Cellular Network Infrastructure » Core Network (CN) » osmo-sip-connector

h1. Osmo-sip-connector

h2. Introduction

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

h2. Asterisk howto

This howto was created and tested using Debian 8.7.1. It is assumed that there is already a working installation of osmo-nitb already exists. In the following steps we will show how to integrate an existing installation with asterisk using osmo-sip-connector. The following image illustrates how the network components wile be laid out:

{{graphviz_link()

digraph G{

  //rankdir = LR;

  Phone -> BTS [label = "Um"];

  BTS -> "osmo-nitb" [label = "A.bis"];

  "osmo-nitb" -> "osmo-sip-connector" [label = "MNCC"];

  "osmo-sip-connector" -> "asterisk" [label = "SIP"];

  "BTS" -> "asterisk" [label = "RTP"];

}

}}

h3. Preparations

As we assume that we use an existing and working osmo-nitb configuration as a starting point most of the relevant dependencies are already installed. However, we still miss osmo-sip-connector and asterisk as a major components.

* Install dependancies:

The extension we are about to make requires to install some additional packages.

<pre>

sudo apt-get install libsofia-sip-ua-glib-dev

sudo apt-get install asterisk

</pre>

* Install osmo-sip-connector:

The installation of osmo-sip-connector is as straight forward as you know it already from other osmocom projects:

<pre>

git clone git://git.osmocom.org/osmo-sip-connector.git

cd osmo-sip-connector/

autoreconf -fi

./configure

make

sudo make install

</pre>

h3. Configuration of osmo-sip-connector

* Osmo-nitb:

Osmo-nitb does not require any change of of its configuration. The connection between osmo-sip-connector is configured using the command line option “-M”:

<pre>

osmo-nitb -c ./openbsc.cfg -d option –debug=DRLL:DCC:DMM:DRR:DRSL:DNM -M /tmp/bsc_mncc

</pre>

The “-M” option disables the internal call control handler of osmo-nitb and offers an unix domain socket, where an external call control handler can be connected. In the described situation this will be osmo-sip-connector.

* osmo-sip-connector

Osmo-sip connector acts as a translator between the classic ISDN mncc and SIP. The configuration is rather simple.

<pre>

app

mncc

  socket-path /tmp/bsc_mncc

sip

  local 10.9.1.110 5069

  remote 10.9.1.110 5060

</pre>

Again, we have to setup a socket path. This path has to be the same as we provided with the commandline option “-M” for osmo-nitb earlier. For SIP we need to setup a remote and local IP-Address and port. In most cases Asterisk and osmo-sip-connector will run on one and the same machine, so it is ovious that local and remote IP-Address are the same. The type of connection between osmo-sip-connector and Asterisk is a classic SIP-Trunk. For asterisk it appears as just another PBX. Osmo-sip-connector is started via commandline as follows:

<pre>

osmo-sip-connector -c ./osmo-sip-connector.cfg

</pre>

h3. Configuration of asterisk

The following section will discuss a minimal Asterisk configuration that is able to route calls between mobile phones. The described case assumes a freshly installed asterisk with virgin configuration files. In the following we will edit extensions.conf and sip.conf which can be found in the /etc/asterisk directory. The example below is not special to GSM, its rather a normal SIP-trunk configuration.

* Set up the sip-trunk towards osmo-sip-connector:

The first that has to be taken care of is the connection to osmo-sip-connector. Two connections have to be created. One for outgoing and one for incoming calls. The following lines need be appended to sip.conf:

<pre>

[GSM]

type=friend

host=10.9.1.110

dtmfmode=rfc2833

canreinvite=no

disallow=all

allow=gsm

context=gsmsubscriber

port=5069

</pre>

The connection are named GSM, as we will reference to this connection later it is recommended to keep the name short. The context name “gsmsubscriber” references the context where mobile originated calls are routed into the dialplan.

* Set up a dialplan to route calls:

Now asterisk and osmo-sip-connector are connected. The following example introduces a very basic dialplan configuration to route calls between mobile phones. The following example can be added to extensions.conf:

<pre>

[gsmsubscriber]

exten=>_XXXXX,1,Dial(SIP/GSM/${EXTEN})

exten=>_XXXXX,n,Playback(vm-nobodyavail)

exten=>_XXXXX,n,HangUp

</pre>

The name in “[]” is the context name which we already defined in sip.conf (“gsmsubscriber”). Every incoming call, that is entering through the “GSM” connection is directly routet to “[gsmsubscriber]”. The following three lines will check if the dialed number is exactly 5 digits long and if so, just dial it through the “GSM” connection. This causes the call to be routet back through the “GSM” connection back to osmo-nitb. If the subscriber is not available, the call will fail and alternatively a voice message “vm-nobodyavail” will be played. Finally the call is hung up.