Transcript PPS - Florian Consulting Inc.

Asterisk PBX
PSTN to VoIP and never back again
Tomas Florian
Intro
IT Consulting: florien.ca



Linux / Windows interoperation
Application development
High availability hosting
Working with Asterisk for the last year in
small office environments
Beta stage of Asterisk based solution for
communication company launching
service between Canada and India.
Summary
VoIP (the big picture)
Asterisk




Architecture
Channels
Example
Dialplans
Getting Started
Questions
The Big Picture
Some Telco
stuff ??
Good old
internet 
PSTN and the Internet
Asterisk
PSTN and the Internet
PSTN
“Hard wired”
Expensive
Proprietary
Local
Plain Old Reliability
Cell phone service
911 compatibility
vs.
VoIP
Flexible
Cost effective
Open
World wide
Bleeding edge
Not yet
“Hello Calgary … I’m
in Tokyo and I need
help”
Flexibility
Branch offices / Virtual Offices
Telecommuting
Automated agents
Custom applications




Press 1 to have your new emails read to you
Today’s weather forecast
Games
… “just” another interface to the web (audio)
Summary
VoIP (the big picture)
Asterisk

Architecture
Asterisk
Open Source
Created by Mark Spencer (1.0 in 2004)
Sponsored by Digium (Hardware)
Minimum requirements




Linux / FreeBSD
Pentium 300 MHz
128 MB RAM
600 MB Disk
Asterisk
* … good name choice
Connector between all kinds of audio
protocols and technologies





PSTN
SIP/RTP
IAX
H323 (Netmeeting)
…
Architecture
Applications
Codecs
*
Channels
File formats
Architecture
Paging
…
VM
…
G.723.1
WAV
*
GSM
…
MP3
PSTN
SIP
Summary
VoIP (the big picture)
Asterisk


Architecture
Channels
Channels
ZAP
Asterisk
SIP, IAX, H323
SIP: Session Initiation Protocol
VoIP favorite
Wide hardware and software phone support
Hardware phones

Example: BT100
Hardware adapters

PAP2
Software phones

X-Lite
SIP : Introduction
Invite
OK
ACK
BYE
UA - Caller
OK
UA - Callee
Port 5060 UDP
Human Readable Text (similar to HTTP)
SIP : Introduction
Invite
OK
ACK
Caller
RTP
Callee
RTP (Real-time Transport Protocol)


Separate UDP stream (media path)
Sends voice data
SIP Proxy and Registrar
Register
Register
SIP Proxy and Registrar
Invite
Invite
SIP Proxy and Registrar
RTP
Other VoIP Channels
IAX (Inter Asterisk eXchange)
H323 (Netmeeting)
Channels
ZAP
Asterisk
SIP, IAX, H323
ZAP: Interface to the PSTN
Reuse existing PSTN infrastructure
Downstream from Telco


lines from Telco terminating at Asterisk
FXO
Be the Telco


lines from Asterisk going to existing phone sets
FXS
Digium TDM400, X100P (fancy voice modem)
Summary
VoIP (the big picture)
Asterisk



Architecture
Channels
Example
Office PBX Example
Internet
PSTN
Office
The Office
SIP interface
Ethernet (LAN): IP 192.168.0.10


VoIP phones on a switch
Each phone needs IP (assigned by DHCP)
Office
PSTN
ZAP interface
1 Line: 222-1234




FXO interface card X100P
Plug in the physical line
One conversation per line
More lines needed for
real-life scenario
PSTN
The Internet
WAN interface instead of LAN


Server - Routable IP 139.142.2.2
The phones (clients) may be behind their own
firewall or NAT but as long
as they can contact the
server
Internet
Office PBX Example
Internet
PSTN
Office
Variation
PSTN
Datacenter
Internet
Office A
Office B
Summary
VoIP (the big picture)
Asterisk




Architecture
Channels
Example
Dialplans
Dialplans
The glue that holds everything together
Scripting language
Matches extension and launches a certain
application
Example
Extension
Application
Arguments
1234,1,Dial(SIP/1234)
Priority
Dialplans
Priorities
1234,1,Dial(ZAP/2|15)
1234,2,Voicemail(u1234)
Patterns
_9XXXXXXX,1,Dial(Zap/2)
Variables
_123XXXX,1,Dial(SIP/{$EXTEN})
Dialplans
Contexts



Security mechanism
Channel
Time of day
Example
[from-local]
_9XXXXXXX,1,Dial(Zap/1)
[from-untrusted]
_91234444,1,Dial(Zap/2)
Logical code blocks … include, etc.
Dialplans
Arithmetic
If,Else,Goto
Time of day
Macros
Applications
Answer
DBget / DBput
Festival
MP3Player
Queue
Record
Meetme
System
Digging Deeper
/etc/asterisk




sip.conf
extensions.conf
zapata.conf
…
Asterisk CLI
ethereal
ngrep
Summary
VoIP (the big picture)
Asterisk




Architecture
Channels
Example
Dialplans
Getting Started
Getting started
Asterisk@Home
Asterisk
AMP
Meetme
Music on hold
Flash panel
Call logs
Sugar CRM
Home automation
Wipes your HD!
Resources
asterisk.org (home)
voip-info.org (wiki)
[email protected] (mailing list)
digium.com (asterisk creators – hardware)
E-bay (hardware)
Books


VoIP Telephony with Asterisk (Paul Mahler)
Asterisk: The Future of Telephony (online)
florien.ca (paid support)
Questions
?
NAT : SIP/RTP
Opening ports (ugly)



5060 UDP
10000 – 20000 UDP
Hard code external IP
STUN (elegant but not 100% reliable)



Detects external IP
Detects external port
Keep alive
Media Proxy

Send all voice through Asterisk
NAT: IAX2
NAT problems? What NAT problems?
Single UDP port
High performance, low overhead
…
Lack of hardware support


DIAX
Digium Iaxy
Codecs
G.711 (ulaw/alaw)

64 Kbps
G.726 (half rate G.711)

32 Kbps
GSM (Cell phone codec)

12 Kbps
G.723.1 (also in Netmeeting)

6.3 or 5.3 Kbps
Many others …
Other Platforms
SipXpbx
SER (SIP Express Router)
Vocal