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
To be able to use this feature from controlling agents which don't
directly support the new dictionary keys, add an alternative syntax via
the `flags` list. This is in line with all other similar dictionary keys
(e.g. `codec-` etc)
Change-Id: I7210c74c9cd3b31338052efa1c3504fe775c1c46
These are part of the extended tests which are normally disabled. Some
change in libopus results in different encodings in these cases, making
these tests fail with newer libopus versions. Accept an alternative test
result.
Change-Id: I846830a47651bf7a7f83fa0bb1f1220db77db71e
Transcoding is flow-specific, so it doesn't make sense to have a flag
for it in a call_media section. Instead we use the transcoding flag set
on the call_subscription objects (on subscribers), and set/unset a flag
on the monologue struct, depending on whether any media flows (going to
subscribers) have transcoding enabled.
Change-Id: Id671d56e56a22eaa8e56f6d449770b0c7b086cea
We have the subscription objects available now, so we can directly set
the transcoding flag without having to do a hash table lookup.
Change-Id: I2b85f34ca4d03dfaf81d92ea252902d1ee194efd
Move the `ice_slow_timer()` functionality to a separate thread,
so that we do the work more efficiently, and not be dependent
on the call_timer runs by poller.
Furthermore it makes more sense to keep in the `ice.c`,
since it obviously has to do with ice timing.
Additionally:
Update the test-stats due to these changes in the `call_timer()`
We have to call the `ice_slow_timer()` now explicitely from
the test-stats.c, because the `call_timer()` is not anymore
responsible for providing stats counters rate calculations.
Change-Id: I03377dd59ea71c27497e1f4d30164075f05165cd
Similarly as for the `stats_rate_min_max()`,
move the `stats_counters_calc_rate()` functionality
to the same separate thread, so that we do the work
more efficiently and not be dependent on the call_timer
runs by poller.
Furthermore it makes more sense to keep in the `statistics.c`,
since it obviously has to do with statistics.
Additionally:
Update the test-stats due to these changes in the `call_timer()`
We have to call the `stats_counters_calc_rate()` now explicitely from
the test-stats.c, because the `call_timer()` is not anymore
responsible for providing stats counters rate calculations.
Change-Id: I1682eb76e3057f0f431c27b9633717d965313a1a
Support the "id" parameter for the "join" as "publisher" message.
Preempts choosing a random feed ID.
Add a test case for this.
Change-Id: Iae8c2f50864adf913b288085aa70d5427e0a4456
Simply stripping a supplemental codec doesn't actually affect the
remaining real codecs. Add a special-cased version of codec_touched() to
ignore such a manipulation.
Add a matching test case.
Change-Id: I4a91292dd38e1114837c2dc841afe07d87cff6cb
We have to call the `stats_rate_min_max()` now explicitely from
the test-stats.c, because the `call_timer()` is not anymore
responsible for providing call rate stats.
Change-Id: Id896ac086660a94b8d1d6fe520b1aa68791cd351
Moving code for handling (queuing, dequeuing) trickle ICE fragments into
ice.c, where it makes no sense. No functional changes.
Change-Id: Ib68f82e8d58efe066fdc48cd32ca9869cdeab846
We're also in controlling role for subscribe answers.
Make sure we don't clobber the source media's ICE options when adding a
subcriptions.
Change-Id: I5361462aefdbbe6411841332b69a8dc4b0e1e013
1) a=ice-options can be a session-level attribute. Use the correct
lookup function to check for both media-level and session-level
attributes.
2) The RTP endpoint address must be filled in before we do the trickle
ICE check, which uses 0.0.0.0:9 as determining factor.
3) Adapt a test case.
Change-Id: Ic0caffc85791131173848d28f5a652ad9d9124db
It substitutes a specified `a=` line taken from the concerned
media attributes list. If such line has been not found,
the attributes list remains untouched.
It subsitutes one attribute at a time, so one attribute into
another attribute.
Change-Id: Ie0a48ba46a1b196fbe33b09dedc40e4498640e34
These tests depends on an additional feature ("end event") that hasn't
been implemented yet. Disable these tests for now.
Change-Id: I54186cb5a1ed9119497fc05d7760d207a981cffa
Generate the output fmtp= string based on the preferences received from
the opposite side. Also add the required format printing function.
Change-Id: I12124efe0b9876c6571bc32c1c45744af80b83d3
This commit parses out the string, but doesn't do anything with the
values except for the FEC flag. Move the FEC on/off switch from the
extra codec options into the `fmtp` string.
Change-Id: I51f74f7cb62dd49a9af9815920f077bf300cfa33
Similar to the existing media_player, but instead of simply producing
its own standalone output media stream, the audio_player takes over the
entire media stream flowing to the receiver, including media forwarded
from the opposite side of the call, as well as media produced by the
media_player.
Change-Id: Ic34ecf08fc73b04210cfffb1a7d795b462ece5ea
... as an extra offset for newly added sources, based on the difference
between the last runtime (read time) of the buffer and the current time.
Change-Id: Ie99e24f0697f0950f0fcfa1e5e58b8f4be134018
A simple circular audio buffer that allows mixing multiple sources of
audio. Sources are tracked by SSRC and all sources are expected to
provide audio in the same format (same clock rate, channels, sample
format).
Only one consumer per buffer is supported, which is expected to retrieve
buffered audio at regular intervals (ptime) and so continuously empty
the buffer.
The first audio source to write into the buffer at the leading edge of
the circular buffer has its audio simply copied into the buffer, with
the leading edge advanced, while other later sources writing into the
buffer mixed into the existing buffered audio at their respective write
positions.
Change-Id: I0f6642b036944508f2a33420359de488ef8b991c
The test number 1142:
"SDP attr manipulations - add a= line for media audio, two times"
used to fail on the buster builds, because of the hash tables
implementation we had in the `command_values` of the `struct sdp_command`.
It didn't necessarily keep the same order, as was used to add values.
Now as the SDP manipulations structures is re-designed, we can re-add it.
Change-Id: Icc7d030894d45c9d62e91669915eafb348d05d97
The test number 1142:
"SDP attr manipulations - add a= line for media audio, two times"
fails on the buster builds, because of the hash tables implementation
used in the `command_values` of the `struct sdp_command`.
It doesn't necessarily keep the same order, as was used to add values.
Temporarily removing it, until the `struct sdp_command` is re-designed.
Change-Id: I077771c00c5dc8907e42d7757540e3ffa2063af6
Don't change to a new port for sendonly streams as this causes problems
with NAT. A device receiving a sendonly SDP with a new port won't send
any RTP to the new port, leading to a closed (non existent) NAT mapping.
Change-Id: I2ea2163eb9f1203226bd781b53f421c790a86f0a
New section of option flags has been introduced for SDP body
attributes manipulations.
Three levels of the SDP session are concerned:
- session level (global one)
- media level - audio
- media level - video
Three different actions are supported for now:
- add
- remove
The value of the command has a wildcard matching approach.
Other attributes apart `a=` can not be edited by this functionality.
So such headers as: `c=`, `s=`, `o=` cannot be touched.
Change-Id: I939d4582839096b2399f7ded865e91ff6eb960a4
(cherry picked from commit 3f06c18793fe95e5b070044a0291a3e1528ac6e4)
Only parse out a codec type if the codec is given with any parameters
set (and not just by name). Parsing out a codec type when just its name
is given leads to a codec type with default parameters set (such as the
number of channels) which can lead to a stricter matching than desired.
When a codec is given just by name we want to match all codecs of that
type, not just the ones with the same parameters as the default ones.
Change-Id: I583bf4045dbd55291d8dc596310730024853d386
There's no point in sampling Graphite-specific stats, nor in reporting
them out, if Graphite is not enabled.
Change-Id: If8014513832485f38d81b478c695391129c21dff
Distinguish between two different types of "gauge" type metrics: Actual
gauges which (at least conceptually) have a single continuous value, and
metrics which are comprised of discretely sampled values, possibly from
multiple sources.
Real gauges with continuous values don't have mean/average/deviation
values directly associated with them, as calculating these requires
sampling or some other analysis.
Sampled metrics on the other hand do have these associated values.
Clarify which function does what and where each value comes from.
Change-Id: Iff5dd844b70ff70979b1b8c84dc7734d44b3da20
Rename structs and variables to make it clear that these min/max values
are min/max per-sec rate values.
Carry mins and maxes separately from averages. This changes the meaning
of $command_ps_avg away from an "average of averages" to an actual
average, which is more accurate.
Calculate this average based on per-interval differences and interval
duration (stats_rate_min_max_avg_sample).
Side effect: As rtpe_latest_graphite_interval_start is now set in
print_graphite_data instead of in graphite_loop_run, the test now
reports a different "interval calls duration".
Change-Id: I67b1118c18ca2464a48c4836fca3cfdb4d53c898
Perform accumulation of stats only once (i.e. increasing an actual
counter) and report stats based on differences to previous values,
instead of carrying multiple stats counters for each metric and
resetting each counter to zero whenever stats are reported.
`rtpe_stats` is the global master accumulator.
`_intv` variables are intermediate and local storage for values sampled
from `rtpe_stats` at regular intervals.
`_rate` and `_diff` variables hold stats calculated from `rtpe_stats`
and the respective `_intv` variable whenever the sampling and reporting
occurs.
`stats_counters_calc_diff` is used to calculate stats as differences
between `rtpe_stats` and the last sampled `_intv`
`stats_counters_calc_rate` does the same but calculates a per-second
rate, based on a microsecond duration.
Eliminate now-useless struct global_stats_ax
Change-Id: Ic4ca630161787025219b67e49b41995204d60573
This fixes the payload counts not being tracked correctly when payload
types repeat after the tracker rolls over.
Change-Id: I16208ef73f3af3b051b96541a4c145b323cef7b2
According to:
{
"request" : "join",
"ptype" : "subscriber",
"room" : <unique ID of the room to subscribe in>,
"use_msid" : <whether subscriptions should include an msid that references the publisher; false by default>,
"autoupdate" : <whether a new SDP offer is sent automatically when a subscribed publisher leaves; true by default>,
"private_id" : <unique ID of the publisher that originated this request; optional, unless mandated by the room configuration>,
"streams" : [
{
"feed" : <unique ID of publisher owning the stream to subscribe to>,
"mid" : "<unique mid of the publisher stream to subscribe to; optional>"
"crossrefid" : "<id to map this subscription with entries in streams list; optional>"
// Optionally, simulcast or SVC targets (defaults if missing)
},
// Other streams to subscribe to
]
}
{
"videoroom" : "attached",
"room" : <room ID>,
"streams" : [
{
"mindex" : <unique m-index of this stream>,
"mid" : "<unique mid of this stream>",
"type" : "<type of this stream's media (audio|video|data)>",
"feed_id" : <unique ID of the publisher originating this stream>,
"feed_mid" : "<unique mid of this publisher's stream>",
"feed_display" : "<display name of this publisher, if any>",
"send" : <true|false; whether we configured the stream to relay media>,
"ready" : <true|false; whether this stream is ready to start sending media (will be false at the beginning)>
},
// Other streams in the subscription, if any
]
}
Change-Id: Ieb38d4f562686283457a963334056b27927be974
This is a new option flag, which provides a possiblity
to select specific crypto suite(s) for the offerer from
the given list of crypto suites received in the offer.
This will be used later on, when processing an answer from
the recipient and generating an answer to be sent out towards offerer.
Furthermore, this is being decided not when the answer is processed,
but already when the offer is processed.
Flag usage example:
`SDES-offerer_pref:AES_256_CM_HMAC_SHA;AES_256_CM_HMAC_SHA1_32;`
Change-Id: I2b22b38347d24f27331482e18b92305fbadb2520
Older versions of glib handle the trailing padding in base64
differently, resulting in the final character of the encoded string
coming out different.
Change-Id: I2bc1057de15f5e0bddb3de4df6cef7c4fcf7ebcc
We have to cover more cases with tests, such as:
- SDES re-ordered crypto suites, but one suite from the 'SDES-order:'
flag was not in the offer
- SDES re-ordered crypto suites, but one suite from the 'SDES-order:'
flag was not in the offer and the recipient selected it
Change-Id: I1051c01d45c4b494f768692f85d1e0c41a0ea2d2
This is a new option flag, which provides the ordered list,
in which to add crypto suites into the SDP body.
Right now they're always added in the order given in the source code.
Flag usage example:
`SDES-order:AES_256_CM_HMAC_SHA;AES_256_CM_HMAC_SHA1_32;AES_192_CM_HMAC_SHA1_80;`
This means — those listed SDES crypto suites will be added
into the generated SDP body at the top of crypto suites list, in the given order.
But, each of them is added, only if it is about to be added/generated.
In other words, the `SDES-order:` flag itself doesn't add crypto suites,
it just affects the order of those suites to be added.
And the rest of non-mentioned suites, which are also to be added,
will be appended after those given, in the free manner of ordering.
Important thing to remember - it doesn't change the crypto suite tag
for the recipient, even though changing the order of them.
Additionally.
This flag does not contradict with `SDES-nonew`, `SDES-only-` and `SDES-no-` flags.
It just orders the list of crypto suites already prepared to be sent out.
Change-Id: I0fec54f9e2f3cd4913e905e8afe825712f82d1ae
This type is used as const everywhere except internally, so make it part
of the typedef for brevity.
Change-Id: Ic4afe037b392239a991d5380c6708903011da29e
Add a new flag to only accept these individual crypto suites
and none of the others.
For example, `SDES-only-NULL_HMAC_SHA1_32`
would only accept the crypto suite `NULL_HMAC_SHA1_32` for
the offer being generated.
This also takes precedence over the `SDES-no-` flag(s),
if used together, so the `SDES-no` will be not taken into account.
This has two effects:
- if a given crypto suite was present in a received offer,
it will be kept, so will be present in the outgoing offer; and
- if a given crypto suite was not present in the received offer,
it will be added to it. The rest, which is not mentioned,
will be dropped/not added.
Flag name: 'SDES-only-<crypto name>'
Additionally: add another new flag 'SDES-nonew'.
It will not add any new crypto suites into the offer.
It takes precedence over the `SDES-no` and `SDES-only` flags,
if used in combination.
Change-Id: Ic4fa03957ee3d4d24b0c4f3fd003eada05f49b0b
Support fake bind() on non-INET sockets. This fixes the tests when
building against libwebsockets19 4.3.2-1.
Change-Id: I91e5271e8b0cd5a0fa10317bae059615b234926c
Instead of going through ffmpeg to en/decode Opus, use libopus directly,
which allows us to benefit from additional features that aren't
available when going through ffmpeg.
Change-Id: I017c276cfa9755cefe95c8da26691446b718d4c8
Some codecs (e.g. Opus) can natively encode audio with various clock
rates without producing an output that is locked to that clock rate and
without requiring resampling the input. Add an appropriate callback
function and adapt tests.
Change-Id: Id788c4d4c05e20f93cce7e910f9f265b381cbe34
Add a flag to force increasing the SDP version,
even if the SDP hasn't been changed.
And cover it with tests.
Flag name: 'force-increment-sdp-ver'
Additionally fix the name of the 'sdp-version' flag
in the 'rtpengine-ng-client' tool.
Change-Id: I466792668b0cd313b5e21b248dd14cd599333cbd
Reset the iteration counter for the frequency list when codec handlers
are set up, which happens while setting a new block mode. This makes
sure that a newly set list of frequencies is iterated starting from the
top.
Change-Id: I7ab4e28a1a998c11e38e26cf5b1f17f01299d5ad
Support multiple tone frequencies for DTMF-security=tone to enable
audibly distinguishing multiple consecutive DTMF events from one
another.
Change-Id: I6fa33a5768aae198220d0b0cc4c53308c5661a52
Usually supplemental RTP types (DTMF) are listed after the primary audio
codecs. In the case of the order being reversed, fix `single-codec` so
that it doesn't strip the actual audio codec that is listed after the
DTMF type.
Change-Id: I1b03b89e31bebf4de303b643dcf08d2ffb90ebaf
related to #1497#1549
> ======================================================================
> ERROR: setUpClass (__main__.TestWSJanus)
> ----------------------------------------------------------------------
> Traceback (most recent call last):
> File "/code/t/auto-daemon-tests-websocket.py", line 114, in setUpClass
> eventloop.run_until_complete(get_ws(cls, "janus-protocol"))
> File "/usr/lib/python3.10/asyncio/base_events.py", line 646, in run_until_complete
> return future.result()
> File "/code/t/auto-daemon-tests-websocket.py", line 23, in get_ws
> cls._ws = await connect(
> File "/usr/lib/python3/dist-packages/websockets/legacy/client.py", line 622, in __await_impl__
> transport, protocol = await self._create_connection()
> File "/usr/lib/python3.10/asyncio/base_events.py", line 1089, in create_connection
> transport, protocol = await self._create_connection_transport(
> File "/usr/lib/python3.10/asyncio/base_events.py", line 1107, in _create_connection_transport
> protocol = protocol_factory()
> File "/usr/lib/python3/dist-packages/websockets/legacy/client.py", line 160, in __init__
> super().__init__(**kwargs)
> File "/usr/lib/python3/dist-packages/websockets/legacy/protocol.py", line 154, in __init__
> self._drain_lock = asyncio.Lock(
> File "/usr/lib/python3.10/asyncio/locks.py", line 78, in __init__
> super().__init__(loop=loop)
> File "/usr/lib/python3.10/asyncio/mixins.py", line 17, in __init__
> raise TypeError(
> TypeError: As of 3.10, the *loop* parameter was removed from Lock() since it is no longer necessary
>
> ----------------------------------------------------------------------
Change-Id: I3178c54ed7eb40b9cc06769c1f1e237e0d58f966
Keep a running lifetime total of all "gauge" type metrics. Also track
the square of the sums of all "gauge" type metrics in order to determine
the standard deviation.
Change-Id: I23f60774a6421636f1a913674c7d1b54a1c5f702
To safeguard against non-refcounted objects being left over in a log
info piece (e.g. a string on the stack), add this new function to pop
pieces from the stack until the desired one is removed. This is needed
in case of a unpaired log_info_* without a matching log_info_pop.
closes#1511
Change-Id: I689de14d034df779521dfdf59f923fdbf7fabc9b
The order between receiving the STUN success and the triggered check is
not guaranteed, therefore resolve possible race condition by expecting
the two packets in either order.
Change-Id: Ibef9907cd4116bc5f3b7d17d936007c8efcabd3b
The codec answer routine resets the codec storage and so also resets the
clock rate tracker for "touched" codecs. This leads to all codecs seen
as "not touched" in the answer routine, which in turn leads to
supplemental codecs present in the answer SDP that should not be there.
Use the "for transcoding" flag for previously present codecs to retain
the "touched" status across the codec answer routine.
Change-Id: Idc4624606f7f10d7983e22ddf856432b07421157
* Use an explicitly created global event loop
* Await websocket connection closure
Relevant to #1497
Change-Id: I600189f5383ca7e5da8b45460508c1ddcddede0b
When doing the initial answer, the packet_stream endpoint port isn't
filled in yet. Use the stream_params port instead to test for rejected
streams.
closes#1499
Change-Id: I8f315d95521f874fb8c5e6222263d017800b5fc9
This eliminates a spurious false warning log message for rejected
streams that use a dummy payload type
Change-Id: Id628cafb8d7c4ea576cd01ff35f5dd9cd2151280
Richard Fuchs
Donat Zenichev
Orgad Shaneh
Michael Prokop
Victor Seva