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The caller's most preferred codec is selected out of the union of codecs,
which both parties support.
Since codec negotiation is done automatically, there is no need to define
codec for TCH/F and TCH/H via VTY anymore.
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If MNCC application requests a half rate channel, the channel might not be
available, due to different cell configuration, so the full rate channel
is used instead.
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Some RTP endpoints may not check for bad frame indications, so a frame
that is marked as bad may be still forwarded, which creates anoying noise.
This patch drops these frames. It depends on the other RTP endpoint how
dropped frames are handled. (insert silence, extrapolate speech...)
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Traffic cannot sent to BTS, if there is (currently) no logical channel
associated with the transaction.
This happens, if TCH traffic is received from upper layer, but there is
no lchan available before completing immediate assignment, handover or
assignment process.
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The same radio link timeout value is used for BTS and MS side.
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When reading from RTP socket, the first read() may fail right after
connecting to remote socket. Subsequent read() will work as it should.
If the remote socket does not open fast enough, the transmitted RTP
payload can cause an ICMP (connection refused) packet reply. This causes
the read to fail with errno=111. In all other error cases, the errno is
logged at debug level. In all error cases, reading is not disabled.
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After OpenBSC stalled for some reason (e.g. CPU overload or database
access) or after speech frames have been lost (MNCC application problems /
hold/retrieve call), the timestamp and the sequence number of the RTP
socket state must be corrected. The amount of incrmentation is calculated
from the elapsed time. Not incrementing timestamp and sequence number would
cause all frames to be dropped by ipaccess BTS, because the BTS expects
frames with more recent timestamps.
If speech frames are received too fast, they must be dropped. The timestamp
and sequence number of the RTP socket state are not changed in this case.
Incmenetating timestamp and sequence number would causes high delay at
ipaccess BTS, because the BTS would queue the frames until the timestamp
matches the current time.
There is a simple test case:
Make a call between two phones and check the delay. (When using LCR, make
a call to the echo test.) The press CTRL+z to suspend OpenBSC process for
a few seconds and enter "fg" to continue OpenBSC process. There shall be no
change in the delay, even after repeating this test many times.
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Since EFR/AMR/HR codecs use dynamic RTP payload, the payload type can
be set. If it is set, the frame type must be set also, so OpenBSC
knows what frame types are received via RTP.
This modification only affects traffic beween application and MNCC
interface, not the RTP traffic between OpenBSC and BTS.
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Instead of forwarding traffic through MNCC interface, traffic can
be forwarded to a given RTP peer directly. A special MNCC message
is used to control the peer's destination. The traffic can still be
forwarded through MNCC interface when this special MNCC message is
not used.
It also works with E1 based BTSs.
In conjunction with LCR's "rtp-bridge" feature, the RTP traffic
can be directly exchanged with a remote SIP endpoint, so that the
traffic is not forwarded by LCR itself. This way the performance
of handling traffic only depends on OpenBSC and the remote SIP
endpoint. Also the traffic is exchanged with the SIP endpoint
without transcoding, to have maximum performance.
Increment MNCC version to 5.
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If a bad TRAU frame is received, it is forwarded to MNCC application
as GSM_BAD_FRAME. The application can now handle the GAP of missing
audio. (e.g. by extrapolation)
If TRAU frames are forwarded via RTP, bad frames are dropped, but frame
counter and timestamp of RTP sender state is incremented.
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AMR rate is currently fixed to 5.9k.
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Rename method mncc_rcv_tchf() to mncc_rcv_data(), because the check applies
to all types of data frames, not only TCH/F data.
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The new definitions are: half rate and AMR
Change of definition name for bad frame, because it applies to all types of
traffic, not only TCH/F.
Increase MNCC interface version to 4. Version 3 is skipped, because it was
used by older version of Linux-Call-Router which is incompatible with the
current version of the MNCC interface.
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This is not a solution, just a workarround to get rid of llround()
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The interface can be accessed through CTRL and a socket. But currently
it is only available when the socket interface has been configured.
Create the interface all the time but only listen on the socket when
a path has been specified.
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This can be used for the description field that requires some
special handling for newlines.
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For the max power reduction we will need to have a different range
method. It will need to check if the value is even. Make the set,
get and verify methods available through a macro.
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With the current multiplication we might end up with 19999 as
time on i386. When we round it ends up as 20000 on i386 and
should work the same on AMD64.
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On AMD64 we had a difference in the test result most likely due
the bigger size of integers.
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Should fix on AMD64:
mgcp_test.c: In function ‘sendto’:
mgcp_test.c:318:10: warning: format ‘%d’ expects argument of type ‘int’, but argument 5 has type ‘size_t’ [-Wformat]
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So far, the jitter computation has been based on output timestamps.
This patch uses the input timestamps instead and resets jitter
computation on SSRC changes.
Sponsored-by: On-Waves ehf
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Currently, when the SSRC changes within a stream and SSRC fixing is
enabled, the RTP timestamp between the last packet that has been
received with the old SSRC and the first packet of the new SSRC
is always incremented by one packet duration.
This can lead to audio muting (at least with the nanoBTS) when the
wallclock interval between these packets is too large (> 1s).
This patch changes the implementation to base the RTP timestamp offset
on the wallclock interval that has passed between these two packets.
Ticket: OW#466
Sponsored-by: On-Waves ehf
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Currently micro-secs and RTP rate get mixed when the transit value is
computed in mgcp_patch_and_count().
This patch changes get_current_ts() to accept the desired rate as
argument and to use it for the time conversion instead of always
converting to microseconds. If microseconds are needed,
get_current_ts(1000) can be used.
The arrival_time is now measured in 1/rtp_end->rate seconds so that
it can be directly compared to RTP timestamps as required by RFC3550
(section 6.4.1, see definition of 'interarrival jitter').
Sponsored-by: On-Waves ehf
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Currently the stats (jitter, transit) cannot be checked properly,
since they depend on the wallclock time.
This patch fakes clock_gettime (CLOCK_MONOTONIC) to reflect the
scheduling time of the RTP packets. In addition, the RTP statistical
value are written to stdout.
A RTP test case with a SSRC change along with a reference time delay
has been added.
Sponsored-by: On-Waves ehf
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Currently the conn_mode and the output_enabled flags are printed to
stdout.
This patch modifies this to print the output_enabled flags to stderr
instead. The bits in conn_mode are shown as RECV, SEND, and LOOP.
This does not reduce the significance of the test, since there is an
assertion already that verifies the values of the output_enabled
flags with respect to the conn_mode.
Sponsored-by: On-Waves ehf
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Incoming DTAP messages from MS are discarded during silent calls,
which leads to the repeated delivery of SMS since the ACKs are not
being processed.
This patch adds some log messages that have been helpful to track
this down.
Sponsored-by: On-Waves ehf
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E1 based BTS use TRAU muxer to decode TRAU frames. After changing
channel from one timeslot to another (due to handover or assignment),
the TRAU muxer must be updated. The call reference of the call is
disconnected from the old channel and connected to the new channel.
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Add two optional arguments to the imsi-deny rule
for the reject cause and verify that it is saved
out.
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The filtering architecture already allowed to specify a reject
reason but this has not been used for the access-lists. Extend
the access-list to include a reject reason and extend the test
case to honor it.
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Currently, the conn_mode field is reset after it has been checked.
This patch disables this behaviour and only adds a mark (bit) to
detect modifications.
Sponsored-by: On-Waves ehf
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This patch changes implementation and the mgcp_connection_mode enum
in a way that net_end.output_enabled (bts_end.output_enabled) flag
always matches the MGCP_CONN_SEND_ONLY (MGCP_CONN_RECV_ONLY) bit of
conn_mode.
Based on this, the conn_mode bits are then used instead of the
output_enabled fields within mgcp_protocol.c.
Sponsored-by: On-Waves ehf
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Currently RTP output_enabled is set to 1 on initialisation, which
does not semantically match the initial value of conn_mode
(MGCP_CONN_NONE).
This patch changes this initial value to 0.
Sponsored-by: On-Waves ehf
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It seems that also the Ericsson RBS2000 code was assuming that
we always use the bts-global TSC, rather than the possibly different
TS-specific TSC.
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We don't want every caller to check for ts->tsc == -1 and then
using ts->trx->bts->tsc instead. Rather, introduce a new inline
function to retrieve the correct value.
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We introduce a new feature indicating if the given BTS model
supports a TSC that is different from the BCC (lower 3 bits of BSIC).
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If the TS has a specific, different TSC than the BTS (beacon),
we should use that with preference over the TSC of the BTS.
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This patch (hopefully) fixes the new defects reported by coverity.
Addresses:
** CID 1156986: Negative array index read (NEGATIVE_RETURNS)
/tests/gsm0408/gsm0408_test.c: 419 in test_si_range_helpers()
/tests/gsm0408/gsm0408_test.c: 423 in test_si_range_helpers()
/tests/gsm0408/gsm0408_test.c: 427 in test_si_range_helpers()
** CID 1156987: Unchecked return value from library
(CHECKED_RETURN)
/src/libmgcp/mgcp_protocol.c: 1150 in mgcp_keepalive_timer_cb()
** CID 1156988: Unchecked return value from library
(CHECKED_RETURN)
/src/libmgcp/mgcp_protocol.c: 983 in handle_modify_con()
Sponsored-by: On-Waves ehf
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The check is removed from gsm48_cc_rx_setup() and gsm48_cc_rx_call_conf().
Receiving a layer 3 message implies that the transaction has a subscriber
connection and a logical channel.
This patch fixes the Coverity issues with CID 115311 and CID 1155312.
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So far, a single dummy packet has been sent immediately after the
reception of a MDCX message. There is no dedicated keep alive
mechanism (it just worked because the audio from the MS has always
been forwarded to the NAT until the 'mgcp: Set output_enabled flags
based on the MGCP mode' patch).
This patch adds explicit, timer based keep alive handling that can be
enable per trunk. A VTY command 'rtp keep-alive' command is added for
configuration which can be used to set the interval in seconds, to
send a single packet after the reception of a CRCX/MDCX when RTP data
from the net is expected ('once'), or to disable the feature
completely ('no rtp keep-alive'). In 'send-recv' connections, only
the initial packet is sent if enabled (even when an interval has been
configured). The default is 'once'.
Note that this removes the mgcp_change_cb() from mgcp_main.c.
Sponsored-by: On-Waves ehf
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Currently a dummy packet is only sent to the RTP port. This is not
enough if RTCP must also cross the SNAT.
This patch sends an additional dummy packet to the RTCP net
destination if omit_rtcp is not set.
Sponsored-by: On-Waves ehf
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Rename the timestamp variable to make in clear, that the input
timestamp is meant. Add a helper variable to illustrate the offset
computation.
Sponsored-by: On-Waves ehf
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