copyrights © 2006. all rights reserved. introduction to voip chetan vaity august 2006

Copyrights © 2006. All rights Reserved.

Introduction to VoIP

Chetan VaityAugust 2006

Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com

Lets make some VoIP calls…

Broadvoice

Indian PSTN

US phone

Indian phone

Internet

1

US PSTN

2

3


What is VoIP

Transfer of voice conversations over an IP based network Also known as:

IP Telephony Internet telephony Broadband telephony Voice over Broadband


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


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


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'


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


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


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)


Speech example

Wel 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


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.


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


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


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


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/


Network problems

Delay Jitter Echo Congestion Packet loss Disordered packet arrivals


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


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


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


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


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


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


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


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?


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

copyrights © 2006. all rights reserved. introduction to voip chetan vaity august 2006

Documents

copyright gs lab

media slide

voice slide

broadband slide

number of protocols

family of protocols

qos slide

separate transport protocols