Simply take the xmlrpc-callback address as string and don't try to parse
it out. Store it in the call object as string as well.
Obsolete `created_from_addr`. The string form `created_from` is all we
need.
Change `created_from` to `str` as well.
Change-Id: Ib67b57b1d2d474d7b033f56ef8be59f71e44641b
The GLib slice allocator has been obsoleted. Having a mixture of two
allocators can lead to hidden issues on systems that use different
implementations. Take the leap and replace everything with g_new/g_free.
Change-Id: I025c8383ef47b2c2472573360407fd6a6ca339b3
This mode is only supposed to be used by cases when:
- in-dialog offerer doesn't have own MoH capabilities; and
- still want the recipient to hear the MoH music; hence
- an offerer checks whether a recipient is capable of MoH,
and launches a player based on given capabilities;
- the rest functions the same as with usual MoH hold;
Other information:
- `mode=reflect` is only to be used within dialog,
so, it's not meant for a session origination,
and hence stands alone from other MoH option flags
that actually give a metadata (e.g. blob data)
- `mode=reflect` hence contradicts with `mode=sendrecv`
which in its turn serves another purpose
Usage:
- `moh => { mode => 'reflect' }`
Change-Id: I8d0d55f2711c6b47bfca17691582c1ffe66eae3d
Protects against double MoH played,
e.g. when inadvertently two rtpengine instances try to trigger MoH.
This gives a clue to the current rtpengine instance,
that another one already started MoH for this call.
By default is set to false.
Change-Id: I7cc36a177b0ce1bdb64ff8b42bf31f13fb1e4c91
Switch from specialised handling of config sections (used to load
signalling templates) to a more general approach using a callback
mechanism. This allows us to add more information to the config file
while keeping the details of the underlying GKeyFile hidden. Use a typed
hash table for type safety.
Change-Id: I71ddfda0202b47df363bcc5acf1725078774f8f1
Change old code to directly return `str` objects instead of pointers to
allocated ones. Largely a no-op change but makes some code easier and
eliminates some old kludges.
Change-Id: I2be19dd24bb7ff046d86cc32a3af235283e65dd0
Make `moh-max-duration` available for both
music-on-hold functionality as well as
for the media player.
For that to work, do the following:
- keep `moh-max-duration` config option only for MoH,
if not set (so 0) by default is 1800000ms (half an hour)
- for the play media functionality introduce flag option
`repeat-duration`, by default is disabled
Policy changes:
- duration counter can be used in common with repeats
counter, but then takes a precedence over it.
Hence if first a duration is underflown, then EOF triggered.
Otherwise if the duration counter is still positive, but
repeats are negative, then do EOF based on repeats.
- the repeats counter will always count down during each
iteration, even when used together with the duration counter
For MoH to survive, the repeats counter is simple set to 999
to let the duration counter always win over repeats one
- MoH cannot take duration disabled, since otherwise
would make no sense for it. Hence always takes internally
defined value 1800000ms (half an hour) if not defined
by the configuration option
Backwards compatibility:
- is kept in regards of repeats counter
- is kept in regards of the play media functionality
Change-Id: I48ff3c17c9bed31f80c3106b275b703a9ccb4b26
Introduced Music on Hold functionality:
- available only for the offer/answer model,
no other scenarios (publish, subscriber etc.)
are covered with it
- it gets advertised always at the beginning of
the call (original offer/answer exchange)
- can be added for both sides: offerer/answerer
- the one who advertises its MoH capabilities
with its SDP offer or answer, can later trigger
MoH using sendonly SDP and unhold remote
party using sendrecv SDP
- MoH covers only audio type of media sessions
- there is no specific selection of media sections
to be held, thus, if one audio media puts the
call on hold, the whole call is held
- list of parameters to be given when advertising
MoH capabilities: a sound source (file, blob or db source);
sendonly/sendrecv hold; zero-connection hold;
At least an audio file source must be given
- MoH cannot be mixed with the play media functionality,
the last one triggered will override previous one
- MoH must be unheld to stop the media being sent
towards a recipient, otherwise only a termination
of monologues will stop this packets stream
Change-Id: Iefd83ced79c14dadad936348a1d529007d6e7b3b
There is enough overlap between the two enums that it doesn't make sense
to carry along both. Unify.
Change-Id: I9de8fbdb8d78a2002d8c1b62cea57188c937d61d
Allow to-tag updates in answers only until a non-provisional response is
received. Add a flag to optionally suppress this feature.
Change-Id: I67110ede662f618ba5a355500d3d84050e6318cc
If we receive an answer from a to-tag that hasn't previously seen and no
corresponding monologue exists (created from a previous offer),
previously we would treat this is a separate and new call branch, create
a brand new monologue and dissociate the previous one. This may lead to
unexpected results as this new monologue has been created without the
same initialisation as was done for the original one, and so may be left
with incorrect or incomplete data (e.g. SRTP keys, codec information,
interface bindings, etc).
Improve this by treating an unexpected and unseen to-tag as an alias to
the already existing to-tag. Going forward both tags can be used
interchangeably to refer to the same monologue.
Add a flag to suppress this new behaviour, in case some situation is
made worse by it.
Change-Id: Ie8f838eebd50d29d3549031998a2eb1f610b04bb
In order to be able to control, which media types
one wants to have in the call session,
add support of the "sdp-media-remove" flag.
Syntax:
"sdp-media-remove" : ["<media-type>", "<media-type>", ...]
Additionally, add according unit tests.
Change-Id: Ic52456f8124319992ea9ca8c161daefb1df46b59
Convert existing sdp manipulation helpers
to work with arrays directly, independently from
which array is actually given.
Moreover, don't use the `G_N_ELEMENTS()`,
since we already have the `__MT_MAX`
as the size of the array, so we can directly use
that instead of relying on the G_N_ELEMENTS macro
(for array size checks).
Change-Id: I38e71dab30a467b933c213c8157fa06651ebaf04
Add `call_ng_process_flags()` based parsing
as for other opmodes, like offer and answer.
This keeps the backwards compatibility with
the older "flags" parsing approach on the
module side, as well as adds the possibility
to parse rtpp-flags on the daemon side.
As an advantage, there is no need to use
specific local parsing for things like
to/from tags, call-id, delete-delay etc.
Additionally:
- this commit introduces flags-flags parsing
for the "fatal" flag.
However, as before is only taken into account
by the `call_delete_ng()` processing,
so no functional change.
- this commit introduces main-flags parsing
for the "delete-delay" flag, which is also
only taken into account by the `call_delete_ng()`
processing, so no functional change.
- this commit adds To-tag options flag prasing
into the `call_ng_flags_flags()` function,
and is used by `call_delete_ng()` specificially,
for cases when more specific identification of
monologues to be deleted is used.
Change-Id: Ia992e5375a2f86318d9ad193a7857dd589038eed
Support session level group attribute
for the sdp_create approach.
Additionally: fix the `testVideoroomWebRTCVideo` test,
because as it turned out it used to lose the `a=group:`
attribute before.
Change-Id: I799dfe0dcc5e0708864150cd0e0262a61dd4cc05
Added support of the CT (conference total) bandwidth
attribute for the SDP session level. See RFC8866.
Will be required later for the SDP formatting in `sdp_create()`.
Change-Id: Ifc64f68d7acee8ce253882f4fa480bbf7ad7c0bd
... so that the desired wav channel can be controlled when producing a
mixed audio file
When a mixed wav file is created, the channels in the wav container are
currently allocated in the same order as each SSRC is received, meaning
it is impossible to know which channels have been allocated to the offer
or answer side of the call. Furthermore if there is a reinvite or media
file played, these are also allocated in the order that SSRC is received
- so an "answer" could end up sharing a channel with an "offer" with no
way of knowing this.
This patch allows you to specify how many channel slots should be
allocated within the mixer, and allows you to then specify which slot is
assigned to each media in the call (this will usually be 2 slots in
total, slot 1 for answer, slot 2 for offer or vice versa).
Ported from https://github.com/sipwise/rtpengine/pull/1852Closes#1857Closes#1852
Change-Id: I010208427cabc3a48d6ef7bd3a84e9a5bdcfd492
Added support of the AS bandwidth attribute for the
SDP session level. Will be required later for the SDP
formatting in `sdp_create()`.
Change-Id: I1bde4659679de6e60bdad12c0578ced2c1983300
Richard Fuchs
Donat Zenichev
Michal Hajek
Peter Kelly