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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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queue status
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The code in the BSC/NAT called ipaccess_rcvmsg_base without
checking if the protocol is IPA. This lead the BSC to respond
to SCCP messages with an "ID ACK". From a quick look neither
the code of ipaccess_rcvmsg_base in OpenBSC nor the copy of
libosmo-abis ever checked the protocol header. So this code
has been wrong since initially being created in 2010.
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This is an incompatible database schema change. Store the type of
the address in the database for both the sender and the receiver.
Currently it is possible to use SMPP to store a SMS and the NPI
and TON will be lost on the delivery of the SMS. The schema is
changed to make the delivery always use the right NPI/TON. This
patch is not ready for the master branch as there is no upgrade
path for the HLR yet.
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Signed-off-by: Alvaro Neira Ayuso <anayuso@sysmocom.de>
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sgsn_main.c: In function ‘main’:
sgsn_main.c:345:2: warning: implicit declaration of function ‘gprs_sndcp_vty_init’ [-Wimplicit-function-declaration]
gprs_sndcp_vty_init();
^
sgsn_main.c:354:2: warning: implicit declaration of function ‘sgsn_gtp_init’ [-Wimplicit-function-declaration]
rc = sgsn_gtp_init(&sgsn_inst);
^
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CC gprs_gmm.o
gprs_gmm.c: In function ‘gsm48_tx_gmm_att_ack’:
gprs_gmm.c:350:11: warning: unused variable ‘ptsig’ [-Wunused-variable]
uint8_t *ptsig, *mid;
^
gprs_gmm.c: In function ‘gsm48_rx_gmm_auth_ciph_resp’:
gprs_gmm.c:524:6: warning: variable ‘rc’ set but not used [-Wunused-but-set-variable]
int rc;
^
gprs_gmm.c: In function ‘gsm48_rx_gmm_att_req’:
gprs_gmm.c:703:9: warning: implicit declaration of function ‘sgsn_acl_lookup’ [-Wimplicit-function-declaration]
!sgsn_acl_lookup(mi_string))) {
^
gprs_gmm.c:632:40: warning: variable ‘old_ra_info’ set but not used [-Wunused-but-set-variable]
uint8_t *cur = gh->data, *msnc, *mi, *old_ra_info, *ms_ra_acc_cap;
^
gprs_gmm.c: In function ‘gsm48_rx_gmm_ra_upd_req’:
gprs_gmm.c:915:6: warning: variable ‘rc’ set but not used [-Wunused-but-set-variable]
int rc;
^
gprs_gmm.c:910:11: warning: variable ‘ms_ra_acc_cap’ set but not used [-Wunused-but-set-variable]
uint8_t *ms_ra_acc_cap;
^
gprs_gmm.c: At top level:
gprs_gmm.c:458:12: warning: ‘gsm48_tx_gmm_auth_ciph_req’ defined but not used [-Wunused-function]
static int gsm48_tx_gmm_auth_ciph_req(struct sgsn_mm_ctx *mm, uint8_t *rand,
^
gprs_gmm.c:501:12: warning: ‘gsm48_tx_gmm_auth_ciph_rej’ defined but not used [-Wunused-function]
static int gsm48_tx_gmm_auth_ciph_rej(struct sgsn_mm_ctx *mm)
^
gprs_gmm.c:1169:13: warning: ‘msgb_put_pdp_addr_ipv4’ defined but not used [-Wunused-function]
static void msgb_put_pdp_addr_ipv4(struct msgb *msg, uint32_t ipaddr)
^
gprs_gmm.c:1180:13: warning: ‘msgb_put_pdp_addr_ppp’ defined but not used [-Wunused-function]
static void msgb_put_pdp_addr_ppp(struct msgb *msg)
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The old ipa_msg_recv() implementation didn't support partial receive,
so IPA connections got disconnected when this happened.
This patch adds the handling of the temporary message buffers and uses
ipa_msg_recv_buffered().
It has been successfully tested by jerlbeck with osmo-nitb and
osmo-bsc.
Ticket: OW#768
Sponsored-by: On-Waves ehf
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In case the max_power_reduction changes, issue a new Set Radio
Carrier Attributes command. OML 12.21 allows to not include the
ARFCN list and the semantic I picked/understand is that a partial
update is possible.
Fixes: SYS#267
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Add the control interface with no hierachy right now and implement
the first command to list IMSI + Context Address of active sessions.
sgsn_cmd_handle could share more code with bsc variant.
Fixes: SYS#264, SYS#265
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For GPRS the look-up via bts/trx does not make any sense and would
introduce bad depdencies for the SGSN. Move the look-up code to a
new file and introduce new setup methods.
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This is only useful for small networks. List the IMSI and MSISDN
of all active subscribers.
Fixes: SYS#266
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Sadly there is no proper foreign key relationship on the tables
that related to the Subscriber. This means we can't use a DELETE
with Cascade and need to delete everything by hand. To make things
worse maybe the SMS/Paging code is still using the subscriber
making the operation more dangerous. I had added NULL checks for
sender_id/receiver_id at 30C3 so we should not crash in this
situation.
Fixes: SYS#274
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The test has been manually verified. Executing the select for
the subscribers showed:
sqlite> select * from Subscriber;
1|2014-03-23 12:12:46|2014-03-23 12:19:09|2620345||445567|1||0|
This created a subscriber with the right IMSI, MSISDN and has
it authorized.
Fixes: SYS#275
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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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In MT-SMS the message is being delivered. Make the naming follow
that. The schema still refers to "sent" while it should be "delivered"
too.
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Add the framework for adding more setting commands.
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I have manually tested this by entering the VTY command and
observing the CTRL interface using wireshark.
Ticket: OW#1129
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Each RP-DATA should have a unique msg reference. Currently 42 is
used for all of these. Remember the last reference we used and
increment it on the next SMS. Do not track if the reference is
still in use a clash is a lot less likely now. First unless SMPP
is used only one SMS is delivered at a time, second the transaction
space is a lot smaller than the one for the reference.
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In file included from bsc_api.c:34:0:
../../include/openbsc/trau_mux.h:62:15: warning: ‘struct decoded_trau_frame’ declared inside parameter list [enabled by default]
const struct decoded_trau_frame *tf);
^
../../include/openbsc/trau_mux.h:62:15: warning: its scope is only this definition or declaration, which is probably not what you want [enabled by default]
../../include/openbsc/trau_mux.h:64:15: warning: ‘struct decoded_trau_frame’ declared inside parameter list [enabled by default]
const struct decoded_trau_frame *tf);
^
../../include/openbsc/trau_mux.h:66:2: warning: ‘struct decoded_trau_frame’ declared inside parameter list [enabled by default]
const unsigned char *data);
^
../../include/openbsc/trau_mux.h:68:2: warning: ‘struct decoded_trau_frame’ declared inside parameter list [enabled by default]
const unsigned char *data);
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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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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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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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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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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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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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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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This patch make it possible to have a valid endpoint that drops all
outgoing RTP packets. The number of dropped packets is shown by the
VTY 'show mgcp' command. By default, this feature is disabled. To
enable packet dropping, the corresponding output_enabled field must
be set to 0.
Ticket: OW#1044
Sponsored-by: On-Waves ehf
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f0 is currently set to arfcns[0] in range_enc_determine_range(),
while GSM 04.08 requires f0 to be ARFCN 0 in range1024 encoding.
This patch modifies range_enc_determine_range() to force f0 to be 0
if this encoding is used. This way the case distinction in
range_enc_filter_arfcns() is not longer necessary.
Sponsored-by: On-Waves ehf
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The iPhone5 (US) appears to have some issues with the SIs generated,
or the nanoBTS is not sending them correctly.
Add a configurable hack to put all bands into the SI2/SI5 message.
It is enabled by the bts VTY command 'force-combined-si'.
This is a quick change without much reflection and watching for side
effects. I have verfied that a network with ARFCN 134 and neighbors
ARFCN 130 and 512 do not get generate the SI2ter and announce everything
inside the SI2.
This patch is conceptually based on 'si: Add a hack to disable
SI2ter/SI2bis/SI5ter/SI5bis messages' (692daaf2d2).
Ticket: OW#1062
Sponsored-by: On-Waves ehf
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Speech codings which are not supported by BTS will be removed from the
bearer capability information element after parsing. This way it is not
required for the MNCC application to consider support of each BTS.
Only GSM full rate is supported by default.
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This option is a workarround for a bug found in Nokia InSite BTS firmware
version 3.0.0. There is no RELease CONFirm message for local end release.
Nokia MetroSite with firmware version 4.178.16 is not affected.
TS 04.06 Chapter 5.4.4.4 "Local end release procedure" states that a
confirm must be sent by layer 2 when receiving a local end release
request.
In order to correctly switch a channel (handover or assignment), local
end release is required.
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Currently, all timestamps are force to SeqNo*d + C which is more than
required by the nanoBTS which seems to be sensitive to alignment
errors only (dTS != k*d, d = ptime * rate = 160).
This patch replaces the force_constant_timing feature by a
force_aligned_timing feature. The timestamp offset will only be
changed (and timestamp errors counted) when the alignment does not
match to the raster based on ptime (default 20ms).
The VTY interface does not change.
Sponsored-by: On-Waves ehf
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Decoding and encoding of FR and EFR TRAU frames are put into seperate
functions. CRC check is done to detect bad EFR TRAU frames.
The test case includes FR and EFR transcoding.
EFR support was tested with Nokia InSite BTS and Siemens BS11.
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Currently the SDP 'ptime' media attribute is never set in generated
MGCP responses.
This patch optionally includes the 'ptime' attribute if
packet_duration_ms is != 0. This behaviour can be enabled/disabled
by using the VTY command "sdp audio-payload send-ptime" (enabled by
default).
Sponsored-by: On-Waves ehf
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Currently the local connection options have been stored as a string.
This patch replaces this string by a struct (that still contains a
string) along with the parsed fields (only the packetization period
at the moment).
It also re-adds the calls to set_local_cx_options() to the
handle_create_con() and handle_modify_con() functions. Except for
the test program this has no side effects, since the LCO values
aren't used yet.
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Since the packet duration is given in ms with the 'ptime' RTP media
attribute and also with the 'p' MGCP local connection option, the
computation is changed to use this value (if present). The
computation assumes, that there are N complete frames in a packet and
takes into account, that the ptime value possibly had been rounded
towards the next ms value (which is never the case with a frame length
of exact 20ms).
Sponsored-by: On-Waves ehf
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This forces the output timing to fulfill
dTS = dSegNo * fixedPacketDuration
where dSegNo = seqNo - lastSeqNo.
If timestamp patching is enabled, the output timestamp will be set
to lastTimestamp + dTS. This kind of relative updating is used to
handle seqNo- and timestamp-wraparounds properly.
The updating of timestamp and SSRC has been separated and the patch
field of mgcp_rtp_state has been renamed to patch_ssrc to reflect
it's semantics more closely. The offset fields are now used always
and will change the corresponding header field if they are != 0.
Ticket: OW#1065
Sponsored-by: On-Waves ehf
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Currently the output SSRC is always forced to be the same if SSRC
patching is enabled.
This patch modifies this to optionally restrict the number of SSRC
changes that will be corrected.
Note that the configuration only allows for the 'once' mode and 'off'.
Sponsored-by: On-Waves ehf
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This patch adds a packet_duration field to mgcp_rtp_state which
contains the RTP packet's duration in RTP timestamp units or 0, when
the duration is unknown or not fixed.
Sponsored-by: On-Waves ehf
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This adds datastructures and a VTY frontend to configure the
different type of RTP header patching: SSRC and timestamp.
Note that timestamp patching is not yet implemented.
Sponsored-by: On-Waves ehf
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So far the payload type used in RTP streams has been taken from the
trunk configuration in NAT mode.
This patch changes the implementation to use the payload type
announced in the SDP part of MGCP messages and responses. SDP
descriptions more than one m=audio line are not yet supported
properly (always the last one is taken).
Ticket: OW#466
Sponsored-by: On-Waves ehf
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The current implementation increments the seqno but does not increment
the RTP timestamp, leading to two identical timestamps following one
after the other.
This patch fixes this by adding the computed tsdelta when the offset
is calulated. In the unlikely case, that a tsdelta hasn't been
computed yet when the SSRC changes, a tsdelta is computed based on
the RTP rate and a RTP packet duration of 20ms (one speech frame per
channel and packet). If the RTP rate is not known, a rate of 8000 is
assumed.
Note that this approach presumes, that the per RTP packet duration
(in samples) is the same for the last two packets of the stream being
replaced (the first one).
Sponsored-by: On-Waves ehf
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This patch modifies the patch_and_count() function to check for RTP
timestamp inconsistencies. It basically checks, whether dTS/dSeqNo
remains constant. If this fails, the corresponding counter is
incremented. There are four counter for this: Incoming and outgoing,
each for streams from the BTS and the net.
Note that this approach presumes, that the per RTP packet duration
(in samples) remains the same throughout the entire stream. Changing
the number of speech frames per channel and packet will be detected
as error.
In addition, the VTY command 'show mgcp' is extended by an optional
'stats' to show the counter values, too.
Ticket: OW#964
Sponsored-by: On-Waves ehf
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