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VOIP with Asterisk
Ed Lally CHLUG meeting April 2007

1

Agenda
 Terminology  The

PSTN and VOIP  About Asterisk  About FreePBX  About Trixbox  Demonstration

2

Terminology
 VOIP:

Voice Over Internet Protocol  PSTN: Public Switched Telephone Network  POTS: Plain Old Telephone Service  TDM: Time-division Multiplexing  PBX: Private Branch Exchange  IVR: InteractiveVoice Response  ACD: Automated Call Distribution  DID: Direct Inward Dial
3

Terminology
 SIP:

Session Initiation Protocol  RTP: Real-time Transfer Protocol  IAX: Inter-Asterisk eXchange Protocol  μLaw: Codec for speech used by PSTN  aLaw: Similar codec used in Europe

4

The PSTN and VOIP
 The

public phone network has been designed to carry voice calls reliably in realtime.
 PSTN

is a circuit-switched network

 Each

call gets a dedicated circuit for the duration  The circuit directly links the two endpoints  No inherent latency, delay, or dropped sound
 Bandwidth

is about 300-3500Hz (good for voice)  Calls can be multiplexed for more efficient transmission (e.g., ISDN at 2 channels or T1 lines at 24 channels)
5

The PSTN and VOIP
 Thepublic IP network (aka the Internet) has been designed to carry data reliably.
 Internet
 Each

is a packet-switched network

connection shares the network with other packets  Different packets may take different paths to the same destination  Packets may arrive late, in the wrong order, or not at all
 Bandwidth

is variable based on physical connection layer, time of day, etc.
6 The PSTN and VOIP


VOIP attempts to send voice over the public IP network.
 





Voice channels are encoded and packetized for transmission Receiver must reassemble packets and decode them  Many implement a “jitter buffer” to compensate for small network delays Connections use Real-Time Transfer Protocol to reduce delivery time  Tradeoff is that sender doesn’t know if packetshave been received VOIP packets can leverage QOS (Quality of Service) flags if the traversed networks support them

7

About Asterisk
 Officially,

Asterisk is an Open Source hybrid TDM and packet voice PBX and IVR platform with ACD functionality. Mark Spencer, Asterisk Inventor

8

About Asterisk


Unofficially, Asterisk is quite possibly the most powerful, flexible, andextensible piece of integrated telecommunications software available. Its name comes from the asterisk symbol, *, which in UNIX (including Linux) and DOS environments represents a wildcard, matching any filename. Similarly, Asterisk the PBX is designed to interface any piece of telephony hardware or software with any telephony application, seamlessly and consistently. Mark Spencer, Asterisk Inventor
9 Why Use Asterisk?
 Allows
  

you to implement a phone system at low cost
Software is F/OSS Can use commodity hardware Compare to Nortel’s Business Communications Manager: combines decade-old technology for $515K

 Implement
 

the system you want

Extremely flexible configuration Can be extended through code or scripts
10

Why Use Asterisk?
 Transition
 Use
 Canto VOIP at your pace

analog cards to connect to PSTN

use anything from POTS to BRI to Channel Bank

 Use

ATAs (e.g., Linksys or Sipura adapters) to connect regular phones  Use soft phones for pure VOIP  Mix-and-match

11

Why Not Use Asterisk?
 If

you want a pure VOIP solution, you need to shop around for VOIP providers.  Support among VOIP providers varies.  SIPdifficulties with firewalls and NAT  Fax support not fully baked-in

12

Asterisk Concepts








Channel: The medium over which a call arrives or leaves (e.g., analog line, SIP connection, etc.) Dialplan: The configuration that describes how to handle all calls Context: A block of instructions in the dialplan that describes how to handle calls on a particular channel Extension: An...
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