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
Richard Fuchs
Donat Zenichev
Victor Seva