Rapid VoIP Application Development On Linux

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Rapid VoIP Application Development On Linux

• Moisés Humberto Silva Salmerón
• System i Access for Linux Development

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Agenda Demystifying VoIP

• What VoIP is not
• What is VoIP?
• Consequences of VoIP
• What is a PBX?
• Common VoIP solutions
• Next generation VoIP solutions

3
Agenda Application Development On Asterisk

• What is Asterisk?
• Development with Asterisk Extension Language
• Asterisk Gateway Interface
• Asterisk Manager Interface

4
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.

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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.

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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.

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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.

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Common VoIP solutions

• CISCO Call Manager.
• Avaya.
• Nortel.
• 3COM.

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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.

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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.

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Application Development On Asterisk
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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 ).

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What is Asterisk?

• 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 )

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What is Asterisk?

• Non-IP supported technologies include
• E1 / T1 ( CAS, CCS, HDB3 etc )
• ISDN / PRI
• MFC / R2
• FXO / FXS
• SS7
• TDMoE ( Time Division Multiplexing over Ethernet )

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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.

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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.

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Asterisk Extension Language

• Hello World AEL

context default _XXX gt
Answer() Playback(hello-world)
Hangup()
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Asterisk Extension Language

• Variables are defined and used in a similar way
  to other scripting languages
• myvariablemyvalue
• myvariable
• Substring can be obtained using var03
• String length LEN(var)
• String concatenation myvar1myvar2

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Asterisk Extension Language

• Matching different patterns.

context common_users _XX gt
Answer() Dial(SIP/EXTEN)
Hangup() _XXXX
gt Answer()
Dial(IAX2/EXTEN) Hangup()
_0NX gt Answer()
Dial(Zap/g1/EXTEN) Hangup()

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Asterisk Extension Language

• 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 ...

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Asterisk Extension Language

• Catching Dial() return value.

context dial_catcher _X. gt
Answer()
Dial(IAX2/userrsakey_at_myvoipserver.host.com/EX
TEN) 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()

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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. gt
Answer() Dial(SIP/test) handle-dial-res
ponse(DIALSTATUS,test) Hangup()

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Asterisk Extension Language Commonly used
Asterisk applications

• Answer()
• Dial()
• Playback()
• Background()
• Playtones()
• SetLanguage()
• Bridge()
• MeetMe()
• Festival()

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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.

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Asterisk Gateway Interface
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Asterisk Gateway Interface

• Hello World AGI in PHP

lt?php s do s fread(STDIN,
1024) while ( FALSE strpos(s,
agi_accountcode) ) print EXEC Playback
hello-world ?gt
context default _XXX gt
Answer() AGI(hello-world.php)
Hangup()
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Asterisk Gateway Interface

• 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.

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Asterisk Gateway Interface

• Reading call information values.

!/usr/bin/php lt?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_envparameter_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
" ?gt
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Asterisk Gateway Interface

• Asterisk sends a command response that programs
  can read from standard input.
• Response format 200 resultltresgt (optional
  data)
• Common responses
• 200 result1
• 200 result1 (some variable value)
• 200 result0 (timeout)

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Asterisk Gateway Interface

• Reading response values.

print "GET VARIABLE EXTEN\n" response
fread(STDIN, 1024) eregi(response_regexp,
response, matches) result
matches1 data isset(matches2) ?
matches2 '' file_put_contents('/tmp/agi.lo
g', "result is result\ndata is data",
FILE_APPEND)
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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.

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Asterisk Manager Interface
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Asterisk Manager Interface

• 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.

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Asterisk Manager Interface

• AMI messages have the following format

ltMessage typegt ltMessage valuegtltCRLFgt ltHeader1gt
ltHeader 1 valuegtltCRLFgt ltHeader2gt ltHeader 2
valuegtltCRLFgt ltHeader Ngt ltHeader N
valuegtltCRLFgt ltCRLFgt
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Asterisk Manager Interface

• 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.

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Asterisk Manager Interface

• 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"
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Asterisk Manager Interface

• 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.

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Asterisk Manager Interface

• 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 ).

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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?

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References http//www.voip-info.org/http//
www.asterisk.org/http//www.asteriskdocs.org/htt
p//www.freeswitch.org/
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Questions Session
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Contact Information Moisés Humberto Silva
Salmerónmoyhu_at_mx1.ibm.commoises.silva_at_gmail.com
http//www.moythreads.com/