monologue_subscribe_request()`
- `monologue_subscribe_answer()`
This will be used by `janus.c` related functionality.
Change-Id: I1c50b5b9da08e7d8cb2c98eb6995d8610386d6ed
Move `a=extmap:` attribute to the `call_media`'s
`sdp_attributes` using `stream_params`.
Required by the `insert_sdp_attributes()` used in `sdp_create()`.
Additionally revert `extmap` unit test checks in the
`testVideoroomWebRTCVideo`.
Change-Id: If63e4e8733ea0899f34fae1f1d38997e9e2c081c
Move `a=ssrc-group:` attribute to the `call_media`'s
`sdp_attributes` using `stream_params`.
Required by the `insert_sdp_attributes()` used in `sdp_create()`.
Change-Id: I67025dc83dd8e5e46422b6dc1bc54aa9c838dfea
We have to move the list of attributes with following types
to the `call_media`'s `sdp_attributes` using `stream_params`:
ATTR_SSRC
ATTR_MSID
ATTR_OTHER
This is then required by the `insert_sdp_attributes()`
used by:
- `sdp_replace()` -> `replace_sdp_media_section()`
- `sdp_create()`
Additionally: move the `insert_sdp_attributes()`
to the usage of the `append_attr_to_gstring()`, which
gives SDP attrs manipulations control over attributes
being inserted into SDP. Previously not supported.
Additionally: `process_media_attributes()` takes new
argument `bool strip_attr_other`, in order to control
whether or not to strip `ATTR_OTHER/ATTR_SSRC/ATTR_MSID`.
For now there is only one user of it `replace_sdp_media_section()`,
but for later on, we can control this using this bool.
Additionally: `print_sdp_media_section()` takes new
argument `bool print_other_attrs`, in order to control
whether or not to strip `ATTR_OTHER/ATTR_SSRC/ATTR_MSID`.
This is required, because the sdp replacing/creating mechanism
needs a controllable way of adding them, users are:
- `janus_videoroom_join()`
`janus_videoroom_configure()` - always need them
- `call_publish_ng` - always needs them
- `call_subscribe_request_ng` - never needs them
(otherwise leads to a duplication of such attributes)
- `call_offer_answer_ng()` - requires them always
- `call_subscribe_request_ng()` - never needs them
(otherwise leads to a duplication of such attributes)
Change-Id: I9c83d99da1603acb55443c462797b2cd1e72477d
Stop using the call subscriptions model in the
subscribe request/answer functionality and move to
the media subscriptions.
Change-Id: I8aab2b1b4cdf9a3c5a04172c395ca509295ce0a3
The global list of monologues is populated in reverse order from the
list of medias. Iterate the list backwards so that the resulting medias
are in ascending order.
Change-Id: Icadca7a6cecb636ca6e021c0286b44e5cd3cdc84
With the "foreign" flag now stored in call_flags, restoring a call from
Redis restores this flag as well, overwriting the desired flag as we had
set it through call_get_or_create().
Reverse the flag setting by taking it out of call_get_or_create() (where
it's always false anyway, except when coming from Redis) and setting it
explicitly with call_make_own_foreign() after restorting call and its
flags.
Change-Id: Ib68be2aeedfa988b7555e426fa337657e1062245
Start using the media subscriptions model
(based on newly introduced `media_subscription` objects)
in scope of:
- `redis_encode_json()`
- `json_link_medias()`
- `rbl_subs_cb()`
Change-Id: I3f7267ab156b361d7e7bec4ff91a8976a7be02ee
Janus reports inactive and disabled streams only as being disabled and
doesn't report the codec in use. Mirror this behaviour.
Change-Id: Icf193f60a054b803afea02f048dcd8d26632bc47
Convert each listener entity into a list.
Support a list of values for each option so that multiple
ports/addresses can be listed.
Keep previous behaviour unchanged: If ANY address is given, open
listeners for IPv4 and IPv6.
Change-Id: Ic54f28d1262f60d5e5c9d824a95e7c33ebc2aba9
Previous implementation assumes that we use the `call_subscription`
objects in:
- `call_offer_answer_ng()`
- `call_update_lookup_udp()`
- `call_request_lookup_tcp()`
when appealing to the `call_get_mono_dialogue()`, in order to
get the `call_subscription` objects, in order to then pass it
for usage in the `monologue_offer_answer()`, where the most important
again is to use monologue references stored inside
the given `call_subscription` objects.
Instead of using the `call_subscription`, just use `call_monologue`
objects as a base data objects for this work,
which will allow us in the coming commits to deprecate
the `call_subscriptions` based model and
get to the subscription model based on medias.
Change-Id: Ia9ee5ba66522929acbceca28854ebccd3705635a
These had unintentionally been disabled and so have fallen derelict.
Replace the makefile `with_amr_tests` conditional with
`RTPENGINE_EXTENDED_TESTS`. The previous `with_amr_tests` was not
actually set anywhere and so these tests never ran. The new
`RTPENGINE_EXTENDED_TESTS` is used as switch for other tests already.
Bring the tests up to date so that they compile and run.
Allow for alternative AMR bit encodings as different versions of the
codec produce different outputs.
Use an inline `struct cb_args` to pass these expected strings to the
callback functions.
Update `int` to `size_t` where appropriate.
Make a copy of the format string in `__sdp_pt_fmt` as the parsing
function temporarily writes into the string, which means a readonly
string literal doesn't work.
Update the AMR tests in test-transcode to the new API and new expected
values.
Change-Id: Ic3332e696522a28595ea06be9996001d3ab7686e
Fix for:
make: *** No rule to make target '../daemon/poller.c', needed by 'codec.c'. Stop.
make: *** Waiting for unfinished jobs....
Change-Id: I94bb758cc28cbaf2bec320213dd40a4fd075f12c
Donat Zenichev
Richard Fuchs
Tom Briden
Steven Ayre