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updated installation steps for opensips

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This commit is contained in:
gezimbll
2024-02-08 08:42:54 -05:00
committed by Dan Christian Bogos
parent 8fe3a74998
commit 1ec7cbccf8
2 changed files with 114 additions and 12 deletions
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2
data/tutorials/osips/opensips/etc/opensips/opensips.cfg
View File

@@ -199,7 +199,7 @@ route{
}
# do lookup with method filtering
if (!lookup("location","m")) {
if (!lookup("location","method-filtering")) {
t_reply(404, "Not Found");
exit;
}

124
docs/tutorial.rst
View File

@@ -128,11 +128,11 @@ Regarding the SIP Servers, click on the tab corresponding to the choice you made
.. code-block:: bash
apt-key adv --keyserver keyserver.ubuntu.com --recv-keys 049AD65B
echo "deb https://apt.opensips.org buster 3.3-releases" >/etc/apt/sources.list.d/opensips.list
echo "deb https://apt.opensips.org buster cli-nightly" >/etc/apt/sources.list.d/opensips-cli.list
apt-get update
sudo apt-get install opensips opensips-mysql-module opensips-cgrates-module opensips-cli
curl https://apt.opensips.org/opensips-org.gpg -o /usr/share/keyrings/opensips-org.gpg
echo "deb [signed-by=/usr/share/keyrings/opensips-org.gpg] https://apt.opensips.org bookworm 3.4-releases" >/etc/apt/sources.list.d/opensips.list
echo "deb [signed-by=/usr/share/keyrings/opensips-org.gpg] https://apt.opensips.org bookworm cli-nightly" >/etc/apt/sources.list.d/opensips-cli.list
apt-get update
sudo apt-get install opensips opensips-mysql-module opensips-cgrates-module opensips-cli
Configuration and initialization
--------------------------------
@@ -217,22 +217,124 @@ Regarding the SIP Servers, we have prepared custom configurations in advance, as
- user accounts configuration not needed since it's enough for them to only be defined within CGRateS;
- for simplicity, no authentication was configured (WARNING: Not suitable for production).
- creating database for the DRouting module, using the following command:
.. code-block:: bash
opensips-cli -x database create
After creating the database for DRouting module populate the tables with routing info:
.. code-block:: bash
insert into dr_gateways (gwid,type,address) values("gw2_1",0,"sip:127.0.0.1:5082");
insert into dr_gateways (gwid,type,address) values("gw1_1",0,"sip:127.0.0.1:5081");
insert into dr_carriers (carrierid,gwlist) values("route1","gw1_1");
insert into dr_carriers (carrierid,gwlist) values("route2","gw2_1");
To start OpenSIPS_ with the prepared custom configuration, run:
.. code-block:: bash
.. code-block:: bash
sudo mv /etc/opensips /etc/opensips.old
sudo cp -r /usr/share/cgrates/tutorials/osips/opensips/etc/opensips /etc
sudo systemctl restart opensips
/usr/share/cgrates/tutorials/osips_native/opensips/etc/init.d/opensips start
To verify that OpenSIPS_ is running, run the following command:
.. code-block:: bash
.. code-block:: bash
opensips-cli -x mi uptime
Since we are using OpenSIPS_ with DRouting module we have to set up a SIP entity that OpenSIPS_ can forward the calls to for our setup.
In this example we use SIPp a free Open Source test tool / traffic generator for the SIP protocol.
The install SiPp use commands below :
.. code-block:: bash
apt update
apt install git pkg-config dh-autoreconf ncurses-dev build-essential libssl-dev libpcap-dev libncurses5-dev libsctp-dev lksctp-tools cmake
git clone https://github.com/SIPp/sipp.git
cd sipp
git checkout v3.7.0
git submodule init
git submodule update
./build.sh --common
cmake . -DUSE_SSL=1 -DUSE_SCTP=0 -DUSE_PCAP=1 -DUSE_GSL=1
make all
make install
Write SIPp XML scenario named uas.xml or to your liking with the content below,this scenario will simulate calls with OpenSIPS_ .
Change "OpenSIPS_IP" in the line *<sip:OpenSIPS_IP:[local_port];transport=[transport]>* with your OpenSIPS_ IP.
.. code-block:: XML
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uas' scenario. -->
<!-- -->
<scenario name="Basic UAS responder">
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<!-- Adding ignoresdp="true" here would ignore the SDP data: that -->
<!-- can be useful if you want to reject reINVITEs and keep the -->
<!-- media stream flowing. -->
<recv request="INVITE" crlf="true"> </recv>
<!-- The '[last_*]' keyword is replaced automatically by the -->
<!-- specified header if it was present in the last message received -->
<!-- (except if it was a retransmission). If the header was not -->
<!-- present or if no message has been received, the '[last_*]' -->
<!-- keyword is discarded, and all bytes until the end of the line -->
<!-- are also discarded. -->
<!-- -->
<!-- If the specified header was present several times in the -->
<!-- message, all occurrences are concatenated (CRLF separated) -->
<!-- to be used in place of the '[last_*]' keyword. -->
<send>
<![CDATA[ SIP/2.0 180 Ringing [last_Via:] [last_From:] [last_To:];tag=[pid]SIPpTag01[call_number] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Content-Length: 0 ]]>
</send>
<send retrans="500">
<![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:];tag=[pid]SIPpTag01[call_number] [last_Call-ID:] [last_Record-Route:] [last_CSeq:] Contact: <sip:OpenSIPS_IP:[local_port];transport=[transport]> Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]>
</send>
<recv request="ACK" optional="true" rtd="true" crlf="true"> </recv>
<recv request="BYE"> </recv>
<send>
<![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Content-Length: 0 ]]>
</send>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<timewait milliseconds="4000"/>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
Run the SIPp with the command below:
.. code-block:: bash
sipp -sf uas.xml -p 5082
opensipsctl moni
@@ -281,7 +383,7 @@ Just as with the SIP Servers, we have also prepared configurations and init scri
.. code-block:: bash
/usr/share/cgrates/tutorials/osips_native/cgrates/etc/init.d/cgrates start
sudo systemctl restart opensips
.. note::
If you have chosen OpenSIPS_, CGRateS has to be started first since the dependency is reversed.