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
Donat Zenichev
Richard Fuchs
Tom Briden