Age | Commit message (Collapse) | Author | Files | Lines |
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In case of the RTP bridge mode we need to select the codec
ourselves. Rely on the same (incomplete) codec selection that
can be done using the mncc-int configuration node. This might
gain bearer capabilities support.
In case of a SDCCH a TCH/F will be attempted to be assigned.
This is an open issue for both modes and there should be a
preference for full or half-rate channels somewhere.
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Implement sending MDCX on the newly allocated channel and send
the data to the same destination as the currently connected one.
This way the receiver can implement RTP RFC Appendix A.1 and
deal with the new source.
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For the LCR rtp-bridge audio should directly flow to the
remote system. In contrast to the original patch audio
will now flow directly from the BTS to the remote system.
This assumes that BTS and the remote system are in the
same network segment and can directly communicate.
There are various limitations in the first iteration of
the implementation:
We could (and in the future) should delay the assignment
but currently we are forced to pick the channel and move
it to the audio state. In case we are located on a SDCCH
we always need to change but if we are on a TCH we could
send the ipa.CRCX and change the audio state a lot later.
The net effect is that the audio codec selection needs to
be done in the NITB code and not in the system connected
to it.
This only works with ip based systems. For E1 systems one
could still use the RTP socket or even try to move this
out of the process.
There is no code for handover handling and it relies on
the remote system dealing with the SSRC change of the
system.
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This adds the protocol definition for the RTP bridge extension
of Andreas Eversberg and bumps the protocol version.
I added the missing mncc mappings from value to string.
[ 5cf8fb10ea3addcae74d37f4dbf1c1be664df53e protocol extension
5dac90de38990b188f499c602bf18a4f232070e8 payload extension]
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When using multiple interfaces on a system one can now configure
which will be served for the BTS ports and which will be served
for the network. The direct usage of source_addr is now only to
initialize the MGCP receiving port itself.
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Make it possible to bind the call-agent to a specific IP address
and the network and bts end to different ip addresses. Begin by
clarifying which source ip address we want to have.
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Use the existing ulaw encode/decode to support PCMU as well.
The MERA VoIP switch has some severe issues with the GSM codec
and it appears easier to enable transcoding for it.
The mera switch doesn't appear to cope with codec change
between a SIP 180 trying and the 200 ok connection result.
Inserting the codec is touching too many places. Ideally we
should have the transcoding function as pointer in the struct
as well but the arguments differ.. so it is not a direct way
forward.
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Iridium is a satellite network which operates a GPRS-like that allows you to
get speeds up to 128kbit/s. However, it takes from 5 to 6 secs to get the
bandwidth allocated, so the conversation is garbled during the time.
This patch uses the new dummy padding support in libosmo-netif that is
controlled through the osmux osmux_xfrm_input_open_circuit().
This includes a new VTY option for osmux.
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We have a lot of legacy that I am afraid to break. We have
everything in place to make a good codec selection (e.g. if
we can avoid transcoding, pick the one with best quality or
the lowest speed). Right now I have a specific case where
from all options I want to pick GSM. Guard the codec compat
check behind the disallow transcoding option to make sure
to not break legacy application.
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The SDP file handling will get more complicated in terms of
codec selection so let's remove it from the protocol handling
before we start blowing it up in size.
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This struct was removed when we switched to strtok_r for
parsing the data. Remove the left-over.
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The parsing code assumed that there will be a single payload
type and this assumption is clearly wrong. Forward all of the
payload types. The code is still only extracting the first
type from the list. The variable name has been renamed to
reflect this.
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In case foreign simcards are used we can not do authentication
and ciphering. In case a TMSI is re-used too early and we do
page using TMSI we can't know which of the two MS is responding
to us. We could change the "secure channel" routine to ask for
the IMSI and only then stop the paging.
As we don't have ciphering there is not much use in using the
TMSI. Add a mode "no assign-tmsi" that will not assign the TMSI
during LU. Now CM Service Request and Paging Response will
work using the IMSI. There can't be a clash with that.
[ciaby fixed the vty write to use the right name]
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in_address is not 'accidently' included by FreeBSD when we include
the osmocom/core/select.h header file. We need to include a bit
more.
In file included from mgcp_protocol.c:38:
../../include/openbsc/mgcp_internal.h:134:21: error: field has incomplete type 'struct sockaddr_in'
struct sockaddr_in forward;
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Struct osmo_msc_data contains int core_ncc, which is actually the
MNC part of the PLMN, not to be confused with the Network Colour
Code.
The following patch renames this field for clarity and consistency
with the standards.
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We don't need to consume all the entropy of the kernel but can
use libcrypto (OpenSSL) to generate random data. It is not clear
if we need to call RAND_load_file but I think we can assume that
our Unices have a /dev/urandom.
This takes less CPU time, provides good enough entropy (in theory)
and leaves some in the kernel entropy pool.
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We are using the token to find the right bsc_config and
then we can use the last_rand of the bsc_connection to
calculate the expected result and try to compare it with
a time constant(???) memcmp.
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Check if the NAT has sent 16 bytes of RAND and if a key
has been configured in the system and then generate a
result using milenage. The milenage res will be sent and
noth the four byte GSM SRES derivation.
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Generate 16 byte of random data to be used for A3A8 by
the BSC in the response. We can't know which BSC it is
at this point and I don't want to send another message
once the token has been received so always send the data
with an undefined code. The old BSCs don't parse the
message and will happily ignore the RAND.
/dev/urandom can give short reads on Linux so loop
around it until the bytes have been read from the kernel.
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Instead of doing open/read/close all the time, open the
FD in the beginning and keep it open. To scare me even
more I have seen /dev/urandom actually providing a short
read and then blocking but it seems to be the best way
to get the random byes we need for authentication.
So one should/could run the cheap random generator on
the system (e.g. haveged) or deal with the NAT process
to block.
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Unfortunately the basic structure of the response is broken.
There is a two byte length followed by data. The concept of
a 'tag' happens to be the first byte of the data.
This means we want to write strlen of the token, then we
want to write the NUL and then we need to account for the
tag in front.
Introduce a flag if the new or old format should be used.
This will allow to have new BSCs talk to old NATs without
an additional change. In the long run we can clean that up.
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In the upcoming authentication improvements it is nice to
separate the finding of the config from the post-allow
handling of it.
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The code to do that doesn't belong to the control interface, so
abstract it out to a separate function gsm_bts_set_system_infos().
[hfreyther: Fix the coding style...]
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If no server is specified the default list will be used. This
allows to separate the servers for the local network and GRX
from each other.
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For real networks we need to check if the requested APN string
is allowed and then resolve the GGSN address through DNS. There
are countries with two or three digit MNCs and one could either
try to keep a list of countries that have two/three digits or
just try both of them. I have opted for the later for the ease
of the implementation.
C-Ares doesn't allow to cancel a request so we will need to
have the MMCTX and the Lookup have different lifetimes. We simply
set ->mmctx to NULL in case the MMCTX dies more early.
The selected and verified apn_str will be copied into the out
parameter. In case no static APN/GGSN config is present and the
dynamic mode is enabled a request will be made.
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c-ares is an asynchronous DNS resolver and we need it to
resolve the GGSN address. This is integrating the library
into our infrastructure. We will create and maintain a list
of registered FDs (c-ares is currently only using one of
them) and (re-)schedule the timer after events occurred.
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Include the hlr-Number of the subscriber in the CDR. This is useful
for debugging and understanding which equipment was used during the
test. In contrast to the MSISDN the '+' is emitted as the number
must be in international format already.
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Copy the hlr-Number into the sgsn_data and use it during
the purgeMS. There is no unit test that looks at the data
we send so I manually verified this by looking at the output.
Below is the output of the test that purges the subscriber.
<000f> gprs_subscriber.c:170 SUBSCR(123456789012345) Sending GSUP, will send: 0c 01 08 21 43 65 87 09 21 43 f5 09 07 91 83 61 26 31 23 f3
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Implement it similar to the msisdn_enc/msisdn_enc_len and
extend the testcase to include it as well.
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The charging_id is provided by the GGSN. Copy it into the CDR
part of the data structure so it will remain present until after
the pdp context has been deleted.
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This is consuming the new signals and allows to install several
different CDR/observing/event/audit modules in the future. For
getting the bytes in/out the code would have had to undo what the
rate counter is doing and at the same time adding a "total" to
the ratecounter didn't look like a good idea, the same went for
making it a plain counter.
Begin writing the values one by one and open/closing a new FILE
for every log messages. This is not efficient but easily deals
with external truncation/rotation of the file (no fstat for and
checking the links and size). As usual we will wait and see if
this is an issue.
Add some new members to our PDP context structure to see what it
is about.
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All calls should and do go through the
sgsn_mm_ctx_cleanup_free function.
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sgsn_create_pdp_ctx should use the subscribed QoS. When selecting
the PDP context we inject the QoS to be used into the TLV structure
and use it during the request. Assume a "qos-Subscribed" structure
only with three bytes and prepend the Allocation/Retention policy
to the request.
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The MSISDN should be present for "security" reasons in the first
activation of a PDP context. Take the encoded MSISDN, store it for
future use and then put it into the PDP activation request.
The MM Context contains a field for a decoded MSISDN already. As
we need to forward the data to the GGSN I want to avoid having to
store TON and NPI in another place. Simply store the data in the
encoded form.
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Add roundtrip test for the new QoS IE. It will be consumed in
later commits.
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Extract the new MSISDN IE from the GSUP message and verify that
it is read/written to the message.
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It is a bit arbitary to decide which one is the global
and which one is the local one. We might change it around.
I don't think we want to introduce it based on BTS.
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Remove the last occurence of NAT datastructures in the filtering
module and add the ctx to the filter request structure.
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For the BSC we will have the gsm48_hdr and don't need to
find data within SCCP. For legacy reasons we need to
initialize con_type, imsi, reject causes early on and
need to do the same in the filter method.
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