Osmo-sip-connector¶

osmo-sip-connector translates between MNCC and SIP protocols. It does not handle RTP by itself but with the help of external SIP server it can be used for tests.

Sample configuration:

app
mncc
  socket-path /tmp/bsc_mncc
sip
  local 10.9.10.105 5069
  remote 10.9.10.105 5060

Running osmo-sip-connector:

osmo-sip-connector -c ~/.config/osmocom/osmo-sip-connector.cfg

Running NITB:

./src/osmo-nitb/osmo-nitb -c ~/.config/osmocom/open-bsc.cfg -l ~/.config/osmocom/hlr.sqlite3 -d DLMUX:DRTP -m

The configuration above assumes that SIP server is running on the same machine. Attached is example configuration file for Kamailio https://www.kamailio.org SIP server which can be used to route calls between mobile phones. It also handles 2 special numbers 500 (routed to sip:music@iptel.org) and 600 (routed to sip:echo@iptel.org): by dialing them you can use echo test or hear nice music from your mobile.

Note: in attached kamailio.cfg, for 64bit systems, you may need to adjust

mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" 

N. B: Those numbers are meant only as an example for quick tests - please consider running your own Asterisk instance if you expect more than couple of calls, do not abuse http://www.iptel.org/service

TODO: Add asterisk and other SIP servers configuration example.

It looks a bit like this: