After a lot of work[1] the gsm tester can finally:
* Start a virtual bts
* Mobile/virtphy processes
* Complete LUs for 10 MS.
The next step (besides having a proper test verdict) is scaling this beyond what a simple physical set-up can provide. Let's go to 100+ BTS and 10k subscribers but somehow I am still holding it wrongly and would like to have feedback to see how to evolve the gsm tester.
What do I need to change to scale this up and how to externally parameterize it?
* suite.conf:
Add one line per virtual bts to reserve (I would prefer to specify type+num)
* resources.prod.conf:
Add more Virtual BTS. I started to pick IPs from 127.0.1.0/24 as it avoids having to configure special IPs.
* register_default_mass.py:
I need to somehow know how many BTS (and NITBs) were reserved. Or in the long run the topology of how to connect M BTS to N BSCs? Currently I can run until I get an exception but that is not desirable.
What's missing:
* High-level API of the SuiteRun to get me the toplogy of the network. It seems undesirable in the specific suite to discover how many BTSs were reserved in suite.conf or what the topology is.
* ARFCN usage. Besides the redundancy all my BTS currently have the same ARFCN. There should be an easy way to configure an band+arfcn pool.
* IMSI/MSISDN pooling. I would like to specify pools of IMSI/MSISDN pairs (and size) and then draw from it. I needs these to program into the mobile, NITB/HLR/AuC and for client to client SMS transfers.
* Configure these capacities from the outside. Changing from 1 to 256 BTS should be a single line (or even a parameter change).
Any idea or comment on how to achieve this?
cheers
holger
[1] Hindsight is 20/20 and the difficulty was not adding scripting to mobile but getting the GSM tester to spawn the network and we are unfortunately not done yet.
Hello Everyone
I was trying to set priority bit in paging type 1. I tried modifying the function rsl_paging_cmd in abis_rsl.c . But when i am checking the same in wireshark, I couldn't view it. Any help on this.
BR
Dear fellow Osmocom developers,
I'm a bit surprised to notice that not more people have signed up for
OsmoDevCon 2019. I guess it was mostly an oversight when the date was
originally announced, and not a lack of interest? ;)
All details about the event are available at the related wiki page at:
https://osmocom.org/projects/osmo-dev-con/wiki/OsmoDevCon2019
Please enter your name at
https://osmocom.org/projects/osmo-dev-con/wiki/OsmoDevCon2019#Requested
in case you would like to attend. Registering early allows proper
planning. Thanks!
Looking forward to meeting old and new Osmocom developers in April 2019.
Regards,
Harald
--
- Harald Welte <laforge(a)gnumonks.org> http://laforge.gnumonks.org/
============================================================================
"Privacy in residential applications is a desirable marketing option."
(ETSI EN 300 175-7 Ch. A6)
I have trx, bts, bsc, msc, stp, hlr, and sip installed...
asterisk runs on a separate machine, i.e. my ordinary pbx, with just the
recomended
changes to sip and extensions.
When dialling an external number I do not get any call progress
(ringing) in the MS.
Phone is quiet as I dial the number, when the call is answered voice is
"suddenly" there,
this is a difference from "latest" where call progress was audiable but
had only oneway audio.
I run HLR with the -U switch, there seems to be a difference between
"latest" and "nightly",
what does the -U switch do, modify the database, or just run HLR in
backward compatibility mode?
Should I delete / reregister phones?
Regards,
Gullik
Hie
My last work with osmo when it was an NITB. I noticed that with the new
split, we have sigtran SS7. I woul like to know if it can handle roaming
with other GSM networks.
Regards
shingy
Hi,
Where am I supposed to set initial rx-gain and tx-power ( tx-attenuation) ?
I see no suitable keywords in the BTS ( bts-uhd ) nor in the BSC.
btw, sdr is Ettus B100 driven by uhd-trx, and shows 10 dB attenuation.
Regards,
Gullik
Hello,
I have recently started to deploy OpenBSC . Initially I hope to
contribute to manuals and howtos
and with bug reports and testing. I have followed the suggested line,
and installed everything from
binarys, which puts for instance osmo-msc at 1.2.0. The platform is
Orange Pi Zero, and UHD / B100,
but will evolve to OPI Z + limesdr-mini.
I have encountered the "swapped address" bug reported some 3-4 months
ago, but is there in
osmo-msc 1.2.0, i.e. resulting in one way audio, since the sdp is
addressed to a byte reversed
address. Apart from that problem I have some minor stability issues
(more later ).
so, my entry #1
Latest .deb packages have an issue with requesting the RTP stream for
audio TO Mobile
with a bogus IP address:
16:20:17.739790 IP (tos 0x0, ttl 64, id 50362, offset 0, flags [none],
proto UDP (17), length 698)
orangepizero.lan.5065 > slottet.lan.sip: [udp sum ok] SIP, length: 670
INVITE sip:1000@192.168.1.80:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.170:5065;rport;branch=z9hG4bKcgB6KF1r9Z8Zj
Max-Forwards: 70
From: <sip:0757576000@192.168.1.170:5065>;tag=9U40t8B3a8p5r
To: <sip:1000@192.168.1.80:5060>
Call-ID: 590931be-7be8-1237-51bb-0242ed556d0c
CSeq: 938485738 INVITE
Contact: <sip:192.168.1.170:5065>
User-Agent: sofia-sip/1.12.11devel
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,
SUBSCRIBE, NOTIFY, REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Length: 129
v=0
o=Osmocom 0 0 IN IP4 170.1.168.192
s=GSM Call
c=IN IP4 170.1.168.192
t=0 0
m=audio 4036 RTP/AVP 0
a=rtpmap:0 GSM/8000
Asterisk on 192.168.1.80, osmo stack on 192.168.1.170,
Best Regards,
Gullik
Dear Osmocom community,
Who deals with configuration of nano3g ip.access here? How do you configure it?
We use the latest ip.access firmware that has a bunch of bugfixes and it doesn't
support "older" telnet configuration methodology:
https://osmocom.org/projects/cellular-infrastructure/wiki/Configuring_the_i…
So, do you just use other/older firmware or do you have TR069 ACS set up?
Kind regards,
Mykola
