Hi Gullik,

first of all, thanks for testing and reporting the issues you see.

Where are you installing your packages from? Osmocom repositories? latest [1] or nightly [2]?

Packages from "latest" start to be a bit old nowadays, and have some known bugs. So in general if you are planning to contribute and you are getting used to the system, I'd recommend using nightly to follow development more tightly, at least until we have a new release. If you find an issue, please give it a try on nightly and see if the bug is still there, then submit a ticket in redmine. Patches welcome too of course.

Fyi, I plan to do a new release of all components soon in mid January when I'm back from Holidays.

[1] https://osmocom.org/projects/cellular-infrastructure/wiki/Latest_Builds
[2] https://osmocom.org/projects/cellular-infrastructure/wiki/Nightly_Builds

On 12/17/18 10:00 AM, Gullik Webjorn wrote:
It looks like this is addressed in change 11139, however this was 3 months ago, and build date

of osmo-msc_1.2.0_armhf.deb seems to be nov-10, but it seems not in there.

Would the fix be in osmo-msc_1.2.0.140.1051c_armhf.deb ?

Gullik


On 2018-12-17 09:46, Gullik Webjorn wrote:
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:[email protected]:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.1.170:5065;rport;branch=z9hG4bKcgB6KF1r9Z8Zj
        Max-Forwards: 70
        From: <sip:[email protected]:5065>;tag=9U40t8B3a8p5r
        To: <sip:[email protected]: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


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