Osmo-sip-connector » History » Revision 12
Revision 11 (dexter, 02/06/2017 03:25 PM) → Revision 12/87 (dexter, 02/06/2017 03:26 PM)
h1. 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:
<pre>
app
mncc
socket-path /tmp/bsc_mncc
sip
local 10.9.10.105 5069
remote 10.9.10.105 5060
</pre>
Running osmo-sip-connector:
<pre>
osmo-sip-connector -c ~/.config/osmocom/osmo-sip-connector.cfg
</pre>
Running NITB:
<pre>
./src/osmo-nitb/osmo-nitb -c ~/.config/osmocom/open-bsc.cfg -l ~/.config/osmocom/hlr.sqlite3 -d DLMUX:DRTP -m
</pre>
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
<pre>
mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/"
</pre>
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:
{{graphviz_link()
digraph G{
//rankdir = LR;
Phone -> BTS [label = "Um"];
BTS -> "osmo-nitb [label = "A.bis"]"; "osmo-nitb";
osmonitb -> "osmo-sip-connector" osmosipconnector [label = "mncc"];
}
}}
