Dear Alexander,
Sorry for the lots of mails :-)
I found the issue, but first let me go through what I did:
1. Osmo-BTS fiarwaves_rebase: I tried it. it compiles fine, but the problem is
the same.
2. I tried to use fairwaves/master for Libosmo-abis, but the Fairwaves/master
branch of OpenBSC is not compiling with the Fairwaves/master branch of
Libosmo-abis. This is the error log:
CC abis_nm.o
CC abis_nm_vty.o
CC abis_om2000.o
CC abis_om2000_vty.o
CC abis_rsl.o
CC bsc_rll.o
CC paging.o
CC bts_ericsson_rbs2000.o
CC bts_ipaccess_nanobts.o
In file included from bts_ipaccess_nanobts.c:39:0:
/usr/local/include/osmocom/abis/ipaccess.h:7:8: error: redefinition of âstruct
ipaccess_unitâ
struct ipaccess_unit {
^
In file included from bts_ipaccess_nanobts.c:38:0:
/usr/local/include/osmocom/gsm/ipa.h:11:8: note: originally defined here
struct ipaccess_unit {
^
make[3]: *** [bts_ipaccess_nanobts.o] Error 1
make[3]: Leaving directory `/root/osmocom/openbsc/openbsc/src/libbsc'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/root/osmocom/openbsc/openbsc/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/root/osmocom/openbsc/openbsc'
make: *** [all] Error 2
3. I tried to revert LibORTP to an older version, so I reverted from 0.24.2 to
0.22.0, recompiled everything and now the voice calls are working fine.
So there is definitely a compatiblity issue with ORTP version 0.24.2 and the
compile script is not taking care of that.
Now I am using Libosmo-abis master, OpenBSC Fairwaves/master, Osmo-BTS
fairwaves_rebase and Osmo-TRX master and everything seems to work on a B200.
Regards,
Csaba
----- Eredeti üzenet -----
Feladó: "Sipos Csaba" <[email protected]>
Címzett: "Alexander Chemeris" <[email protected]>
Másolatot kap: "OpenBSC Mailing List" <[email protected]>
Elküldött üzenetek: Vasárnap, 2015. Szeptember 20. 11:35:17
Tárgy: Re: Half sided calls
Hi Alexander,
Just attached the log with the L1C set to notice level. For this test I did not
used "-m" nor "-P", just pure OpenBSC with two phones calling each other.
I would highlight the following line:
<0014> trau/osmo_ortp.c:132 osmo-ortp(52995): network_error
>From the very beggining I get a sense that maybe the RTP is not working (I got
>half sided calls on SIP with asterisk previously because of RTP port issues).
>I am using ORTP version 0.24.2 maybe it is too new?
One more thing: I am using the master branch of Libosmo-Abis. Can that be a
problem?
Regards,
Csaba
----- Eredeti üzenet -----
Feladó: "Sipos Csaba" <[email protected]>
Címzett: "Alexander Chemeris" <[email protected]>
Másolatot kap: "OpenBSC Mailing List" <[email protected]>
Elküldött üzenetek: Vasárnap, 2015. Szeptember 20. 11:10:54
Tárgy: Re: Half sided calls
Dear Alexander,
> Any reason it's specific to B2x0?
Not really, now that UmTRX also uses the standard UHD branch, it should be the
same. The only reason is that I have a B200 and a B210 (finally) to test, but I
dont have for example a UmTRX, so I cant validate my method against that piece
of hardware.
> So it seems like your downlink doesn't work rather than uplink.
Trust me, its the uplink which is not working. When I used LCR (through MISDN
and chan_lcr) to hook OpenBSC to Asterisk, I was able to hear both the test
music and a test tone, but the echo test was not working. When I left LCR and
Asterisk out (just two mobiles calling each other) I was not able to hear
anything. I am using normal Full Rate calls (no AMR, no EFR, no HR).
Anyway is this the desired way to hook OpenBSC to Asterisk? I know LCR also
have a SIP method to connect to Asterisk, but I never figured that out and have
good experience with the MISDN method.
> which branch are you testing with?
For Osmo-BTS I used Fairwaves/master, for Osmo-TRX I used the master branch
(the fiarwaves/master is still not compiling due to the compile time
instruction set issue we identified a few weeks ago). I can try
fairwaves/rebase for Osmo-BTS if you want.
> I'd suggest checking that your uplink works
I use RX gain 12, the MS TX power is limited to 20dBm and I also have a duplex
filter. With this setup "show lchan" say the uplink is around -47dBm the
downlink is around -67dBm. The TRX log does not indicate any saturation issue
(or any other problems during the call).
Will try and enable the extra logging and try to extract some meaningful
information from that first, after that will try fairwaves/rebase.
Thanks for your help!
Regards,
Csaba
----- Eredeti üzenet -----
Feladó: "Alexander Chemeris" <[email protected]>
Címzett: "Sipos Csaba" <[email protected]>
Másolatot kap: "OpenBSC Mailing List" <[email protected]>
Elküldött üzenetek: Szombat, 2015. Szeptember 19. 20:09:54
Tárgy: Re: Half sided calls
Hi Sipos,
On Sat, Sep 19, 2015 at 5:05 PM, Sipos Csaba
<[email protected]> wrote:
> I am in the process of creating the Wiki page for Ettus B200/B210 with
> OpenBSC, GPRS and Asterisk.
Any reason it's specific to B2x0? Configuration should be 99% common
for all SDR devices and I think it makes more sense to have a single
page for that, like the old network_from_scratch.
> I am quite close, both data and calls are working, but the voice calls are
> half sided. The downlink direction works, but the uplink does not.
>
> I tried without Asterisk and LCR (between two phones) and still the calls are
> half sided.
>
> In the mean time I got these messages in the Osmo-BTS log:
>
> <0006> scheduler.c:276 PH-DATA.req: chan_nr=0x0a link_id=0x00 fn=1284768 ts=2
> tr
> x=0
> <0006> scheduler.c:1036 TCH/F has not been served !! No prim for trx=0 ts=1
> at f
> n=1284764 to transmit.
> <0006> scheduler.c:1036 TCH/F has not been served !! No prim for trx=0 ts=2
> at f
> n=1284764 to transmit.
Log message suggests that downlink doesn't receive a frame from RTP
side and thus transmits a filler frame. So it seems like your downlink
doesn't work rather than uplink.
It's not clear what do you mean that a call is half sided? According
to the logs, downlink frames are not served at both timeslots, so you
should not hear anything on both sides of the call.
I'd suggest checking that your uplink works, e.g. by looking at
osmo-trx logs. If you have the latest one, it should output a warning
if Rx side is saturated, which is the most possible reason for the
issue.
Another possible issue is codec. I have tested only with GSM-FR so
far. If you enable more debugging in osmo-bts, you should be able to
see decoding result at least for gsm-fr codec. I recommend you enable
at least NOTICE level for L1C and see if there are any fail messages
from here:
http://cgit.osmocom.org/osmo-bts/tree/src/osmo-bts-trx/gsm0503_coding.c?h=201509-fairwaves-rebase
One more thing - which branch are you testing with? I've been testing
201509-fairwaves-rebase. If you're using something else - I'd be
curious what happens if you try 201509-fairwaves-rebase.
--
Regards,
Alexander Chemeris.
CEO, Fairwaves, Inc.
https://fairwaves.co
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