Transcript Asterisk - KReSIT Convergence

Asterisk and VoIP issues
Chetan Vaity
March 2007
Copyrights © 2006. All rights Reserved.
What is Asterisk
 Asterisk is an open source software IP PBX
 Digium is the primary developer and sponsor of Asterisk
 In addition, Digium develops PBX hardware and provides contract support
and development services for Asterisk
 Protocols
 SIP (Session Initiation Protocol)
 SIP is primarily used in setting up and tearing down voice calls
 Media protocols
 RTP
 RTCP
Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com
Getting started…
 Get a Linux box
 Install Asterisk
 Get source from http://www.asterisk.org/
 Create SIP users by adding entries to
sip.conf
 Modify the dialplan by editing
extensions.ael
 Install softphones on client computers
and register
 Try X-Lite 3.0 from http://www.xten.com/
 Make a call!
Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com
Getting started…
 Example of a user entry in sip.conf
 [user1]
secret=123user1
type=friend
host=dynamic
context=sipextensions
mailbox=301
callerid="user1" <301>
dmtfmode=rfc2833
canreinvite=yes
 Example dialplan in extensions.ael
 context sipextensions {
301 => {
Dial(SIP/user1);
Hangup();
};
302 => {
Dial(SIP/user1);
Hangup();
};
};
Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com
Asterisk features
 Connect to the PSTN world
 Hardware - PCI cards from Digium to connect to analog PSTN or digital
(PRI) lines
 Asterisk can then act as a VoIP to PSTN gateway.
 Feature-rich and Extensible
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Extension patterns
Conference, Voicemail
Recording
TTS integration with Festival
AGI – Run your own programs!
 Say cricket score
Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com
Voice Quality
 Bad links
 Packet loss
 Media goes over UDP
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Jitter
Delay
Disordered packet arrivals
Codecs should be resistant to packet loss
 iLBC used by Skype, GoogleTalk etc. (developed by GIPS)
Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com
Delay and Jitter
 A delay of less than 150 ms is acceptable and usually goes unnoticed
by humans
 With delay greater than 400 ms, conversation starts becoming irritating
 Coder delay is the time taken to compress a block of PCM samples
 This delay varies with the codec used and processor speed
 For G.729, delay is around 30ms
 Packetization delay is the time taken to fill a packet payload with
encoded speech
 Queuing delay and Propagation delay at various network components
 Jitter buffer delay
 Variation in delay of packets is called Jitter
 The effects of jitter can be mitigated by storing voice packets in a buffer
upon arrival, before playing out
 Increases delay by the length of the buffer
Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com
Echo
 Echo in telephony systems is caused by two main phenomena
 Electrical echo due to imperfect impedance matching
 Acoustic echo due to microphone pickup of audio output
 Echo becomes noticeable only when there is a delay between speaking
and hearing your voice echoed. (more than about 50 ms)
 In PSTN calls, there is always echo, but it remains unnoticed because
the delay is quite small
 VoIP intrinsically has packetization, depacketization and processing
delays built into its protocols
 VoIP phones don't cause echo. They just make it audible by introducing an
extra delay
 Echo cancellation: Subtract from the received signal
 Based on the response of the system to a short spike of sound
 Echo cancellation is a hugely CPU-intensive process
Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com
Advantages of VoIP
 Reduction in costs
 Uses the internet for long distance calls
 Uses underutilized existing network capacity
 Functionality
 Especially for computer users – (click on name to call)
 Merging of Data and Voice infrastructures
 No need for separate cabling
 Mobility
 Wherever you are connected to the Internet, you can receive VoIP calls
Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com
Disadvantages of VoIP
 Quality
 Due to low/variable bandwidth
 Echo
 Internet connection
 VoIP usage is entirely dependent on the quality, reliability and speed of the
internet connection
 If the net is down, you have no telephony
 Power
 No phone calls in a power outage
 Security (lack of it)
 Anyone listening on the LAN can potentially capture packets and
reconstruct the conversation
Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com