VoIP - Chetan Vaity
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Transcript VoIP - Chetan Vaity
Introduction to VoIP
Chetan Vaity
August 2006
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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:
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
SIP messages: INVITE, ACK, BYE, REGISTER
SIP responses:
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
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
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
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
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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
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
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
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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