Download VoIP - Chetan Vaity

Introduction to VoIP
Chetan Vaity
August 2006
Copyrights © 2006. All rights Reserved.
Lets make some VoIP calls…
Indian PSTN
Indian phone
2
1
US PSTN
3
Internet
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US phone
Broadvoice
What is VoIP
 Transfer of voice conversations over an IP based network
 Also known as:
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IP Telephony
Internet telephony
Broadband telephony
Voice over Broadband
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Essentials
 What happens in a VoIP call?
 Establish connection with the target
 Various protocols
 Capture voice, digitize and encode
 Codecs
 Transfer over network
 Network issues
 Interface with PSTN
 Decode and reproduce voice
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Protocols
 Signaling protocols
 SIP (Internet Engineering Task Force)
 H.323 (International Telecommunications Union)
 All voice/video communications are done over separate transport
protocols, typically RTP
 Media protocols
 RTP
 RTCP
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Protocols – SIP
Session Initiation Protocol
SIP is primarily used in setting up and tearing down voice or video calls
SIP clients traditionally use port 5060 to connect to SIP servers
SIP acts as a carrier for the Session Description Protocol (SDP), which
describes the media content of the session, e.g. what IP ports to use,
the codec being used etc.
 It is human readable and request-response structured
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 SIP messages: INVITE, ACK, BYE, REGISTER
 SIP responses:
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100 Trying
180 Ringing
200 OK
404 Not found
 SIP shares many HTTP status codes, such as the familiar '404 not found'
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Protocols – H.323
 H.323 is actually a family of protocols
 H.323 ties together a number of protocols to allow multimedia
transmissions over an unreliable packet based network
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H.225 for call control and signaling
H.245 for exchanging terminal capabilities and creation of media channels
H.235 for security
RTP/RTCP for media
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Protocols – RTP (Real-time Transport Protocol)
 Media applications are less sensitive to packet loss, but typically very
sensitive to delays.
 UDP is a better choice than TCP
 RTP generally runs over UDP
 RTP provides
 payload-type identification
 sequence numbering
 timestamping
 It does not guarantee any QoS
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Protocols - RTCP
 Real-time transport control protocol (RTCP) is the counterpart of RTP
that provides control services.
 The primary function of RTCP is to provide feedback on the quality of the
data distribution.
 Statistics on a media connection
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bytes sent
packets sent
lost packets
jitter
round trip delay
 An application may use this information to increase the quality of
service perhaps by limiting data sent or maybe using a low compression
codec instead of a high compression codec
 RTCP uses (RTP port + 1)
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Speech example
Wel
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come
to
G
S Lab
Codecs
 Convert speech to a digital format suitable to be transmitted over the
network
 Most codecs utilize compression to reduce the bandwidth requirement
 But, heavy compression algorithms take time. This adds a delay to the
conversation
 Human speech is a very special signal and its characteristics are
exploited in these algorithms
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Pulse Code Modulation
 A PCM representation of an
analog signal is generated by
measuring (sampling) the
magnitude of the analog signal at
uniform intervals, and then
quantizing it to a code.
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G.711 (µ-law)
8000 samples per second
8 bits per sample
64 kbps
Logarithmic PCM (because the perceived loudness by humans is
logarithmic)
 µ-law: used in North America and Japan
 a-law: used in the rest of the world
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Linear Predictive Coding
 LPC starts with the assumption that a speech signal is produced by a
buzzer at the end of a tube
 The vocal tract (the throat and mouth) forms the tube, which is
characterized by its resonances
 The buzz is characterized by its intensity (gain) and frequency (pitch)
 LPC analysis produces
estimates for the pitch,
gain and a set of
numbers for the
resonances
 Voiced and Unvoiced
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GSM codec
GSM uses linear predictive coding (LPC)
Speech is divided into 20 millisecond units (frames)
LPC parameters are determined for each frame
The number of bits needed to send these parameters is the bit-rate of
the codec
 For GSM, the bit rate is 13kbps
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Comparison between codecs
Codec
Bit rate
Quality (MOS)
G.711
64000
4.1
G.729
8000
3.9
G.723.1
5300
3.6
LPC-10
2400
2.7
Source for wave samples: http://www.signalogic.com/
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Network problems
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Delay
Jitter
Echo
Congestion
Packet loss
Disordered packet arrivals
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Network issues - Delay
 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
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Jitter
 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
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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
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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
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Services
 Packet8, Vonage, Verizon
 A black box with a phone attached
 The user experience is almost indistinguishable from normal PSTN
 The term “VoIP” is not used, instead – “Internet telephone” or “Digital
telephone”
 Broadvoice
 Allow direct connect of SIP phones
 Aimed at tech-savvy clients
 Allows
 Skype
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Rely on the software client on the computer
Peer to peer
Routes calls through other Skype peers on the network
Proprietary, closed source
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Legal Issues
 As the popularity of VoIP grows, and PSTN users switch to VoIP in
increasing numbers, governments are becoming more interested in
regulating VoIP in a manner similar to legacy PSTN services
 In some countries, governments fearful for their state owned telephone
services, have imposed restrictions on the use of VoIP
 In India, it is legal to use VoIP. But it is illegal to have VoIP gateways inside
India. This effectively means, people who have PCs can use it to make a
VoIP call to any number. But if the remote side is a normal phone, the
gateway that converts VoIP call to PSTN call should not be inside India
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Cougar
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What is it?
What can it do?
What software does it use?
How do I make calls?
Whom should I contact if I can’t?
Where to get more info?
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References
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Wikipedia
http://www.linuxjournal.com/article/8424
http://www.cisco.com/warp/public/788/voip/delay-details.html
http://research.edm.uhasselt.be/jori/thesis/onlinethesis/chapter4.html
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