Osmo-sip-connector » History » Revision 13
Revision 12 (dexter, 02/06/2017 03:26 PM) → Revision 13/87 (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" "osmo-nitb [label = "A.bis"]; "A.bis"]"; osmonitb -> "osmo-sip-connector" [label = "mncc"]; } }}