First post

[Gullik Webjorn]

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 at 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 at 192.168.1.170:5065>;tag=9U40t8B3a8p5r
         To: <sip:1000 at 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