OpenBSC

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14 participants
4246 discussions

Ideas/proposals for Osmocom project submissions to NLnet or prototypefund?
by Harald Welte 20 Sep '22

20 Sep '22

Dear Osmocom community, during the past years, we've been applying for (and often receiving) grants for the development of certain parts of the Osmocom project. There are two organizations (NLnet foundation and the German prototype fund) where the next call for proposals ends in 10 days from now. So if any of you has some ideas about Osmocom related work that you would want to see implemented and for which you can argue that it fits the scope of the related funds, please feel free to bring them up here. For NLnet, the scope should fit either * https://nlnet.nl/entrust/ * https://nlnet.nl/useroperated/ For prototype fund, the scope should fit https://prototypefund.de/jetzt-bewerben-fuer-runde-13/ - specifically I think it is likely that we could fit a number of "our" topics in https://prototypefund.de/softwareinfrastruktur/ which explicitly lists implementation of communication protocols. Given that I have plenty of experience with both funds, I'd be willing to work on a concrete proposal and submit it before the deadline on Sept 30 / Oct 1. In terms of "who would do the actual work": With NLnet projects we can involve freelancers across Europe, with prototype fund probably only in Germany. In the worst case, if you propose some idea but have no time to work on such a project yourself, or you are not eligible due to the restriction to EU/DE, sysmocom could carry out the implementation. Let me know if anyone of you has some ideas as to what kind of osmocom related topics might be suitable candidates. Thanks, Harald -- - Harald Welte <laforge(a)osmocom.org> http://laforge.gnumonks.org/ ============================================================================ "Privacy in residential applications is a desirable marketing option." (ETSI EN 300 175-7 Ch. A6)

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mscgen
by Neels Hofmeyr 14 Sep '22

14 Sep '22

RFC about mscgen mscgen is our tool of choice to generate ladder diagrams from the "msc" text input format. We use it in user manuals and on our redmine wiki, and there are also various .msc files in doc/ subdirs, as assistance for development. I've written a lot of ladder diagrams, for documentation as well as figuring out how to implement things. I've developed a love/hate relationship, and recently I found a bug in mscgen. Things have come up that I'd like your opinion on: * ladder_to_msc.py Writing .msc files by hand quickly started to annoy me: - Having to write '[label="..."]' all the time. - Multiline labels (e.g. '(note)' block) require "foo\nbar\nbaz" - input symbols for arrows are weird / non-intuitive - cannot use 'msc' as entity, weirdly conflicts with the outer 'msc { ... }', which is annoying for GSM diagrams featuring an MSC. So quite some while ago I wrote a python script that reads my personal favorite format in which i enjoy writing ladder diagrams, and outputs plain .msc file format. I put it in libosmocore/contrib/ on a branch, have been using it ever since, but i never submitted it for merging, because it seemed like a personal preference thing, not worth complicating the doc tool chain for. So far I was, if at all, committing only the resulting .msc to git. But that creates a situation where only I have the *original* source in .ladder format, so (hypothetically) collaborating on ladder diagrams becomes convoluted. Here is an example of my favorite ".ladder" format: ------ {hscale=1} upf = User Plane function cpf = Control Plane function upf <- cpf PFCP Association Setup Request CP function Node Id, features upf -> cpf PFCP Association Setup Response UP function Node Id, features upf <> cpf associated upf () cpf start Heartbeat checking ... ------ The equivalent .msc produced by ladder_to_msc.py: ------ msc { hscale="1"; upf[label="User Plane function"],cpf[label="Control Plane function"]; upf <= cpf [label="PFCP Association Setup Request\nCP function Node Id, features"]; upf => cpf [label="PFCP Association Setup Response\nUP function Node Id, features"]; upf abox cpf [label="associated"]; upf rbox cpf [label="start Heartbeat checking"]; ...; ------ I'd like to hear some opinions, if you have any, on whether anyone else likes my new ladder format, and how to deal with .ladder source files. Would it make sense to merge ladder_to_msc.py to libosmocore, install it via 'make install', and add the original .ladder files as well as Makefile rules to convert .ladder to .msc to .png where ever i wrote .ladder diagrams? * mscgen bitrot is likely Recently I've found a segfault in mscgen. I found a fix and tried to submit it. That's when I discovered that: - the original project on code.google.com is dead and gone - the mail addresses of the original author result in bounces - random people put mscgen's trunk on github, exported from google code, no activity is apparent I did reach the Debian maintainer of mscgen by mail. He doesn't have any way to contact the original author, either. It seems there could be a patch in the Debian package, at best. I wonder if it would make sense to officially create an mscgen "fork" on Osmocom, in an effort to save the tool from bitrot and extinction, and offer our mscgen as the new upstream for debian's mscgen packaging. I do have a number of things i'd like mscgen to be able to do, like left-adjust text in "note" blocks or improve the resolution of produced PNG images; I might also implement support for my .ladder format directly in mscgen. I am vague on what my opinion is between the advantages and disadvantages, and it's all only semi important. I just know that I always end up writing .ladder instead of .msc, and that we will always need ladder diagrams. Do you have any opinions on these things? Thanks! pointers: ladder_to_msc.py: libosmocore branch neels/ladder https://cgit.osmocom.org/libosmocore/commit/?h=neels/ladder&id=073ad74435ec… My most recent ladder chart: https://cgit.osmocom.org/osmo-bsc/commit/?h=neels/codec&id=01f370c1b365305f… -- - Neels Hofmeyr <nhofmeyr(a)sysmocom.de> http://www.sysmocom.de/ ======================================================================= * sysmocom - systems for mobile communications GmbH * Alt-Moabit 93 * 10559 Berlin, Germany * Sitz / Registered office: Berlin, HRB 134158 B * Geschäftsführer / Managing Directors: Harald Welte

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Osmocom GUI support
by Ahnaf Tahmid 02 Sep '22

02 Sep '22

Dear OpenBSC team, I am Ahnaf Tahmid Chowdhury, currently building Osmocom graphical user interface. I have created a subscriber page where one can add/remove/update subscriber. The repo is available at: https://github.com/ahnaf-tahmid-chowdhury/osmo-gui Please let me know your thoughts about the application. Thank you

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two distinct SCCP sources?
by Neels Hofmeyr 29 Aug '22

29 Aug '22

I'm confused by what is happening in the pcap linked below: According to wireshark, the SCCP message seems to come from *both* PC=4 and PC=185. A column configured as sccp.calling.pc shows Point-Code 4. A column configured as "Source Address" shows Point-Code 185. Is it even possible that the in-band Calling Address differs from whatever wireshark detects as the SCCP Source Addres? Can anyone explain where the 185 is coming from? apparently not from sccp.calling.pc. The pcap and a screenshot showing the two point-codes and the column config are here: http://people.osmocom.org/neels/Eew7we0I/ Context: The original RESET message (not part of the pcap) was sent to called.pc=4, but I have SCCP routing set up to forward PC=4 to the MSC that sees itself as PC=185. I am expecting to see a RESET-ACK sent from PC=185. IIUC The 4 should have evaporated when the MSC parsed the RESET message. But apparently when answering, it takes the "called.pc" and puts it in the response as "calling.pc" (4 is nowhere in the MSC's confguration, it MUST have taken the "4" from the received SCCP message). So what I am seeing in wireshark is eerily correct: it is the MSC that has "185" in its cfg as local address sending the ACK, and it is responding to a request that was originally sent to "4", and which osmo-stp routed to "185". The response shows calling.pc=4. But how can wireshark know that it was really "185" sending, when no such information is in the SCCP layer of that message? What am I missing? if anyone knows, thanks in advance... ~N

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New repository - libosmo-gprs.git
by Vadim Yanitskiy 18 Aug '22

18 Aug '22

Hi all, as some of you may know, I am working on the MS side implementation of GPRS protocol stack. The end goal is to have something similar to srsUE (part of srsRAN, former srsLTE), but for GPRS. In order to reduce code duplication it was decided to move some stuff from both osmo-pcu.git and osmo-sgsn.git into shared libraries, so that we could re-use existing code in the upcoming MS side implementation. == What is already done * Redmine: https://osmocom.org/projects/libosmo-gprs * Gerrit: https://gerrit.osmocom.org/q/libosmo-gprs * Gitea: https://gitea.osmocom.org/osmocom/libosmo-gprs * osmo-ci.git: https://gerrit.osmocom.org/c/osmo-ci/+/29019 == What is missing * GitHub: https://github.com/osmocom/libosmo-gprs * cgit: https://cgit.osmocom.org/libosmo-gprs/ ** Coverity pulls from this mirror ** Jenkins build verification uses this mirror I will need some help with both GitHub and cgit, as I don't have admin permissions. Please also let me know if I missed something. Best regards, Vadim. -- - Vadim Yanitskiy <vyanitskiy at sysmocom.de> http://www.sysmocom.de/ ======================================================================= * sysmocom - systems for mobile communications GmbH * Alt-Moabit 93 * 10559 Berlin, Germany * Sitz / Registered office: Berlin, HRB 134158 B * Geschaeftsfuehrer / Managing Director: Harald Welte

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shadysim STK installation issue
by Karsten Ohme 11 Aug '22

11 Aug '22

I tried to use shadysim (https://gitea.osmocom.org/sim-card/sim-tools) to install a STK / CAT applet on the sysmoISIM. It was not working for me, also it does not support UICC parameters and it seems that still Python 2.7 is needed. As an alternative I have added support for gpshell (https://kaoh.github.io/globalplatform/) to use SCP02 and pass the SIM/UICC parameters. See the new parameters "uiccSystemSpecParam" and "simSpecParam" for the "install" command https://github.com/kaoh/globalplatform/blob/master/gpshell/src/gpshell.1.md and inspect the https://github.com/kaoh/globalplatform/blob/master/gpshell/helloInstallSTK.… example file. If "brew" ins installed then "brew install kaoh/globalplatform/globalplatform" is sufficient for the installation. I also faced the issue that the sysmoISIM cards are only accepting SIM parameters and no UICC parameters, which is strange or even a bug since these cards are newer and are supporting the newer UICC ETSI specifications.

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Question about SDR aix-tx project
by zhang_xiangjun 20 Jul '22

20 Jul '22

Dear Sir/Madam， I'm a graduate student from a martime college , my major is not computer or communication , I'm trying to do a project about vessel's AIS tx . The equipment which I use is USRP b210 and the software is Gnuradio . After I cloned the code from http://github.com/osmocom/ais-tx/tree/master/gr-aistx and install into gnuradio , it occurs a few mistakes . I tryed to install it in Ubnutu system and dragonOS system ,both failed . Could take a look and help me to solve the problem , there is some pictures attached in this e-mail about the problem . Really thanks a lot , wish you have a nice day . Yours Sincerelly Xiangjun Zhang

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Operating multiple MSs within one instance of mobile
by James Connor 13 Jul '22

13 Jul '22

Hey y'all, This is somewhat related to another post of mine titled "MSC, SIP connector, and Asterisk". I've been working on a project to simulate cell phone usage through construction of a completely virtual GSM network. To that end, I've been planning to use the mobile application packaged with OsmocomBB to simulate the actual MSs. To make this easier, I've been using a single instance of mobile configured with several MSs in the configuration file (alongside separate instances of virtphy). This had been working for a while, where I was able to place calls between the MSs that were configured. However, I've recently run into an issue where the calls are rejecting each other. After digging through the source of mobile a little, it's looking increasingly like multiple MSs are not meant to be used in parallel with each other on the same instance of mobile. The main file I've been looking at is mnccms.c (https://osmocom.org/projects/baseband/repository/osmocombb/revisions/master…). It seems that there's no differentiation made between MSs when checking for any calls with/without hold status, which means that only one active call can be used by a mobile instance at any one time. In fact, based on what I've been reading, I can't figure out how I had this working in the first place. Therefore, I wanted to see if I could get a definitive answer before I invest time into redesigning certain aspects of my framework. Is it possible to use multiple MSs within a single instance of mobile, such that they can all be involved in separate calls simultaneously? Thanks for any help, it's greatly appreciated! - James

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MSC, SIP connector, and Asterisk
by James Connor 11 Jul '22

11 Jul '22

Hey all, I've got a bit of an interesting use case. I've been working to setup the OpenBSC components in order to simulate stress testing a GSM network. I've been able to get communications across the GSM working properly, with MSs able to place calls. However, part of what I'm trying to do involves routing to a simulated PSTN. I turned to sip-connector and asterisk, but my network immediately stopped working once I told the MSC to use an external MNCC. The setup involves two separate computers. The first is the BSS, which has a BSC, MGW, virtual BTS, and two MSs running on mobile and virtphy. The second is the NSS, with the MSC/VLR, HLR, MGW, and STP. The GSM is split between the two devices because of the eventual goal to increase the number of BSSs to better simulate a real-world GSM. This was all working fine, until I modified osmo-msc.cfg to include "mncc external /tmp/msc_mncc". In parallel, I setup osmo-sip-connector and ran it on the NSS through the same "/tmp/msc_mncc" socket, pointing it at Asterisk (also on the NSS). This was all done according to the instructions listed here: https://osmocom.org/projects/osmo-sip-conector/wiki/Howto. Now whenever I place my calls (which I do through mobile's "call" VTY command on the BSS), the call doesn't connect and simply gets released after 30 seconds. After doing some debugging, I've determined that Asterisk is not receiving anything from the sip-connector. Meanwhile, the sip-connector is being told to delete any connection basically as soon as it's created: <0001> mncc.c:1005 MNCC rcvd message type: MNCC_SETUP_IND <0001> mncc.c:566 Created call(5004) with MNCC leg(2147483652) IMSI(001010000000001) <0001> mncc.c:68 Starting Timer for MNCC_RTP_CREATE <0001> mncc.c:164 MNCC sent message type: MNCC_RTP_CREATE <0001> mncc.c:1005 MNCC rcvd message type: MNCC_REL_IND <0001> mncc.c:636 Rcvd MNCC_REL_IND, Cause: RESOURCE_UNAVAIL <0001> mncc.c:648 leg(2147483652) was released. <0002> call.c:90 call(5004) released. The MSC has the following as part of its output when a call is placed: ... <0020> osmo_ss7.c:1933 0: asp-OsmoMSC-asp: xua_cli_read_cb(): sctp_recvmsg() returned 48 (flags=0x80) <0023> m3ua.c:714 0: asp-OsmoMSC-asp: Received M3UA Message (XFER:DATA) <0023> m3ua.c:543 0: asp-OsmoMSC-asp: m3ua_rx_xfer <0023> m3ua.c:566 0: asp-OsmoMSC-asp: m3ua_rx_xfer(): M3UA data header: opc=2233=1.23.1 dpc=185=0.23.1 <0020> osmo_ss7_hmrt.c:276 m3ua_hmdc_rx_from_l2(): found dpc=185=0.23.1 as local <0020> sccp_scrc.c:472 scrc_rx_mtp_xfer_ind_xua: HDR=(CO:CODT,V=0,LEN=0), PART(T=Destination Reference,L=4,D=00000007), PART(T=Segmentation,L=4,D=00000000), PART(T=Data,L=6,D=000422040120) <0021> sccp_scoc.c:1664 Received CO:CODT for local reference 7 <0021> sccp_scoc.c:1698 SCCP-SCOC(7)[0x555adb5ddf40]{ACTIVE}: Received Event RCOC-DT1.ind <0021> sccp_user.c:175 Delivering N-DATA.indication to SCCP User 'OsmoMSC-A' <0011> sccp_ran.c:108 (GERAN-A-7) sccp_ran_sap_up(N-DATA.indication) <0003> ran_peer.c:591 ran_peer(GERAN-A:RI-SSN_PC:PC-1-23-1:SSN-BSSAP)[0x555adb5b1010]{READY}: Received Event RAN_PEER_EV_MSG_UP_CO <0007> ran_peer.c:407 msc_i(IMSI-001010000000001:MSISDN-1234567891:TMSI-0x9B96EF97:GERAN-A-7:CM_SERVICE_REQ)[0x555adb5d97e0]{READY}: Received Event MSC_EV_FROM_RAN_UP_L2 <0011> ran_msg_a.c:790 msc_i(IMSI-001010000000001:MSISDN-1234567891:TMSI-0x9B96EF97:GERAN-A-7:CM_SERVICE_REQ)[0x555adb5d97e0]{READY}: RAN decode: BSSMAP: CLEAR REQUEST <0007> msc_i.c:85 msc_a(IMSI-001010000000001:MSISDN-1234567891:TMSI-0x9B96EF97:GERAN-A-7:CM_SERVICE_REQ)[0x555adb5daa50]{MSC_A_ST_COMMUNICATING}: Received Event MSC_A_EV_FROM_I_PROCESS_ACCESS_SIGNALLING_REQUEST <000b> msc_a.c:196 msc_a(IMSI-001010000000001:MSISDN-1234567891:TMSI-0x9B96EF97:GERAN-A-7:CM_SERVICE_REQ)[0x555adb5daa50]{MSC_A_ST_COMMUNICATING}: + msc_a_ran_dec: now used by 2 (cc,msc_a_ran_dec) <0011> ran_msg_a.c:790 msc_a(IMSI-001010000000001:MSISDN-1234567891:TMSI-0x9B96EF97:GERAN-A-7:CM_SERVICE_REQ)[0x555adb5daa50]{MSC_A_ST_COMMUNICATING}: RAN decode: BSSMAP: CLEAR REQUEST <0011> msc_a.c:1612 msc_a(IMSI-001010000000001:MSISDN-1234567891:TMSI-0x9B96EF97:GERAN-A-7:CM_SERVICE_REQ)[0x555adb5daa50]{MSC_A_ST_COMMUNICATING}: RAN decode: BSSMAP Clear Request <0007> msc_a.c:1433 msc_a(IMSI-001010000000001:MSISDN-1234567891:TMSI-0x9B96EF97:GERAN-A-7:CM_SERVICE_REQ)[0x555adb5daa50]{MSC_A_ST_COMMUNICATING}: Received Event MSC_A_EV_MO_CLOSE <0007> msc_a.c:733 msc_a(IMSI-001010000000001:MSISDN-1234567891:TMSI-0x9B96EF97:GERAN-A-7:CM_SERVICE_REQ)[0x555adb5daa50]{MSC_A_ST_COMMUNICATING}: State change to MSC_A_ST_RELEASING (X2, 30s) <0011> msc_a.c:769 msc_a(IMSI-001010000000001:MSISDN-1234567891:TMSI-0x9B96EF97:GERAN-A-7:CM_SERVICE_REQ)[0x555adb5daa50]{MSC_A_ST_RELEASING}: Releasing: msc_a use is 2 (cc,msc_a_ran_dec) <000b> msc_a.c:773 VLR subscr IMSI-001010000000001:MSISDN-1234567891:TMSI-0x9B96EF97 + msc_a_fsm_releasing_onenter: now used by 4 (attached,active-conn,CC,msc_a_fsm_releasing_onenter) <000b> vlr.c:309 VLR subscr IMSI-001010000000001:MSISDN-1234567891:TMSI-0x9B96EF97 + vlr_subscr_cancel_attach_fsm: now used by 5 (attached,active-conn,CC,msc_a_fsm_releasing_onenter,vlr_subscr_cancel_attach_fsm) <000b> vlr.c:314 VLR subscr IMSI-001010000000001:MSISDN-1234567891:TMSI-0x9B96EF97 - vlr_subscr_cancel_attach_fsm: now used by 4 (attached,active-conn,CC,msc_a_fsm_releasing_onenter) <0001> transaction.c:230 trans(CC:INITIATED IMSI-001010000000001:MSISDN-1234567891:TMSI-0x9B96EF97:GERAN-A-7:CM_SERVICE_REQ callref-0x80000003 tid-8) Freeing transaction <0005> gsm_04_08_cc.c:237 trans(CC:INITIATED IMSI-001010000000001:MSISDN-1234567891:TMSI-0x9B96EF97:GERAN-A-7:CM_SERVICE_REQ callref-0x80000003 tid-8) tx MNCC_REL_IND ... Any thoughts as to why this one change to my GSM is suddenly stopping it from functioning properly? Thanks, any help is greatly appreciated! - James P.S. I didn't want to include my config files for fear of making this message absurdly long, but let me know if you want to see any and I'll gladly post them.

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Interfacing Osmocom GSM network to USA PSTN
by Mychaela Falconia 02 Jul '22

02 Jul '22

Hello Osmocom community, I have previously alluded in other threads here that I am seeking to interface my Osmocom-based GSM network to USA PSTN. In this post I shall provide details as to exactly what I am doing and how I go about it. I am currently experimenting with using bulkvs.com as my USA PSTN connectivity provider, chosen on the basis of cost economy: it is incredibly, mind-boggling cheap - the monthly "keepalive" cost is only 6 cents (yes, *cents*) per rented phone number! With phone numbers this cheap, I don't have to do any kind of "number economy" options: no need for "phone number NAT" (a single external phone number where a PBX answers, requiring callers to then dial an internal extension number), and no need for a mixed scheme where only one or two test SIMs get real outside numbers while all others get only fake internal numbers - instead I can give a real valid NANP phone number to every test SIM on my test/R&D GSM network. For context, my test GSM network is envisioned as serving two purposes: one is for development and testing of GSM handsets - real phone numbers are less important here - and the other purpose is demonstration to gathered audiences (American patriots and in-person freedom lovers who aren't afraid of some silly virus) for "oooh and ahhh" - and for this second purpose, having real phone numbers for every test phone will be a big boost. The interface to USA PSTN provided by bulkvs.com and other similar low-cost services is a SIP trunk, hence what I need to implement is a form of GSM to SIP gateway. But here is the part where I deviate from what appears to be the established canon in Osmocom-based community cellular networks: I don't want to run a heavyweight PBX like Asterisk or FreeSwitch, instead I want to run only a fairly thin, lightweight layer of extra sw between my OsmoMSC instance (MNCC interface) and the public-Internet SIP interface provided by bulkvs, with the needed thin sw layer put together by me, combining newly written code specific to my use case with bits of code borrowed from elsewhere. Here is the code I have written / put together so far: https://www.freecalypso.org/hg/themwi-system-sw/ The README file at the top level of the code repository explains what is there and how it works, in basic terms. One of the major pieces I will need to implement is an RTP bridge MGW that will transcode between GSM codecs (FR1, EFR, AMR) on the GSM side and G.711 (PCMU or PCMA) on the PSTN-via-SIP side. PSTN-via-SIP connectivity providers such as bulkvs don't support any GSM codecs (not one of them), instead they only support G.711 and G.729. I won't be doing G.729 in ThemWi (transcoding from one lossy vocoder to another seems like a terrible idea), hence I will run with G.711 on my PSTN-over-Internet interface and I will need to implement a transcoder between GSM codec family and G.711. I reason that what I need to implement is probably quite similar to the transcoding function in classic GSM E1 TRAUs, except that in the RTP bridge environment I also have to deal with the added complexities of packet loss, reordering and silence suppression, unpleasantries of the packet world that didn't exist in the olden days of TDM luxury. I don't know whether or not a transcoding RTP bridge MGW is already implemented somewhere in the bowels of either Asterisk or FreeSwitch - the size and complexity of both "major sw" choices is quite intimidating and off-putting. If someone who happens to know the answer gives me a pointer, it will certainly be very appreciated, but if not, then some time later I will undertake the big effort to dig in both projects and look for this transcoder. If the needed transcoder implementation already exists somewhere, then I will need to lift it out of whichever project and integrate it into themwi-system-sw - and if there is no existing published-source implementation, then I will be the first to produce one. M~

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