Age | Commit message (Collapse) | Author | Files | Lines |
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Handover is a high level decision, it can span multiple BSCs
and belongs mostly into the MSC domain.
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For the time being RSL has to know about Layer4 and upwards
and is using the RTP socket class....
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This provides two functions: get_meas_rep_avg() to obtain the sliding
window average of one particular field, and meas_rep_n_out_of_m_be()
to check if at least N out of M measurments are >= BE.
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With this commit, we can successfully hand over a channel from one cell to
another cell. We implement asynchronous intra-BSC (but inter-BTS) handover.
Changes:
* introduce new DHO log category
* extend rsl_chan_activate_lchan() with argument for HO reference
* introduce actual minimal handover decision making in handover_decision.c
* various fixes to bsc_handover_start() in handover_logic.c
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Before this commit, OpenBSC used templates for the SYSTEM INFO
1, 2, 3, 4, 5 and 6 messages. Those templates were patched in
various places to reflect the network config like ARFCN.
Now, we actually generate those SI messages ourselves, using
values from the configuration file, and even calculating neighbor
cell lists.
All bts'es that you have configured in OpenBSC will end up in
the neighbor cell list - which should be more than sufficient for
the current small-single-site networks.
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Code to implement handover control logic. A yet-to-be-implemented
handover algorithm will call bsc_handover_start(old_lchan, new_bts)
to start the handover process.
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Mention the two new header files, do not list isdnsync twice
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The python script is a simple call-agent driving the
client. Currently it is sending a AuditEndpoint message
and is printing the result.
The bsc_mgcp.c is a standalone process that will implement
a MGCP Gateway for the MSC. On call handling the Call-Agent
will ask the Gateway to "CreateConnection" and then this
gateway needs to communicate with OpenBSC.
Currently CreateConnection,ModifiyConnection,DeleteConnection
and Endpoint auditing is implemented.
[mgcp] Send RSIP on start and on first receive of any message
Ignore the first request and send a RSIP. We do that because
we might tunnel UDP through some other things and have no direct
way to connect to the call-agent.
Also the transaction is not checked and we ignore the response
from the call-agent, actually we print the '200 ' or any other
value as unhandled...
[mgcp] Print the MGCP command next to the response code
This allows to see which commands were sent by the server
mgcp: Terminate it with a new line
[mgcp] Make number of endpoints static...
For now this is fixed to the number of endpoints as of the GSM
specification...
[mgcp] The endpoint names seem to be base 16... use strtoul to parse
Use strtoul to parse the base 16 number from the mgw string.
[mgcp] Log the endpoints as hex numbers...
[mgcp] Only send the RSIP on the first incoming message..
Remove call_agent option (also remove the number from the getopt
call).
[mgcp] Start couting at 1 for the mgcp
[mgcp] Slight attempt to improve the grammar of the strings
[mgcp] Share validation routines between DLCX and MDCX
[mgcp] Remove help for dead config options
[mgcp] Specify a different IN addr in the SDP records
In case of NAT traversal be able to listen on a given
interface (like 127.0.0.1) but claim to receive data
at the beginning of the tunnel.
[mgcp] Fix the static copy of the SDP file
WIP verify out factoring broken..
[mgcp] Introduce VTY to the mgcp for config file parsing...
Parse the MGCP config file via the VTY framework.
[mgcp] Handle SDP parameters through VTY..
Currently the payload type, name and rate can be specified
in the config file.
[mgcp] Add an option to bind all rtp ports early
This can be useful for testing and in deployment to make sure
no runtime resource limit can be hit.
[mgcp] Add some API doc comment
[mgcp] Convert the packets of the example server to ascii
This will allow to easily patch the call id... to run the
server in a loop and make it work with the mediagateway
[mgcp] Assign CI_UNUSED... to be more obvious...
[mgcp] Use DEBUG and not DEBUGPC and specially not printf
Improve the logging a bit in the mgcp
[mgcp] Change the fake server to change the call id
This assume the call-agent will just increment the id
as well.... this is true for our implementation
[mgcp] Generate the transaction id dynamically..
This way wireshark will be more happy about it...
[mgcp] Recognize responses from the network..
This is just recognizing the response code and
then is doing nothing with it. Also change the
script to generate response messages...
[mgcp] Improve debug messages for CRCX/MDCX..
Log on which ports the media gateway is listening
and where the other (server) gateway is located
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include/sccp/sccp_types.h contain Q.713 and GSM definitions
include/sccp/sccp.h is the application interface resembling
the esentials of the UNIX socket interface.
src/sccp.c is the actual implementation of SCCP featuring
connection and UDT1 support.
tests/sccp/sccp.c is testing connection creation and formating
of the SCCP messages used by the A-interface. And
it contains a simple fuzzing test to test the
robustnes of the implementation.
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This allows the administrator to use the vty interface to issue a silent
call to a given subscriber by using
"subscriber extension XXXX silent call start"
and stopping that silent call with
"subscriber extension XXXX silent call stop"
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This is the initial checkin of the USSD code from Mike Haben. I didn't
put it in the main branch as I think it still needs some cleanup.
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Share the initialization and bootstraping of the network by moving
the code to a new file and making boostrap_network and shutdown_net
external.
Cleanup the header list after the move and remove trailing whitespace.
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Move everything that is policy, requires access to a DB or is
generally in the domain of the MSC to vty_interface_layer3.c.
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The existing gsm_04_08.c implementation is mixing BSC and MSC
behavior. Move some simple parsing and generation functions over
to gsm_04_08_utils.c to allow a different MSC to define the policy.
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Currently we have circular dependencies from libbsc to libmsc
and this requires to play some linker tricks. The problem will
be solved in two ways, first we will get rid of the circular
dependencies and second we can start using --start-group and
--end-group of the linker to play the tricks for us.
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For the BSC part we still assign a gsm_subscriber to lchan but it
might only contain the TMSI of this subscriber.
For the MSC part we will need the HLR/VLR feature of the gsm_subscriber,
specially the lookup's by number...
So if libbsc.a/libmsc.a are compiled in one app and used the
subscribers will be shared, and if only libbsc.a gets used we will
have more empty gsm_subscriber.c..
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Attempt to split up bsc/msc functionality according to the specs. The
libbsc.a will be responsible for communicating with the BTS, configuring
it, paging, channel allocation and passing layer3 messages in both
ways. libmsc.a will implement the policy and such.
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attach
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the various constructors get called in a non-obvious, linker determined
order, which makes certain objects disappear from the talloc report.
This change moves the talloc context creation into a new talloc_ctx.c file
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* When we first see a subscriber, send the sms
* when the sms completes, send auth req + auth reject and close the channel
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A caller can call rll_establish(lchan, link_id) and a callback to the GSM RLL
code. He will get called back if the RLL link is established or receives some
error message, or the establishment times out.
We need this for proper SMS implementation, where we need to restablish a SAPI3
RLL link before transmitting the actual CP-DATA messages.
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Up until now, we only supported direct RTP streams between ip.access BTS.
With this commit, the user can specify '-P' to the command line to enable
a RTP/RTCP proxy inside OpenBSC. The nanoBTS will then send all their voice
data to OpenBSC, which will relay it to the respective destination BTS (which
can be the same BTS).
The default behaviour remains unchanged. Without '-P' on the command line,
RTP/RTCP is exchanged directly.
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This changeset factors out gsm_transaction as something independent
of call control in preparation to re-use the code from SMS. A
transaction is uniquely identified by either its callref, or by
a tuple of (transaction_id, protocol, subscriber).
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This is Harald's reworked MNCC base, slowly heading towards integration
into master. The key changes are:
* provide much more structure to the data in gsm_mncc
* encode_* and decode_* functions now take a structure rather than tons
of individual arguments (whose order nobody can remember)
* make sure we don't have copies of the same code everywhere by introducing
mncc_set_cause() and mncc_release_ind()
* save horizontal screen space if possible
* make sure we break lines > 80 characters
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