Systems & Technology Group © 2007 IBM Corporation Rapid VoIP Application Development On Linux...

Systems & Technology Group

© 2007 IBM Corporation

Rapid VoIP Application DevelopmentOn Linux

Moisés Humberto Silva Salmerón

System i Access for Linux Development

QSE 2007


© 2007 IBM Corporation2 Aug 31, 2007

: Agenda : Demystifying VoIP

What VoIP is not

What is VoIP?

Consequences of VoIP

What is a PBX?

Common VoIP solutions

Next generation VoIP solutions



: Agenda : VoIP Application Development On Asterisk

What is Asterisk?

Development with Asterisk Extension Language

Asterisk Gateway Interface

Asterisk Manager Interface



: Demystifying VoIP :



: What VoIP is not :

Not yet a full replacement for POTS (Plain Old Telephony Service).

Not a cheap way to make long distance calls.

Not making calls over the Internet.



: What is VoIP? :

Transmission of voice frames using IP network facilities.

UDP is preferred over TCP because of the real time nature of telephony calls.

Integration of telephony services in IP networks.



: Consequences of VoIP :

Cheap calls. Side effect due to the way Internet Service Providers usually charge for Internet access.

Flexible communication.

Easy service integration and convergence. Telephony is now just another service in the IP network.

Number portability. You can be reached wherever you are.



: What is a PBX? :

Private Branch Exchange.

Communication system to connect company private telephony network with the PSTN ( Public Switched Telephony Network ) .

Provide services like IVR ( Interactive Voice Response ) menus, voice mail, conferences.



: Common VoIP solutions :

CISCO Call Manager.

Avaya.

Nortel.

3COM.



: Next generation VoIP solutions : better off with software ...

IBM System i with 3com software on Linux LPAR.

IBM System i with Nortel software on Linux LPAR.



: Next generation VoIP solutions : even better with free software.

Open source solutions exist, and can be used on any Linux computer, including high performance systems like the IBM System i.

IBM System i with Asterisk on Linux LPAR.

IBM System i with FreeSWITCH on Linux LPAR.



: VoIP Application Development On Asterisk :



: What is Asterisk? :

Software PBX project started by Mark Spencer.

Current development is led by Digium, a company founded because of the great Asterisk business potential.

Hundreds of programmers all over the world contribute. Code is under GNU General Public License, however, another business-friendly license is available for a fee.

FreeSWITCH, a similar project, is coming quickly with a more business-friendly license ( Mozilla Public License ).




Supported IP technologies include:

– SIP ( Session Initiation Protocol )

– H.323

– RTP ( Real Time Protocol )

– IAX2 ( Inter Asterisk Exchange )

– SCCP ( Skinny Client Control Protocol )

– MGCP ( Media Gateway Control Protocol )

– Jingle ( Designed by Google & XMMP standards foundation )




Non-IP supported technologies include:

– E1 / T1 ( CAS, CCS, HDB3 etc )

– ISDN / PRI

– MFC / R2

– FXO / FXS

– SS7

– TDMoE ( Time Division Multiplexing over Ethernet )



: Development with Asterisk Extension Language :

AEL is a pseudo-scripting language to control incoming and outgoing calls in the Asterisk PBX.

Common development path is:

– Specify in configuration file which function to execute when new incoming call arrives from specific device.

– Match the dialed number against valid patterns list.

– Start executing applications based on the dialed number and identity of the caller.



: Development with Asterisk Extension Language : Number matching is done using the following patterns:

– X• X means any digit from 0 to 9.

– N• N means any digit from 2 to 9.

– [123]• Brackets are used to group random digits. In this case it

means any number 1, 2 or 3.

– [1-7]• Brackets with 2 numbers separated by a dash represents a

range of numbers. In this case any number from 1 to 7.

– s• The 's' is a special pattern used in devices that do not send a

number to the PBX, like FXS devices.




Hello World AEL

context default {

_XXX => {

Answer(); Playback(hello-world); Hangup();

};

};




Variables are defined and used in a similar way to other scripting languages:

– myvariable=myvalue;

– ${myvariable}

Substring can be obtained using ${var:0:3}

String length: ${LEN(${var})}

String concatenation: ${myvar1}${myvar2}



: Development with Asterisk Extension Language : Matching different patterns.

context common_users {

_XX => { Answer(); Dial(SIP/${EXTEN}); Hangup(); };

_XXXX => { Answer(); Dial(IAX2/${EXTEN});

Hangup(); };

_0NX => { Answer();

Dial(Zap/g1/${EXTEN}); Hangup();

};

};




Some applications, like Dial(), return status through setting special variables, like ${DIALSTATUS}.

Other functionality is offered through the use of special variables, some of the most used are:

– ${EXTEN}: current dialed number

– ${LANGUAGE}: two-letter language code

– ${CHANNEL}: current channel name

– ${EPOCH}: Unix time

– More ...



: Development with Asterisk Extension Language : Catching Dial() return value.

context dial_catcher { _X. => { Answer(); Dial(IAX2/user:[rsakey]@myvoipserver.host.com/${EXTEN});

switch (${DIALSTATUS}){

case BUSY:NoOp(Peer is busy!);break;

case CHANUNAVAIL:NoOp(Failure, channel is unavailable);break;

case NOANSWER:NoOp(Nobody answered on the other side!);break;

default:NoOp(Dial status is: ${DIALSTATUS});

};Hangup();

}; };



: Development with Asterisk Extension Language : Creating and using macros.

macro handle-dial-response( response, peer ) { switch(${response}) {

case NOANSWER:NoOp(No response from ${peer});break;

default:Playback(invalid);NoOp(Yikes! Don't know what to do!);

};};

context default {_X. => {

Answer();Dial(SIP/test);&handle-dial-response(${DIALSTATUS},test);Hangup();

};};




Commonly used Asterisk applications:

Answer()

Dial()

Playback()

Background()

Playtones()

SetLanguage()

Bridge()

MeetMe()

Festival()



: Asterisk Gateway Interface :

AGI() is a special Asterisk application to control call flow through any programming language using stdin and stdout.

AGI launches a whole new process that is connected to Asterisk thread only through file descriptors 1 and 2.

– Anything your program writes to STDOUT will be interpreted by Asterisk as a command.

– Anything your program reads from STDIN is a command response sent by Asterisk.

Enabling “agi debug” will help to detect errors.




Hello World AGI in PHP

context default {

_XXX => {

Answer(); AGI(hello-world.php); Hangup();

};

};

<?php$s = “”;

do {

$s = fread(STDIN, 1024);

} while ( FALSE === strpos($s, “agi_accountcode:”) );

print “EXEC Playback hello-world”;

?>




Asterisk send call information to the AGI program, each line in format “agi_parameter: value\n”

Most relevant values are:– agi_request: Name of the called program.

– agi_channel: Name of the created channel.

– agi_language: Language defined for the channel.

– agi_type: Type of channel technology ( SIP, IAX2 etc).

– agi_uniqueid: Unique id generated for this call.

– agi_callerid: Numeric caller ID.

– agi_dnid: Requested number.




Reading call information values.

#!/usr/bin/php <?php$call_env = array();do{ $read_string = fread(STDIN, 1024); $read_lines = explode("\n", $read_string); foreach ( $read_lines as $read_line ) { if ( FALSE !== strpos($read_line, ":") ) { list($parameter_name, $parameter_value) = explode(":", $read_line, 2); $parameter_value = trim($parameter_value); $call_env[$parameter_name] = $parameter_value; } }} while ( !array_key_exists('agi_accountcode', $call_env) );

print "EXEC SayDigits {$call_env['agi_dnid']}\n";print "EXEC DIAL SIP/{$call_env['agi_dnid']}\n";?>




Asterisk sends a command response that programs can read from standard input.

Response format: “200 result=<res> (optional data)”

Common responses:– 200 result=1

– 200 result=1 (some variable value)

– 200 result=0 (timeout)




Reading response values.

print "GET VARIABLE EXTEN\n";$response = fread(STDIN, 1024);eregi($response_regexp, $response, $matches);$result = $matches[1];$data = isset($matches[2]) ? $matches[2] : ''; file_put_contents('/tmp/agi.log', "result is: {$result}\ndata is: {$data}", FILE_APPEND);



: Asterisk Manager Interface :

AMI is a TCP/IP service exposed by Asterisk.

Clients can send commands to Asterisk and read the command results from the TCP socket.

In contrast with AGI, AMI is an interface to the whole PBX, not to just 1 channel in the PBX.

Clients can listen for telephony events like:

– New incoming / outgoing call.

– DTMF digits received / sent.

– Call progress ( Ringing, Answered, Cancelled etc. ).

– Device registration.




3 different AMI messages can be sent/received.– Action

– Response

– Event

Actions are sent by clients, with an optional ActionID.

Response to the action is sent by Asterisk, if an ActionID was provided it is included in the response message.

Event messages are sent by Asterisk at any time.




AMI messages have the following format:

<Message type>: <Message value><CRLF>

<Header1>: <Header 1 value><CRLF>

<Header2>: <Header 2 value><CRLF>

<Header N>: <Header N value><CRLF>

<CRLF>




Commonly used AMI commands.– Login. Authenticate to start a manager session.– Originate. Starts a new call to the specified end point.– PlayDTMF. Send DTMF on the specified channel.– Redirect. Transfer 1 or 2 legs of the call.– Monitor. Start to record audio on the specified channel.– StopMonitor. Stop audio recording on the specified

channel.– Hangup. Hangup the specified channel.– Logoff. End manager session.




Login into the Asterisk Manager Interface.

/* open the manager socket */$manager_socket = fsockopen('localhost', 5038, $errno, $errstr);/* read the Asterisk welcome message */$welcome = fread($manager_socket, 1024);print $welcome . "\n";

/* Write Login Action data in several steps for clarity */fwrite($manager_socket, "Action: Login\r\n");fwrite($manager_socket, "Username: test\r\n");fwrite($manager_socket, "Secret: test\r\n");fwrite($manager_socket, "\r\n");

/* read login response */$response = fread($manager_socket, 1024);print $response . "\n";




After login, we can originate a call to any end point.

/* send Originate Action */ fwrite($manager_socket, "Action: Originate\r\n"); fwrite($manager_socket, "Channel: SIP/33\r\n"); fwrite($manager_socket, "Context: hello-world\r\n"); fwrite($manager_socket, "Exten: s\r\n"); fwrite($manager_socket, "Priority: 1\r\n"); fwrite($manager_socket, "\r\n"); $response = fread($manager_socket, 1024); print $response;

Call will be originated to end point SIP/33 and connected to hello-world context in the PBX to start application execution.




Originate is useful for applications like:– Click to dial ( web pages or stand alone applications ).– Predictive dialer.– Event notification system.– SPIT ( Spam Over Internet Telephony ).



: Conclusion :

PBX in software, either open source or proprietary, is far more flexible than hardware/firmware only based systems.

Flexibility makes telephony systems far more flexible, but probably far more “crackable” as well.

VoIP industry will require innovative applications to serve users and secure networks, it is an emerging industry about to make a boom. IBM will be part of it, will you?



: References :

http://www.voip-info.org/

http://www.asterisk.org/

http://www.asteriskdocs.org/

http://www.freeswitch.org/



: Questions Session :



: Contact Information :

Moisés Humberto Silva Salmerón

[email protected]

[email protected]

http://www.moythreads.com/

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