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Re: Osmo-pcu error: PCU interface version number of BTS (7) is different (9).
by Pau Espin Pedrol 14 Sep '18

14 Sep '18

Sorry forgot to reply-all and didn't send original mail to ML, re-sending now. On 9/14/18 3:25 PM, Pau Espin Pedrol wrote: > Hi Rafael, > > You may want to check at PCU_IF_VERSION in both osmo-pcu.git and > osmo-pcu.git > > It seems that you are running a recent osmo-pcu together with a quite > old osmo-bts (from at least Feb 28 2018, see osmo-bts.git > 4046e3b3dd0cffd53d8d0d1f3e1bf9d0dec83ede), and they are not protocol > compatible. > > So you need to either use a newer osmo-bts (preferred I guess) or switch > to an older osmo-pcu which uses PCU_IF_VERSION 7. > > If you are using NuRan original firmware, you either ask them to update > their stuff or your are pretty much on your own to build new images > (with non-upstream osmocom versions afaik). > > At sysmocom we maintain meta-sysmocom-bsp [1] which together with > system-images [2] and osmocom's meta-telephny [3] allows to build > firmware for both sysmobts and litecell 1.5 machines, together with SDKs > to compile software for them. > > I think generated images are not available publicly, but I guess you can > either attempt building them yourself or perhaps ask sysmocom customer > support to have access to it. > > [1] https://git.sysmocom.de/poky/meta-sysmocom-bsp/ > [2] https://git.sysmocom.de/poky/system-images/ > [3] https://git.osmocom.org/meta-telephony/ > > Regards, > Pau > -- - Pau Espin Pedrol <pespin(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 * Geschaeftsfuehrer / Managing Director: Harald Welte

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Re: Osmo-pcu error: PCU interface version number of BTS (7) is different (9).
by Pau Espin Pedrol 14 Sep '18

14 Sep '18

And again, forgot to include the list, sorry ;) On 9/14/18 3:59 PM, Pau Espin Pedrol wrote: > Hi Rafael, > > I'm not sure right now about the details of the binary firmware > installation and the exact rights you have as customer/user regarding > it, but I guess you can ask NuRan to provide them, or take them from > your current LC 1.5 image you are using if they are available there. I > think the headers are at least provided publicly in gitlab: > https://gitlab.com/nrw_noa > > Regards, > Pau -- - Pau Espin Pedrol <pespin(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 * Geschaeftsfuehrer / Managing Director: Harald Welte

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Osmo-pcu error: PCU interface version number of BTS (7) is different (9).
by Rafael Diniz 14 Sep '18

14 Sep '18

Hi all, I'm trying to test trunk osmo-pcu (using armhf osmo repo) in the NuRan LC 1.5 in the lab, but I get: 20180914124657820 DLGLOBAL <000e> telnet_interface.c:104 telnet at 127.0.0.2 4240 20180914124657821 DL1IF <0001> osmobts_sock.cpp:248 Opening OsmoPCU L1 interface to OsmoBTS 20180914124657821 DL1IF <0001> osmobts_sock.cpp:308 osmo-bts PCU socket /tmp/pcu_bts has been connected 20180914124657821 DL1IF <0001> osmobts_sock.cpp:312 Sending version 0.5.1.3-3e9c to BTS. 20180914124657821 DL1IF <0001> pcu_l1_if.cpp:113 Sending 0.5.1.3-3e9c TXT as PCU_VERSION to BTS PCU interface version number of BTS (7) is different (9). Please re-compile! Any way to make it work without changing BTS software? Thanks, Rafael Diniz

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Re: LimeSDR-USB experience with New Splits Solved!
by Vadim Yanitskiy 14 Sep '18

14 Sep '18

Hi Mychaela, > If Motorola phones running their original firmware [...] > don't work with Osmocom networks [...] it is a very grave > problem from the perspective of Freedom [...] Actually, they do. You can even find some videos on YouTube, where an original Motorola phone runing FreeCalypso firmware was tested with Osmocom based network (most likely, CalypsoBTS). The problem is that the author of this topic is using LimeSDR, that is still not 100% reliable as a PHY for OsmoTRX, and not reliable itself, because even a regular firmware update can brick the board... Moreover, AFAIK the author is not using any stable clock source, such as GPSDO. > Sysmocom has two of my FreeCalypso development boards [...] > The board you've got set up there is the one that did get > properly calibrated before being sold to Sysmocom. Maybe I am missing something, but how is this related to the original topic? > Do you still have that firmware image in the flash, or have > you erased it? Where is the RF from that board connected to - > does it go to a sysmoBTS unit? Would it be possible for someone > from Sysmocom [...] to test and see if the FCDEV3B can successfully > connect to Osmocom network using the official FreeCalypso > firmware on the board, driven via AT commands? I think you could ask this in context of the mentioned issue. > If Sysmocom folks are philosophically against doing any > tests with FreeCalypso firmware [...] Please don't get me wrong, but I am philosophically against abusing the mailing list in order to PR your production. This is not the first time I see some thread mixed with your advertisements. There are other commercial companies also using this mailing list, and let's imagine everybody would constantly PR their production here? It would just result in a mess. Even Sysmocom is not abusing that much, at least I don't see any offers to buy e.g. sysmoBTS when a general question about some BTS is asked... With best regards, Vadim Yanitskiy.

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Nokia packet Abis experiment
by Michael Andersen 14 Sep '18

14 Sep '18

Hi OpenBSC, My company is working on integrating some BTS (Nokia Flexi ESMB) with Nokia packet Abis with OsmoBSC. The Nokia packet Abis seems to be Nokia OML/RSL over SCTP/IUA (RFC 3057). To get started, we have built a little stub that takes the incoming SCTP/IUA connection and carries out ASP_UP/ASP_ACTIVE handshake. After this handshake is complete, we see OML coming in from the BTS. It seems that this OML is in a form that would be understood by "bts_nokia_site.c". So now we would like to continue the experiment by feeding the OML/RSL into a BTS configured in OsmoBSC as type "nokia_site". Now the questions: 1. Is it possible to configure a BTS of type "nokia_site" to run Abis over a unix domain socket ? 2. Is there a better way to prototype this integration ? PS: PCAP trace available on request Best Regards, Michael Andersen

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confused by osmo-bts rsl.c:bts_supports_cm()
by Keith 12 Sep '18

12 Sep '18

Hi All. posting to the list here rather than going down the line of a ticket as I'm using the nitb to configure my bts and so this may not be a bug, but rather a missing back port to legacy. (which may not be appreciated here, I'm aware!) Anyway, this is the scenario: I noticed a lack of audio with latest osmo-bts, and this log message: <0000> rsl.c:1503 invalid mode/codec instructed by BSC, check BSC configuration. I added some logging in osmo-bts to check what was being passed into this line in rsl.c: if (bts_supports_cm(lchan->ts->trx->bts, lchan->ts->pchan, lchan->tch_mode) != 1) and the values are 0 (zero), GSM_PCHAN_TCH_F_TCH_H_PDCH and GSM48_CMODE_SPEECH_AMR but bts_supports_cm() only checks against GSM_PCHAN_TCH_F and GSM_PCHAN_TCH_H Is it that osmo-bsc is sending the actual current channel mode now, whereas the osmo-nitb is sending the configured mode, i.e. TCH_F_TCH_H_PDCH ?? in which case, sorry to bother you, but I will ask: should we try to keep basic functionality between BTS and nitb for a little while longer :) Or are we missing checks for all possible modes in rsl.c:bts_supports_cm() Related commit: http://git.osmocom.org/osmo-bts/commit/src?id=a4bca1155 Thanks!! k/

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Relevant GSM 3GPP standards
by Rafael Diniz 11 Sep '18

11 Sep '18

Hi all, I'm diving into 3GPP standards after all the OsMux discussion. For example, LCLS, 3GPP TS 44.007 standard was mentioned. Trying to figure out where to download it, I reached: http://www.3gpp.org/DynaReport/44-series.htm But it's not there any 44.007. Any hint? Thanks, Rafael Diniz

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logging level all everything
by Neels Hofmeyr 11 Sep '18

11 Sep '18

Hi list, looking at the logging code, I was surprised: turns out the 'logging level all everything', though quite a misnomer, was an important counterpart to 'logging level all (debug|...|fatal)'. It worked like this: Assuming some levels are set... logging level foo fatal logging level bar debug Now 'logging level all' would override all of them forcefully: logging level all error # Now all categories including 'foo' and 'bar' log exactly error,fatal # messages. logging level all debug # Now all categories including 'foo' and 'bar' log all messages from debug # thru fatal. # manipulating individual categories has no effect at all now! logging level foo notice logging level bar error # no change in logging behavior! logging level foo fatal logging level bar debug # the way to get rid of the forced level was: logging level all everything # Now we see above configured behavior of foo == fatal, bar == debug This is *still the case*, only that we're now ignoring the 'everything' keyword. The consequence is that we can force all categories and clamp them to all-the-same level, but we cannot lift that clamp anymore! :/ I think we should never have accepted the removal of 'everything' as such, it should have been replaced with a more accurate command, like 'no logging level all'. I am considering to bring this back now, ... but But now that we haven't had 'everything' for a long time and the outrage about it has been limited, I am also drawn towards completely dropping that feature, or at least renaming it. Damage is done, no need to go back to mad naming. From the logging discussion, we concluded that we want a command to set the log level for all categories at once, one-time; not force-clamping all to one level until the clamp is removed, but as if we set each individual level manually. I've cracked my head on what name other than "all" we could want this command to have, 'logging level (*|each|any|set-all)'? I keep coming back to 'logging level all (debug|...|fatal)' being by far the best name for this. Backwards compatibility: what do users see when we modify 'logging level all' to not be a forceful clamp, but instead setting individual levels once-off? If a user had a config of "all" coming last: logging level foo debug logging level bar fatal logging level all notice then the experience is that all categories are logging at level 'notice', and that is what the user most likely also expects to happen. Changing the 'all' command does not change the logging behavior one bit. If in turn the config has "all" first: logging level all notice logging level foo debug logging level bar fatal Then the experience is still that before changing the command, all categories are logging at level 'notice'. Most likely the user expected though to see foo='debug' and bar='fatal', because they were set after the 'all' -- what other reason could a user have to keep these lines in the config file, which don't have any effect as-is? If we change the 'all' command to not be a forceful clamp but just a one-time set, then the logging behavior changes to what the user likely expected when writing this config. However: if these settings were forgotten in the config file, we would suddenly change the logging behavior and might surprise users ... but I still kinda think we would change it to what the user expected, right? And the user would likely go: "ah, finally it's doing what I wanted all the while!" right? So, I want to make 'logging level all' manipulate each individual category once-off, I want to completely deprecate the 'everything' keyword, and drop the "global clamp" feature entirely. Maybe I'd re-add the clamp as 'logging level force-all (debug|...|fatal)' and 'no logging level force-all' to switch it off. That would be exactly the old clamping feature just with less confusing names. The only thing that makes me write here is that I'm only 90% sure on changing the meaning of an existing command, of 'logging level all'. If there is no negative feedback on this here, that would bring me to 100%. Thanks! ~N

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OsMux + MGW + SIP
by Keith Whyte 11 Sep '18

11 Sep '18

Hi community! I'd like here to set out a few issues that we have identified for going forward with the rhizomatica autonomous networks and the osmocom split stack. Some of this has been very briefly discussed between myself and Pablo, and I bring it to the mailing list so hopefully there will be feedback, ideas and suggestions :) I am cherry picking from previous emails here, (mostly mine) please excuse any repetition. I hope others will post their previous comments on this thread, either verbatim of edited to be relevant to the current discussion. 1) This is a question that relates to voip traffic in and out of the autonomous communities over satellite. Currently this is what we HAVE to have in the local village to support autonomous operation): * BSC * MSC * HLR * Freeswitch (freeswitch is our call control, call routing, billing etc) We can't have any essential backhaul running over the satellite , like A for example as we would loose all functionality when the sat link is down, and we'd burn expensive bandwidth! One option to use osmux from autonomous community (village) is to teach osmo-mgw to speak osmux, then we have a co-located bsc/msc/mgw in the village, and the media transport is osmux to another mgw in our datacentre that is demuxing and converting to RTP suitable to be forwarded to our upstream VoIP provider. The question here is what is signalling the MGW in the data centre? Maybe another freeswitch could signal the osmoMGW via some kind of SIP<->MGCP translator. Or we teach the MGW to speak SIP? Another option that I want to put on the table and see what people say is to look at implementing osmux as a codec in freeswitch. I don't know what that would mean in terms of effort. I don't know if the osmux code can be abstracted, if that is the right term, into it's own external includible library that could then be used to build a freeswitch codec. I looked some implementations of AMR codec for freeswitch, and really it looks like a boiler plate codec with registration, setup and then it calls the encoding and decoding functions in the opencore amr library. Maybe, we can do the same with osmux? Then we could have two freeswitchs signalling and transcoding to/from osmux? - then we are not really working on osmo solution and others in the osmo community might prefer to see osmux fully implemented in MGW before anything else? Here are a number of related tickets from Harald on osmux integration: http://osmocom.org/issues/1683 and http://osmocom.org/issues/2832 as well as https://osmocom.org/issues/2909 I think though, that in 1683, we are stalled by https://osmocom.org/issues/2391 for the split setup. 2) Another question relates to this proposal of a media gateway less mode: https://osmocom.org/issues/3142 I think this suits us, because really our call signalling is all happening in freeswitch and we would prefer I think not to have the media gateway at all most of the time. and in fact for local Mobile to Mobile calls on the same BTS, we would in fact prefer the RTP be local on the sysmobts, or indeed BTS<->BTS! Freeswitch has a "bypass media" parameter, in fact you can even activate this at call processing time before bridging the call, depending on whether it makes sense in terms of direct connection being possible and the lack of transcoding. There's also a "bypass media after bridge" parameter that is automatically using SIP (re)-INVITES to switch the RTP stream from passing through freeswitch or going directly from end point to end point. Using "bypass media" in our profile works nicely; of course, as all it is doing is using the IP address(es) of the osmo-bts in the SDP, so the rtp stream loops on lo in the sysmobts. It would help with something that I mentioned at Osmodevcon, Harald, which was that in some cases we might like to avoid having our RTP go over the (sometimes variable quality) Wifi links between the BTS and the BSC. A problem is that if you use "bypass media" you have no early media and no audible ringing signal, you don't get any audio stream until the B leg picks up. This is what "bypass media after bridge" is for. Unfortunately, osmo-nitb (at least, and I don't believe anything has been done in osmo-bsc related) plus LCR or osmo-sip-connector combinations do not react correctly to the SIP reINVITES, you end up with multiple calls. So all that needs to be fixed / implemented. I noticed mention of implementation of MNCC_RTP_MODIFY on fairwaves branches of OpenBSC and LCR. I've compiled and tested this, but as far as I can see it is still not working, as issuing a SIP reINVITE from freeswitch is still setting up new calls, not just changing the RTP streams. Maybe I also need a fairwaves osmo-bts. So this is something I think I would like to see: * Full support for SIP reinvite in osmo-sip-connector which would then send a MNCC_RTP_MODIFY which ends up sending (I think it's called a IPAC_CRCX?) to osmo-bts which will then switch the stream endpoints. This I believe is also necessary for handover to function with an MNCC socket setup and we don't currently have it, not even in the osmo-nitb. * Implement a no media gateway mode in osmo-msc and have the MSC control the media stream using SIP via MNCC/osmo-sip-connector instead of controlling the MGW using MGCP. K/

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Re: OsMux + MGW + SIP
by Keith 09 Sep '18

09 Sep '18

I was browsing a little through the osmux code in openbsc and libosmo-netif, but I did not quite find the answer to this question I have: Pablo maybe you can shed light? When you divide the OSMUX stream again and turn each circuit ID back into individual RTP stream you have an associated port and IP, right? (on the terrestrial side) I'm wondering if we could have: * local bsc and it's colocated mgw * local msc, with two colocated mgws (we decide which one to signal depending on if the call is local or needs the SAT stream) another point here is I undertand we don't _really_ need two MGWs, so maybe just one in the village, shared between bsc/msc and one in the datacentre. so in the case of a call that is using sat/ osmux, the MSC in the village is signalling the MGW in the datacentre, therefore this MSC knows which IP/port combination to tell to osmo-sip-connector and on to freeswitch, which then signals whatever SIP UA in the data centre So, the fact that osmux is used at all is completely transparent to the SIP UAs Does that make sense? k/

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