For M3UA we should have one time-out for operation we want and
then re-transmit it. As this is too much work right now create
a single timer that waits that after a connect the ASPAC_ACK
state will be reached.
In case of back pressure we can not send messages quick enough
and will drop messages. Add a counter so we can count how often
this happened per link/linkset. The SCTP M2UA code is not changed
as we don't have a link in the hands and adding it is too risky
without tests (not all M2UA have a link index so some might not
have a link).
For the "bridging" to IPA mode we can force the SCTP/M3UA connection
down. This way the remote STP will do a proper link fail-over
procedure instead of the STP throwing data away.
This is not configurable yet.
For M3UA links we can't send SLTM messages but we can
assume that the link is fine. Add a flag that a link
can set in case SLTM should be skipped. The linkset will
then directly wait for routable DPCs and bring the
SCCP part up.
Make it possible to configure the transmit power. These settings
will be in effect immediately (there is no lock between the two
threads but it is a read only).
Because of the issue parsing the MGCP request a '\0' was added to
the queue. This lead to the dtmf scheduler believing that a play
was in progress while the playing code didn't play anything. This
lead to the queue to be stuck and nothing being played at all.
Return the number of tones that should be played and stop using
strlen on the tones.
For some equipment it is the easiest to patch the assignment
complete message transported to the MSC. Add a VTY config to
enable this patching, create a testcase that tests that the
original message is truncated. The setting of the VTY option
has been manually tested. The entire system has not been end
to end tested.
This is mostly a hack to allow IPA/SCCP routing to SCTP/M2UA/MTP3/SCCP
without going through the full stack. The proper way of doing this
requires another round of abstraction for the mtp_link_set class.
* Include stdlib.h before the snmp headers to have a free declaration
* Use sprintf(dest, "%s", str) to avoid format string attacks
* Avoid bogus assignment. This pattern was used for marking something
as unused in the past.
This is a hot fix to make CIC reading (and later status) work on
big endian machines. There might be a more elegant way to do it
and I will explore this later.
For the ISUP/MGCP handling we will need the same code, extract it
from the msc_connection. For the reading code callback is introduced
that will pass the MGCP message to the higher layer.
The RSIP has morphed from a global reset, to a per trunk reset and now
it is possible to reset specific ranges on a trunk. This will be used by
the ISUP filter code in the STP.
For legacy range == -1 will be used. This will reset all endpoints
on the trunk. Use OSMO_MAX on endpoint and number of endpoints in case
number_endpoints is 0.
This code will now free everything from the endpoint to endpoint + range
including endpoint+range.
There were several changes in the upstream code. These include
statistics, DTMF/RQNT, changes in the parsing code and re-transmission
handling. The last item is the main reason to do the merge now.
Create a simple queue for pending DTMF tones, play them using the
MTN API, and then send the next tones once the playback is complete.
The callback and scheduling is done from the same context so no locking
needs to be done.
Introduce a callback for the request and forward the signalrequest
to the callback. This is not a full implementation of MGCP RQNT.
Manual merge and backport from OpenBSC.
For a linkset define where SCCP/ISUP should be send. This config
should probably move up to the application part when real work on
the routing is done. Right now the sccp_opc/sccp_dpc need to stay
inside the mtp_layer3.c to be able to send a TFA for the reachable
OPC and it is easier to keep both (dpc/opc) in the same file.
* This changes bss_patch_filter_msg to return -1 or BSS_FILTER_DTAP
for DTAP messages. This way app_forward_sccp should continue to behave
the same besides now looking into DTAP messages.
* Introduce a direction in case we want to advertize FR into the BSS
side and HR into the other direction.
* Patch AMR HR3 and Fullrate/Halfrate capabilities in the Bearer
Capabilities. Add a test case that is patching the bearer capabilities
MGCP RFC 3435 does not specify that the Connection Id must be
generated with any kind of random. It must uniquely identify
the connection of an endpoint. So we can make it per trunk group
or could even have it per endpoint.
The code does not support multiple connections on the same endpoint
right now but the spec allows it.
The endpoint offset is needed for two reasons, first the API is 0
based here while we are normally 1 based, second because of the trunks
the first usable endpoint would be '2' (0 is CRC, 1 is signalling), but
this endpoint offset falls apart when we would block timeslots inside
this range.
Remove the endpoint offset, in each endpoint we will store the HW DSP
Port (1 based API) and then subtract one to get to the 0 based API for
the Simple API. Print a warning when someone is using the endpoint offset.
It appears that it is possible to have a stale SCTP connection
and this added LOGL_NOTICE and the VTY interface might help to
identify this situation in the future (the mean time of failure
is about five month).
We need some way to forward the failure of one link to another but
they are not normally routed so we can not send a TFP. Right now we
will simply stop responding until both links are up. This should make
the SLTM fail and trigger a re-alignment on both sides. The key here
is that the 2 * SLTM timeout needs to be higher than it takes to re-align
the link. I'm not sure this code will work.
Right now for the virtual trunk 0x0 and 0x1F is blocked, for the
E1 like interface we have 0x0 and 0x1 blocked. This should start
to be configurable in the future.
With this commit we can have more than 30 endpoints that will work. We
ignore the blocked endpoints 0x1 and 0x1f for each trunk and calculate
everything from the right start point.
* Upstream has a separation of BTS and NET side for RTP ports and
can allocate them dynamically.
* Upstream has gained the concept of trunks. We will now have various
trunks to connect audio things.
* We will now be able to utilize multiple trunks and have the endpoints
used properly.
Holger Hans Peter Freyther
Harald Welte