Add per b-channel private data (ftdm_libpri_b_data_t), move the q.931 libpri
call pointer into it and use channel->call_data to make it accessible.
Use ftdm_channel_wait() instead of select() to wait for I/O events on
the D-Channel and rewrite the timeout handling, adding support for custom
timers that can be added to the b-channel private data and started/stopped
with lpwrap_timer_start()/_stop().
The (sorted) list of active lpwrap timers is protected by the timer_mutex,
which is initialized in the now unified lpwrap_init_pri() (see below) and
destroyed in the new lpwrap_destroy_pri() function.
Unify lpwrap_init_pri() and lpwrap_init_bri(). Use the span type to select
BRI/PRI and Point-to-Point vs. Multipoint settings.
Use the new custom lpwrap timer API to implement T302 (aka overlap receiving
digit timeout). The default T302 timeout is 3000 miliseconds and can be set
via the "digit_timeout" or "t302" span configuration parameter
(0 = disable timer, valid range = 3000 ... 30000 miliseconds).
The lpwrap_pri_init() changes made it neccessarry to rewrite the span runtime
thread event loop. The d-channel initialization and libpri event handler
registration have been moved out of the event loop. The b-channel restart
handling in the event loop has been left as-is, pending a further clean up in
the future.
Rename on_info() to on_information() and add a new on_keypad_digit() event
handler to convert incoming KEYPAD DIGIT messages to channel DTMF events
(only useful on BRI PTMP NT-mode).
The channel restart handling has been improved to distinguish between locally
triggered and restarts caused by the remote end (= incoming RESTART messages).
Restarts on the D-Channel now use pri_restart(), causing a Q.921 reset on
the span.
Large patch, i'd have loved to split this into smaller pieces and actually
test some of the bits more toroughly (restart handling), but some people
are f*cking annoying, so here it is.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Move the B-channel message handling into a per-span I/O thread,
to solve most of the problems caused by the intermixed data + control
socket interface of mISDN, missing write poll() support on
mISDN B-channels and the FreeTDM I/O model. This eliminates most of
the audio problems (except for a few minor glitches).
A unix stream socket pair is used as a bi-directional pipe replacement
(the pipe code is still included in this commit, but will be removed later),
with the RX and TX buffer sizes carefully tuned to avoid excessive buffering
(= latency) and a deadlock situation between the write() call in ftdm_write()
and the code in misdn_span_run() that needs a minimum amount of data in the
TX buffer, before sending out a PH_DATA_REQ to the mISDN socket
(see misdn_span_run() comments for more details).
The minimum size for pipes is PAGE_SIZE (4k), which is ~500 ms worth of
audio. A socket pair RX/TX buffer size of 3k, seems to hold a maximum
amount of around 500 bytes data in practice, giving us a much lower
maximum latency than a unix pipe. (The socket pair might be replaced by a
custom ring buffer / fifo data structure to get even more fine grained
control of the maximum latency.)
The newly introduced span_start / span_stop callbacks in
ftdm_io_interface_t are used to start / stop the I/O thread. The callback
functions will wait up to 10 seconds for the thread to successfully
start up or shut down (using a mutex + condition var).
NOTE: Using any of the locking ftdm_span_() functions in the I/O will cause
a deadlock between the I/O thread (trying to lock span->mutex) and the
thread calling ftdm_start()/_stop() (holding the span->mutex).
(The I/O thread currently uses direct span member access to avoid this.)
The I/O thread uses the epoll(7) family of functions for event handling.
An epoll context is created on startup and all B-channel sockets are
registered (READ, PRI and ERR). Before entering the event loop,
the I/O thread will send a signal on the condition variable, to
indicate it has completed the startup procedure.
Incoming b-channel and command pipe events are handled by the event loop.
Payload of incoming PH_DATA_IND frames (= audio data) is sent to the
rx_audio_pipe_in end of the b-channel's socket pair and, if enough data is
available, a PH_DATA_REQ of the same size is sent to the b-channel mISDN socket
to transmit audio.
A MISDN_CMD_STOP command on the event pipe will wake up the I/O thread and
cause it to shut down. All b-channels will be unregistered from the epoll context
and the epoll fd closed. The I/O thread terminates itself after signalling the
successfull shutdown on the condition variable.
TODOs:
- Move D-Channel into I/O thread too
- Custom FIFO/ring buffer for data (even lower latency)
- Improve epoll() code (per-channel struct w/ callback, for epfd.data.ptr)
- Use mISDN DSP for audio (e.g. tone generator, dtmf detector, echo cancel)
- Use a per-port / span control socket to execute channel commands
synchronously, or add misdn_commands (queue?) that can be used that way
- Name I/O threads 'mISDN-%SPAN_NAME%', e.g. 'mISDN-M_BRI1'
(= add ftdm_thread_set_namef(thread, fmt, ...) / ftdm_thread_set_name(thread, name))
TL;DR: "tweak", solves "booboo" with audio
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Hunt for a free channel for incoming calls that do not
preselect a channel (pevent->ring.channel == -1).
Verify the preselected channel for calls that do specify a channel
and in case the channel is already taken, hunt for a free one,
or abort with an error message (if the preselection was exclusive).
TE-mode channel selection is the same as before
(there's still room for improvement, though, but i'll save that for later).
The MSN/DDI filter code is moved into the TE-mode section (only useful there).
The duplicate ring detection had to be reworked. We now store the
call reference (CRV) in caller_data->call_reference of the selected channel
and do a CRV -> channel look up with find_channel_by_cref()
at the top of on_ring().
NOTE: This is only lightly tested (NT/TE mode), i'd either have to
write a lot of custom code to check it toroughly or the need for
a scriptable ISDN stack...
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Callbacks are invoked from ftdm_span_start/_stop().
I/O is started before SIG and shut down in reverse order.
This is needed for ftmod_misdn, to move the mISDN message handling
into a separate thread (solving the mISDN socket vs. FreeTDM API issues).
With these callbacks, the I/O thread can be started after the span I/O configuration
has been (successfully) completed and stopped before destroying the span.
NOTE: Both SIG and I/O callbacks are called with the span mutex locked,
so threads created or destroyed synchronously in either of the custom
start/stop functions, can not use ftdm_span_*() functions that lock
the span mutex (e.g. ftdm_span_get_channel_count()).
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
The len variable can, in certain situations (large burst of incoming non-SLIN audio),
exceed the size of the on-stack frame buffer, which causes ftdm_buffer_read_loop() to
overwrite the dt_buffer pointer.
Use ftdm_min() to make sure len (after conversion to SLIN units) isn't larger
than the frame buffer size.
Also adds are couple more code comments.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Resolves OPENZAP-189
Fixes:
src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'ftdm_isdn_931_34':
src/ftmod/ftmod_isdn/ftmod_isdn.c:902:21: error: variable 'status' set but not used [-Werror=unused-but-set-variable]
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
. added called party number INN variable
ie. <action application="export" data="freetdm_ss7_cld_inn=0"/>
. added multiple variables to set User Service Information IE
variables are :
ss7_iam_usi_trans_cap
ss7_iam_usi_code_standard
ss7_iam_usi_trans_mode
ss7_iam_usi_trans_rate_0
ss7_iam_usi_trans_rate_1
ss7_iam_usi_layer1_ident
ss7_iam_usi_layer1_prot
ss7_iam_usi_layer2_ident
ss7_iam_usi_layer2_prot
ss7_iam_usi_layer3_ident
ss7_iam_usi_layer3_prot
ss7_iam_usi_chan_struct
ss7_iam_usi_config
ss7_iam_usi_establish
ss7_iam_usi_symmetry
ss7_iam_usi_rate_multiplier
. ss7_iam_usi_trans_cap is a string variable, the others are all integers
. ss7_iam_usi_trans_cap has options of :
- SPEECH
- UNRESTRICTED
- RESTRICTED
- 31KHZ
- 7KHZ
- 15KHZ
- VIDEO
default value is SPEECH, if the parameter is wrong, set to SPEECH. If not
set, this field is not present
Fixes:
src/ftmod/ftmod_misdn/ftmod_misdn.c: In function 'misdn_handle_mph_information_ind':
src/ftmod/ftmod_misdn/ftmod_misdn.c:871:3: error: format '%lx' expects argument of type 'long unsigned int', but argument 13 has type 'uint64_t' [-Werror=format]
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Do a "soft" wraparound with modulo, removes the ~0.5s tone glitch.
(Multiply ts.rate (samples) by two to match the offset unit (bytes, 2 per sample).)
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
