OpenBSC February 2023

openbsc@lists.osmocom.org

13 participants
16 discussions

by bryan coxwell

Hi all, Just wanted to share an issue and a quick workaround I found for it in case anyone else has the same problem. I believe a cmd2 update is causing pySim-shell to fail. After installing it on a fresh install of Ubuntu Server 20.04 and getting the following error when I run "python3 pySim-shell -p0": >Using PC/SC reader interface >Autodetected card type: sysmoUSIM-SJS1 >AIDs on card: > USIM: a0000000871002ffffffff8907090000 >Traceback (most recent call last): > File "pySim-shell.py", line 512, in <module> > app = PysimApp(card, rs, opts.script) > File "pySim-shell.py", line 59, in __init__ > super().__init__(persistent_history_file='~/.pysim_shell_history', allow_cli_args=False, use_ipython=True, auto_load_commands=False, command_sets=basic_commands, >startup_script=script) >TypeError: __init__() got an unexpected keyword argument 'use_ipython' If you run into this you can fix it by uninstalling cmd2 and reinstalling cmd2 with "pip3 install cmd2==1.5". Best, Bryan

10 hours, 42 minutes

Re: Help with sccp library

by mosbah abdelkader

> I don't understand. This callback will be called with data you need to write > to the network. In case of MTP Level3 you will need to wrap that around the > msgb you got. I means: is the interaction with mtp3 layer implemented (is sending sccp data by mtp3 implemented by the library?)? Also, what about the reception of data from mtp3 layer. is that implemented in the sccp lib. I am asking these questions because I see the code of mtp3 in the lib but no significant call is present in the sccp part of the lib. Thank you for your help.

1 month

dynamic traffic channels

by Neels Hofmeyr

On Tue, Jun 28, 2016 at 10:05:28AM +0200, Harald Welte wrote: > [translated from german] > is it certain that we switch a channel to PDCH only when > gprs mode != none? A TS can be GSM_PCHAN_TCH_F_PDCH; those are the only ones for which we send a PDCH ACT message. We send a PDCH ACT message - during init (CHANNEL OML's state changed to enabled -> send PDCH ACT), - and upon channel release ack when pchan == GSM_PCHAN_TCH_F_PDCH. So the question is, when we receive a channel release ack, could that be the PDCH release and we switch PDCH right back on by accident? No, because we only receive a chan rel ack when the *TCH/F* is being released. That is because the PDCH release is initiated "internally" from the PDCH DEACT, and thus this condition in common/rsl.c rsl_tx_rf_rel_ack() catches on, which existed before dyn PDCH: if (lchan->rel_act_kind != LCHAN_REL_ACT_RSL) { LOGP(DRSL, LOGL_NOTICE, "%s not sending REL ACK\n", gsm_lchan_name(lchan)); return 0; } In rsl_rx_rf_chan_rel() the rel_act_kind is set to LCHAN_REL_ACT_RSL, but not in the rsl_rx_dyn_pdch(). This is analogous to the ip.access way -- the ip.access nanobts replies to a PDCH DEACT with a PDCH DEACT ACK and doesn't send a separate channel release ack. Maybe we could set rel_act_kind to some new LCHAN_REL_ACT_IPAC_DYN_PDCH for clarity? (But we shouldn't actually send a release ack, to stay compatible.) Even though it works as-is, we should indeed add another flag check: - We do check the flags that no ACT/DEACT is already pending; - And we do send a PDCH DEACT only if ts->flags & TS_F_PDCH_ACTIVE; - But we would send a PDCH ACT despite ts->flags & TS_F_PDCH_ACTIVE. This should never happen, but it would make sense to ensure that. ~Neels

1 month

libosmocore/BSC/MSC: MO voice calls rejected from Siemens S11E (patch included)

by Lennart Müller

Hi, I had a problem placing MO GSM calls from a Siemens S11E: The calls were dropped immediately; Osmo-MSC reports "Cannot compose Channel Type from bearer capabilities" After investigating the SETUP request from the S11E, the phone does not use octet 3a (no extension bit set in IE 3). Wireshark decodes the radio channel requirement as "Full rate support only MS/fullrate speech version 1 supported", so I added a condition to the gsm48_ie.c function of libosmocore to include at least GSM FR in the list of available speech_ver in case octet 3 has no extension. Attached to this message are the Abis-IP PCAP traces of MO calls, and the patch for gsm48_ie.c. Regards, Lennart

1 year, 4 months

Problem while running NIB on Open Cellular Base sation

by Brackley Casinga

Hi, My name is Brackley Cassinga Form DRC, we run a community network called pamoja net where we offer gsm services using osmocom open source software and OC Base station. Recently I have tried to install another base station as the same installed but I could not find any resource guiding through all the steps to take to run NIB on a base station. I'm currently running Ubuntu and I will appreciate if you could guide me on the installation of BSC,hlr,MSC , in order to run a basic gsm network. Thank you. Regards -- *Ir Brackley heshima Casinga **Pacifique* *CEO and Founder of kwanzatechnologie* KwanzaTechnologies ,GlobalElectronics +243977265291 | +243977265291 | Pcassinga(a)gmail.com/ brackley(a)ensemblepourladifference.org www.kwantechnologies.jimdosite.com <http://www.kwantechnologies.com/> | Skype: Brackley cassinga <https://webapp.wisestamp.com/#> Av Semliki N 43

1 year, 4 months

GSM-EFR SID support in libosmo-abis

by SADANAND VISHWAS

Hello, I was going through libosmo-abis codebase to understand how it handles “Rtp to Trau” and vice-versa GSM-EFR conversions. I was able to find “RTP to Trau” and “Trau to Rtp” conversion code of EFR speech frames, but could not find anything related to EFR SID (silence) frames. can you please guide me if above is already present in the library or how can we implement it? Thanks, Sadanand Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows

2 years, 2 months

Play RTP packets payload obtained from the mobile phone in a call service

by morteza ali Ahmadi

Hi, I have developed a UMTS core network using a home nodeB (HNB) which the mobile phones (or UEs) can register on my network and get the required services such as call service. I have implemented all the required procedures for call service and I can establish a successful call for my connected UEs. In one of the most important procedures, i.e **Call Proceeding**, I have identified the coding of speech transferred between UEs and core. Here is my coding options (in wireshark): GSM A-I/F DTAP - Call Proceeding Protocol Discriminator: Call Control; call related SS messages (3) .... 0011 = Protocol discriminator: Call Control; call related SS messages (0x3) 1... .... = TI flag: allocated by receiver .000 .... = TIO: 0 00.. .... = Sequence number: 0 ..00 0010 = DTAP Call Control Message Type: Call Proceeding (0x02) Bearer Capability 1 - (Spare) Element ID: 0x04 Length: 6 Octet 3 0... .... = Extension: Extended .11. .... = Radio channel requirement: Spare ...0 .... = Coding standard: GSM standardized coding .... 0... = Transfer mode: circuit .... .000 = Information transfer capability: Speech (0x0) Octets 3a - Speech Versions 0... .... = Extension: Extended .0.. .... = Coding: octet used for extension of information transfer capability ..00 .... = Spare bit(s): 0 .... 0010 = Speech version indication: GSM full rate speech version 2(GSM EFR) (0x2) 0... .... = Extension: Extended .0.. .... = Coding: octet used for extension of information transfer capability ..00 .... = Spare bit(s): 0 .... 1000 = Speech version indication: GSM full rate speech version 5(FR AMR-WB) (0x8) 0... .... = Extension: Extended .0.. .... = Coding: octet used for extension of information transfer capability ..00 .... = Spare bit(s): 0 .... 0100 = Speech version indication: GSM full rate speech version 3(FR AMR) (0x4) 0... .... = Extension: Extended .0.. .... = Coding: octet used for extension of information transfer capability ..00 .... = Spare bit(s): 0 .... 0101 = Speech version indication: GSM half rate speech version 3(HR AMR) (0x5) 1... .... = Extension: No Extension .0.. .... = Coding: octet used for extension of information transfer capability ..00 .... = Spare bit(s): 0 .... 0001 = Speech version indication: GSM half rate speech version 1(GSM HR) (0x1) So I can see the RTP packets transferred between UE and core. An instance is mentioned here (in wireshark): Real-Time Transport Protocol [Stream setup by RANAP (frame 2950)] 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 0... .... = Marker: False Payload type: DynamicRTP-Type-96 (96) Sequence number: 56611 [Extended sequence number: 56611] Timestamp: 424448575 Synchronization Source identifier: 0x5c260101 (1545994497) RFC 2198: Redundant Audio Data Header 1: PT=ITU-T G.728 0... .... = Follow: Not set .000 1111 = Payload type: ITU-T G.728 (15) Payload: 0028ba44776b3eee7a050039cdaa521cc20ac08d2bcf1818… I have aggregated all RTP packets payload. How can I convert the aggregated bytes to a hearable audio? -- *When there is much light, The shadow is deep...*

2 years, 3 months

Extract and parsing TCAP + CAP parts from SCCP messages

by Hadi Rezaee

Hello there, I was wondering if there is any snippet or sample code for extracting TCAP and then CAP parts from SCCP messages. So, What I've done so far was to connect to SG system as a client and called to 'osmo_sccp_user_bind' function to catch SCCP message: osmo_sccp_user_bind(sccp, 0, handle_sccp_msg, NULL); . . . int handle_sccp_message(struct osmo_prim_hdr *oph, void *ctx) { struct osmo_scu_prim *scu_prim = (struct osmo_scu_prim *)oph; struct osmo_sccp_user *scu = ctx; struct gsm_subscriber_connection *conn; int rc = 0; switch (OSMO_PRIM_HDR(&scu_prim->oph)) { case OSMO_PRIM(OSMO_SCU_PRIM_N_UNITDATA, PRIM_OP_INDICATION): // TCAP parsing ?! break; default: break; } return rc; } Thanks

2 years, 4 months

A new GSM codec libraries and utilities package

by Mychaela Falconia

Hello GSM community, I am pleased to announce the first release of Themyscira GSM codec libraries and utilities package: ftp://ftp.freecalypso.org/pub/GSM/codecs/gsm-codec-lib-r1.tar.bz2 The two principal libraries contained in this package, libgsmefr and libgsmfrp, continue the tradition that was started back in 1990s with the first release of libgsm, a Unix-oriented, C-library-embodied, Free Software implementation of GSM 06.10 speech transcoder. The original libgsm was written and released long before anyone in the hacker community even dreamed of running their own GSM network or building their own GSM MS, but it is now used somewhere deep under the hood (under lots of higher layers) by every GSM network-side implementation that includes an RTP transcoding gateway to other voice networks, activated whenever a GSM voice call is connected using good old FR codec rather than AMR, and is likewise used for the same purpose by anyone seeking to achieve working voice in an SDR-based GSM MS implementation. The present gsm-codec-lib release - a public works project of the Women's Republic of Themyscira - continues in the spirit of libgsm by offering two more similar libraries: 1) libgsmefr is a librified (turned into a library) version of the EFR reference code from ETSI, with all state variables converted from global vars into proper state structures, and with additional code clean-up to make it function as a speech encoder and decoder library on the same principle as libgsm, but for EFR codec instead of FR1. All beyond-speech functions of the original EFR reference code (VAD, DTX, comfort noise, error concealment) are included in libgsmefr, allowing this library to be used to implement a *proper* speech transcoder for GSM-EFR voice service. I have previously noted that the general trend in the FOSS GSM community is to use the popular AMR library libopencore-amrnb not only for AMR, but also for EFR. However, this approach is incorrect: the SID and comfort noise mechanism of EFR is different from AMR, and if you feed the uplink of a GSM-EFR call, containing SIDs and BFI skipped frames, to libopencore-amrnb decoder, there will be strange noises in the output in places where the GSM MS sent SID. Decoding EFR from a GSM call uplink requires a proper EFR library, not AMR, and I wasn't able to find any workable pre-existing implementation prior to producing Themyscira libgsmefr. 2) libgsmfrp is a preprocessor intended to be called prior to gsm_decode() from libgsm, implementing Rx DTX handler functions of GSM 06.11, 06.12 and 06.31. This preprocessor is required for anyone seeking to implement GSM FR service in the traditional way depicted in Figure 1 in 3GPP TS 46.001, as opposed to the "cut corners" way that seems to be favored in the land of Osmocom. When used as part of GSM network deployment, as opposed to lab experimentation on the mobile side of GSM, the two just-released Themyscira libraries work best together with this patch to osmo-bts: https://www.freecalypso.org/hg/themwi-system-sw/file/tip/osmo-patches/osmo-… implementing this extension to RTP transport format: https://www.freecalypso.org/hg/themwi-system-sw/file/tip/doc/RTP-BFI-extens… The "go along to get along" community appears to operate on a consensus that the RTP stream should be paused when the uplink is in DTX, or when a traffic frame was stolen for FACCH or lost to radio errors - but I disagree with that approach, and instead operate my GSM network on the principle of sending an RTP packet in *every* 20 ms slot, be it rain or shine. In the "consensus" approach the speech decoding path in the transcoding MGW never executes at all during frame stream pauses (when no RTP packets are sent), but in my "traditionalist" approach (mimicking the ways of TDM) that decoding function does get called every 20 ms, and it must produce *some* output toward PSTN in every 20 ms window, whether the MS was transmitting or not, and whether this MS output was received or lost. The Rx DTX handler becomes essential in such operation: when there is no speech traffic from the MS, libgsmfrp fills in comfort noise or previous speech frame substitution and muting as appropriate per the rules of GSM 06.31. In the case of EFR this functionality is integral to the original reference code from ETSI and thus appears in libgsmefr, but in the case of FR1 codec this functionality takes the form of a preprocessor, implemented in libgsmfrp. In any case, the two newly released libraries are intended to sit on the same "rank" in the software integration hierarchy as classic libgsm from 1990s, and higher-level projects by various parties should feel free to use them in the same manner how they currently use libgsm and libopencore-amrnb. I encourage gapk and various people's transcoding MGW implementations to take advantages of these newly available production quality GSM codec libraries. With devotion to GSM Forever, (Hasta la Victoria, Siempre,) Mother Mychaela

2 years, 4 months

SCCP RLC

by Neels Hofmeyr

I'm trying to clean up the connection management in osmo-hnbgw. For that purpose I'd like to be notified of the SCCP RLC message: I'd like to free a HNB-to-CN / RUA-to-SCCP context mapping when the SCCP local reference becomes invalid. I kind of thought I would receive a OSMO_SCU_PRIM_N_DISCONNECT PRIM_OP_CONFIRM up the user_sap, but that's not happening. I see the SCCP RLC logged like this: 20230216042453894 DLSS7 DEBUG 0: asp-asp-clnt-OsmoHNBGW: xua_cli_read_cb(): sctp_recvmsg() returned 40 (flags=0x80) (osmo_ss7.c:1906) 20230216042453894 DLM3UA DEBUG 0: asp-asp-clnt-OsmoHNBGW: Received M3UA Message (XFER:DATA) (m3ua.c:714) 20230216042453894 DLM3UA DEBUG 0: asp-asp-clnt-OsmoHNBGW: m3ua_rx_xfer (m3ua.c:543) 20230216042453894 DLM3UA DEBUG 0: asp-asp-clnt-OsmoHNBGW: m3ua_rx_xfer(): M3UA data header: opc=188=0.23.4 dpc=189=0.23.5 (m3ua.c:566) 20230216042453894 DLSS7 DEBUG m3ua_hmdc_rx_from_l2(): found dpc=189=0.23.5 as local (osmo_ss7_hmrt.c:276) 20230216042453894 DLSS7 DEBUG scrc_rx_mtp_xfer_ind_xua: HDR=(CO:RELCO,V=0,LEN=0), PART(T=Destination Reference,L=4,D=000003f0), PART(T=Source Reference,L=4,D=0073dcf0) (sccp_scrc.c:472) 20230216042453894 DLSCCP DEBUG Received CO:RELCO for local reference 1008 (sccp_scoc.c:1799) 20230216042453894 DLSCCP DEBUG SCCP-SCOC(1008)[0x612000008da0]{DISCONN_PEND}: Received Event RCOC-RELEASE_COMPLETE.ind (sccp_scoc.c:1833) 20230216042453894 DLSCCP DEBUG SCCP-SCOC(1008)[0x612000008da0]{DISCONN_PEND}: state_chg to IDLE (sccp_scoc.c:1309) 20230216042453894 DLSCCP DEBUG SCCP-SCOC(1008)[0x612000008da0]{IDLE}: Terminating (cause = OSMO_FSM_TERM_REQUEST) (sccp_scoc.c:539) 20230216042453894 DLSCCP DEBUG SCCP-SCOC(1008)[0x612000008da0]{IDLE}: Freeing instance (sccp_scoc.c:539) 20230216042453894 DLSCCP DEBUG SCCP-SCOC(1008)[0x612000008da0]{IDLE}: Deallocated (fsm.c:568) but nothing arrives at sccp_sap_up() in hnbgw_cn.c Is my idea flawed somehow, or is the prim dispatch missing in libosmo-sigtran? Thanks! ~N

2 years, 4 months

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OpenBSC February 2023