RestComm is a robust, powerful platform that will facilitate building comprehensive real-time communication solutions. The steps below will help you get started with ease.
- On a production system, please take into consideration the following:
- MMS can get into a bad state and report “Too many open files” error because of Linux default value on the total number of file descriptors.
- You can go to the following sites to see how to increase the maximum number of open files in your server:
- http://www.cyberciti.biz/faq/linux-increase-the-maximum-number-of-open-files/
- http://amitbhayani.blogspot.fr/2010/01/javanetsocketexception-too-many-open.html.
Running RestComm on JBoss
JBoss Security
Running a secure application is dependent on multiple factors. Restcomm runs on JBoss which implies that system security implemention can be handled at JBoss level. Please see the link below on how you can make your server environment more secure. https://community.jboss.org/wiki/SecureJboss?_sscc=t.
Start Media Server
- Download
TelScale-Restcomm-JBoss-AS7-XX.XX.GA.zip - Using a terminal of your choice, extract the content of
TelScale-Restcomm-XX.XX.GA.zipto a local directory on your computer. The root directoy into which you extract the content of the .zip file will be referred to as $RESTCOMM_HOME. - Go to
$RESTCOMM_HOME/telscale-media/telscale-media-server/bin - change the permission of the
run.shas follows - sudo chmod +x ./run.sh
- then start the TelScale Media Server as follows
- sudo ./run.sh
- If all is correctly started you will see the following at the end of the bash terminal window
[MainDeployer] [[[[[[[[[ Mobicents Media Server: release.version=3.0.0.FINAL Started ]]]]]]]]]
Start Restcomm
- Open another terminal and proceed as follows:
- Go to
$RESTCOMM_HOME/bin - change the permissions of all the
.shfiles in the bin directory as follows sudo chmod +x ./*.sh- Start RestComm by running the following command
sudo ./standalone.sh -c standalone-sip.xml- If RestComm is correctly started you will see the following at the end of the terminal
INFO [Version] TelScale Sip Servlets 6.1.3.GA-TelScale (build: Git Hash=r8947f2732ee64c76566ed6c0b236204c048538e1 date=201306131639) Started. 17:30:05,854 INFO [Version] ============================================================================== == == == Thank you for running TelScale == == Carrier Grade Communications Platform by the creators of Mobicents == == Copyright 2011-2013 Telestax, Inc. == == http://www.telestax.com/ == == == ==============================================================================
Testing the Demo Applications
Restcomm comes prepackaged with multiple example applications designed to help you quickly get started.
Demo 1 – Testing the Play Verb
Start a SIP phone (see below) and dial 1234@127.0.0.1:5080 . You will hear a welcome message.
Warning
Some SIP phones have codec incompatibility issues, in the above example, we used Ekiga. You may also try Jitsi or Sflphone.
The application bound to the number 1234 can be found at <filename>$RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/hello-play.xml</filename>.
Demo 2 – Testing Say Verb
You must first activate Text-to-Speech before you can proceed.
You can register for a free account and an API key will be emailed to you. Once you have the API key, open the $RESTCOMM_HOME/standalone/deployments/restcomm.war/WEB-INF/conf/restcomm.xml file and find the speech-synthesizer VoiceRSS section. Add your API key as shown below and restart RestComm
<speech-synthesizer class="org.mobicents.servlet.restcomm.tts.VoiceRSSSpeechSynthesizer"> <service-root>http://api.voicerss.org</service-root> <apikey>2901c0aXXXXXXXXXXXXXX</apikey>
Start a SIP phone dial 1235@127.0.0.1:5080 . You will hear a welcome message in multiple languages.
The application bound to the number 1235 can be found at $RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/hello-world.xml.
Demo 3 – Testing Gather Verb
This demo creates a simple IVR system
Start a SIP phone dial 1236@127.0.0.1:5080 . You will hear a message asking you to enter a digit and press star. If the digit is correctly received, Restcomm will replay the number you entered.
<a id="d0e603"></a> The application bound to the number 1236 can be found at <filename>$RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/gather/hello-gather.xml</filename>. and <filename>$RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/gather/gather.jsp</filename>.
Demo 4 – Testing the Dial Sip Noun
This demo makes a call from one SIP phone to another. The Demo uses the SIP noun. You can calll any SIP account. All you have to do is change the content of the dial-sip.xml
In order to use this demo, you may use the default accounts, Alice and Bob, and register them on two separate SIP phones. Start both SIP phones and make sure Alice and Bob are registerd with the password (1234). These users come pre-configured with Restcomm for test purposes.
Start Two SIP Phone Sessions
If you are using the SIP sflphone here is what to do:
In the application dial-sip.xml you can change the default to sip:alice@127.0.0.1:5061?
This will allow you to make a call to Alice. Note that Alice must be registered on port 5061 for the call to succeed.
From the the phone on which Bob is registerd, dial the number 1237 . The phone on which Alice is registered will ring and the connection will be made when you answer the call.
<a id="d0e622"></a> The application bound to the number 1237 can be found at $RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/dial/sip/dial-sip.xml.
Demo 5 – Testing the Client Noun
This demo makes a call from one SIP Client to Another. The demo uses the Client noun
In order to use this demo, you must have user Alice and Bob registered on two separate SIP phones. Start both SIP phones and make sure Alice and Bob are registerd with the password (1234). These users come pre-configured with Restcomm for test purposes.
From the phone on which Bob is registerd, dial the number 1238 . The phone on which Alice is registered will ring and the connection will be made when you answer the call.
<a id="d0e658"></a> The application bound to the number 1238 can be found at <filename>$RESTCOMM_HOME/standalone/deployments/restcomm.war/demos/dial/client/dial-client.xml</filename>.
Demo 6 – Testing Conference Noun
This demo Lets a user join a conference as a moderator and the other user as a participant. The participant will dial 1310 and will hear a hold music. The moderator will dial 1311 and the hold music will stop and the conference will be started.
Most SIP phones will require you to register before you can make a call. You can use the default accounts, Alice and Bob with password (1234)to register.
From the phone on which Bob is registerd, dial the number 1310 . From the phone on which Alice is registered, dial 1311
<a id="d0e691"></a> The application bound to the number 1310 and 1311 can be found at http://127.0.0.1:8080/restcomm/demos/dial/conference/dial-conference.xml and at http://127.0.0.1:8080/restcomm/demos/dial/conference/dial-conference-moderator.xml
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