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
Since we're creating a dummy sfd to hold the SRTCP context when we don't
have an actual RTCP port, we must make sure to remember and re-use this
dummy sfd during a re-invite. Otherwise we end up creating a duplicate
dummy sfd, which is detected as a different sfd and thus triggers an ICE
restart.
Change-Id: Iadc91e163bd15a3cd5f57656b52941724c920143
Explicitly copy SDP up to the format list before printing it out. This
preserves broken input SDP.
closes#1461
Change-Id: I839a200f159f25854c86add244571a948e2c90cf
Special handling for codec lists that were received as part of an
answer: If the list includes a codec that was not offered, ignore that
codec. This prevents transcoders from being set up that were not
requested.
This brought to light some tests that were actually broken.
Change-Id: Iac71056ec5e10b5de5567917974f2c4e0261eb0c
This is useful for functions which are used both from a timer and from
other callers. These functions would reset the logging context at their
end to free the reference held by the logging context, which would
wrongly reset the logging context when the same function was called from
a different code path. Using a stack with push/pop semantics makes it
safe to use these functions from any code path.
Additionally introduce an explicit reset function that clears the entire
stack regardless of context. This reset function is called at the end of
every work iteration in every worker thread, just in case not everything
was popped from the stack.
Change-Id: I0e2c142b95806b26473c65a882737e39d161d24d
commit a2e5cfb8e5
Author: Razvan Crainea <razvan@opensips.org>
Date: Thu Jan 13 16:16:19 2022 +0200
Add tests for subscribe requests on paused media
commit fa58596a9f
Author: Razvan Crainea <razvan@opensips.org>
Date: Wed Jan 12 22:01:27 2022 +0200
Swap media direction check for `subscribe request`
as @rfuchs mentioned in his review, the SEND/RECV media flags are set
according to rtpengine's perspective, not the media flow's one.
commit e1e9a157c0
Author: Razvan Crainea <razvan@opensips.org>
Date: Wed Jan 12 19:27:42 2022 +0200
Fix `subscribe request` SDP media direction
When building the SDP for a `subscribe request` command, take into
consideration the media direction of the source stream - if stream is
`recvonly`, then we do not have anything to send, thus the direction
should be advertised as `inactive`, rather than `sendonly`.
Change-Id: I2d78bbec8ad584774f3c90f0ce5cca42f57f7b0f
Handling of dual stack v4/v6 was previously done by the individual
listener objects for INADDR_ANY listening addresses. If listening on
INADDR_ANY was requested, then each listener would create two instances,
one for IPv4 and one for IPv6. This works fine for INADDR_ANY but fails
for listening on host names that resolve to multiple addresses, such as
`localhost`.
Solve this by relieving the listener objects from handling this and
instead handle it in the code setting up the listeners. If a host name
resolves to multiple addresses, then set up multiple listeners (up to
two supported currently). This allows us to listen on `localhost` by
default and have both 127.0.0.1 and ::1 active. INADDR_ANY is handled
specially by also setting up :: in that case.
Change-Id: I2a1e1d7090d7d23863c7a9bb1e89b85ad2ea44f4
Supplemental codecs such as DTMF use static timestamps while the event
is ongoing, leading to a TS jump when the RTP flow changes back to
audio. The sequencer needs to be aware of this so it doesn't mistakenly
see the next audio packet as overdue and starts to process it
prematurely.
Change-Id: I2faea9aceec21fc04920f6c3c94141725383379f
commit b0c722da69ad088a2eddced12b37c0546a514890
Author: Daniel Hauptmann <dhauptmann@sipwise.com>
Date: Mon Jul 26 15:35:51 2021 +0200
changed flag bit length
in call_interfaces.h changed bit length of reuse_codec from 0 to 1
commit 0313a747532d5987f25fa9edb202aa460bf98dd1
Author: Daniel Hauptmann <dhauptmann@sipwise.com>
Date: Mon Jul 26 15:29:20 2021 +0200
inversed reuse_codec logic
in test-transcode.c and call.c, reuse_codec = 0 (default) will now result in using codec_store_populate instead of codec_store_populate_reuse
commit b876bd686bd30df21a5962aca16fc1c85574f554
Author: Daniel Hauptmann <dhauptmann@sipwise.com>
Date: Mon Jul 26 15:18:19 2021 +0200
adding option to minimalize changes in the codec_store_population
added function codec_store_populate_reuse in codec.c which replaces codec_store_populate but makes fewer changes to the GLists with the old and new codecs
added flag to enable this feature (disabled by default)
commit 6fd0b701c9589b2fae00300801e02a9b5cc397ab
Author: Daniel Hauptmann <dhauptmann@sipwise.com>
Date: Mon Jul 26 14:44:42 2021 +0200
Added Option to minimize change in the codecs
In codec.c added function to populate codec store with the fewest changes between the old and new GList which contains the codecs.
Added new testroutine in test-transcode.c line 1500
Added flag to call_interfaces.h to optionally enable this feature
Change-Id: If58d9a07d114b05dfb75553a87eb4372ae949fbb
commit 3bf554a8fbae7e948343699f40d935693618b764
Author: Daniel Hauptmann <dhauptmann@sipwise.com>
Date: Fri Jul 23 13:58:02 2021 +0200
changing codec-exchange behaviour
in codec.c line 3288 function codec_store_populate now doesnt empty dst and copy new codec from src to it, instead codecs from src will be appended to dst and codec from dst, which are not
being contained by src are being removed
Change-Id: Id6b7ee65595f9cc5c71ef557c7bac5ee38f97cbe
These tests are timing sensitive which makes them unstable under certain
conditions. Remove them from the automated build system.
closes#1309
Change-Id: I432445bce337bbf4d4b80417e532a910b516b8ee
Complete overhaul of the codec handling code:
*) obsolete flags `asymmetric codecs`, `symmetric codecs`, `reorder
codecs`
*) support proper codec offer/answer
*) split codec manipulation (strip/offer/accept/etc) into separate
functions for clarity and better code maintenance
*) fully update codec handlers in both directions after an answer
*) explicit allocation and handling of codecs and payload types in a
codec_store object
*) improve codec matchup logic during answer
*) more explicit handling of supplemental codecs (CN/DTMF)
*) remove now obsolete hacks for handling certain use cases
Change-Id: I996705ba8fe339524c2f70e6bb0fd854f9a1f4fb
This solves problems when the same SSRC is looped through the same call
multiple times in different mono/dialogues, with different parameters.
Change-Id: I1d033cb1f012574d82b5bcbfffe11eb5f983cfd8
We should eventually try to reduce the amount of -Wno-* options, but for
now this is a net improvement.
Change-Id: I3bd03679acbc157c0d1b3c257a542e2eec0e5ee9
This makes the type in line with string(3) functions and eliminates some
compiler warnings.
Also update the related bencode data type.
Change-Id: I7ef4024f4b5a0f737b3dbe03bcd078032395bce6
Richard Fuchs
Michael Prokop
Donat Zenichev
Victor Seva
Razvan Crainea
Stefan Mititelu
hdikme
Daniel Hauptmann
Guillem Jover