Voice over IPVoice over IP

Andreas Mettis University of Cyprus November 23, 2004

OverviewOverview

What is VoIP and how it works.Reduction of voice quality.Quality of Service for VoIP

VoIP

VoIP (voice over IP - that is, voice delivered using the Internet Protocol) is a term used in IP telephony for a set of facilities for managing the delivery of voice information using the Internet Protocol (IP). In general, this means sending voice information in digital form in discrete packets rather than in the traditional circuit-committed protocols of the public switched telephone network (PSTN).

How VoIP WorksHow VoIP Works

The OSI Model

Analog to Digital Analog to Digital

Voice is nothing but air vibration. The microphone converts this vibration

into an equivalent variation of an electrical current.

The amplitude of this current is measured 8000 times every second.

Each reading is coded in binary (ones and zeros).

Each code is made up of 8 bits.

Codec StandardsCodec Standards

Packet by Packet TransmissionPacket by Packet Transmission

Transport LayerTransport Layer

The Real-time Transport (RTP) Protocol provides end-to-end network transport functions suitable for applications transmitting real-time data such as audio, video or simulation data, over multicast or unicast network services.

RTP does not address resource reservation and does not guarantee quality-of-service for real-time services.

Transport LayerTransport Layer

The User Datagram Protocol (UDP), provides a simple, but unreliable message service for transaction-oriented services.

Each UDP header carries both a source port identifier and destination port identifier, allowing high-level protocols to target specific applications and services among hosts.

Network LayerNetwork Layer

The Internet Protocol (IP), is the routing layer datagram service of the TCP/IP suite. The IP is used to route packets from host to host.

The IP packet header contains routing information and control information associated with datagram delivery.

Data Link/ Physical LayerData Link/ Physical Layer

The Ethernet header is attached to the VoIP frame.

At the Physical Layer the data are sent from the sender to the receiver.

VoIP PacketVoIP Packet

Reduction of voice quality

Mean Option ScoreMean Option Score In order to assess the quality of voice

communications in the presence of impairments, it is crucial to study the individual as well as collective effects of the impairments and produce quantitative measures that reflect the subjective rating that listeners would give.

MOS is valuable in that it addresses the human perceived experience, which is the ultimate measure of interest.

Application LayerApplication Layer

Standard Codec type Rate (Kbps) Frame (ms)

MOS

G.711 PCM 64 4.43

G.729 CS-ACELP 8 10 4.18

G.723.1 ACELP 5.3 30 3.83

G.723.1 MP-MLQ 6.3 30 4.00

Application LayerApplication Layer

Voice Activity DetectionVoice Activity Detection

VAD uses the fact that two communication partners seldom speak at the same time.

Bandwidth saving up to 50%.Difficult to distinguish between ambient

noise and silence in transmission.Voice clipping.

Packet SizePacket Size

DelayDelay

Delay incurred in encoding (Algorithmic delay)

Packetization delay (function of the amount of speech data included in a packet)

Sender to receiver delay 1) Propagation delay 2) Transmission delay 3) Queuing delay

Packet losses and DelayPacket losses and Delay

EchoEcho

Echo is caused by the reflection of signals at the four-to-two wire hybrids. This type of echo is present when a voice call involves a combination of VoIP segment in the Internet and a circuit segment in the switched telephone network.

Another cause of echo is the PC-based phones that are equipped with a microphone and loudspeakers.

Why bother about VoIP?Why bother about VoIP?

MONEY,MONEY,MONEY,MONEY,MONEY,MONEY,MONEY,MONEY!!!!!!!!!!

Quality of ServiceQuality of Service

Algorithms usedAlgorithms used

Echo CancellationLoss Recovery: Forward error correction

adds redundancy information into voice streams for aiding the loss correction.

Error Concealment: A replacement for a lost packet is produced which is similar to the original lost packet. This is possible because voice signals exhibit large amounts of short-term self similarity.

Worst Case DesignWorst Case Design

AdvantagesQoS is guarantee.

DisadvantagesToo expensive.The utilization is very small.

RSVPRSVP The sender sends

the PATH, which describes the traffic that is going to create.

The receiver sends the RESV, that it is used to make reservations at every intermediate node.

The RESV packets are routed using the Reverse Path Algorithm.

Sender 1

Sender 2

PATH

PATH

RESV(merged)

RESV

RESV

Receiver B

Receiver A

R

R

R

R

R

RSVP RSVP

Advantages It is possible to assign bandwidth reliably for

eachVoIP session.Disadvantages Some resources remain not used when VoIP data has burst character. The load of routers becomes high and application

to a very large scale network becomes difficult.

Virtual Private NetworksVirtual Private Networks

Advantages QoS can be high. DisadvantagesUtilization of the network might be low.Might cause starvation for other VoIP

traffic.

Differentiated Services ModelDifferentiated Services Model

Diffserv Diffserv

DSCPDSCP

DSCP – Expedited Forwarding DSCP – Expedited Forwarding

EF – PHB ensures a minimum departure rate, independently of any other traffic attempting to transit across the node.

EF – PHB provides a low loss, low jitter assured bandwidth, end to end service through DS domains.

DSCP - Assured Forwarding DSCP - Assured Forwarding

• Best Effort Forwarding

(green, yellow, red)

Admission ControlAdmission Control Admission control unit makes admission

decision to the new request. Admission Criteria is a set of conditions used to

determine if an incoming call is to be accepted. Network QoS state and flow information are

necessary for the admission control unit.

Combination of Diffserv and Call Combination of Diffserv and Call Admission Admission

SIP proxy observes flow information from the router using SNMP.

When a SIP message arrives from the SIP terminal it decides the acceptability of this new call based on flow information and the SIP message log.

Diffserv Packet Marking RuleDiffserv Packet Marking Rule

Green: Basic data of all communication sessions.

Yellow: Additional data of important sessions.

Red: Additional data of normal sessions.

Behavior of the System Behavior of the System Basic data can be protected from packet loss by

dropping additional data packet of normal communication.

In order to guarantee quality of each session, it is necessary to make VoIP flow less than suitable quantity on each link of the network.

Call Admission MethodCall Admission MethodThree kind of VoIP sessions which exist in a system. Sessions generating data traffic Sessions currently in the signaling stage and generating

future traffic. Sessions currently in the signaling stage, but which will

terminate without generating traffic in the future because of some kind of error.

Call Admission MethodCall Admission Method It is impossible to determine whether the session

currently in the signaling stage will generate traffic or terminate by future error.

The log of SIP INVITE message is used and the worst time processing of SIP signaling is recorded to log as TTL value for each SIP INVITE message.

TTL is the worst time to process SIP signaling and is known from statistical data.

ConclusionsConclusions

VoIP is rather easy to implement but difficult to guarantee QoS.

The combination of Diffserv and Call Admission provide a good mechanism for QoS for VoIP.

VoIP offers a lower QoS compared with the PSTN, and can also offers lower costs to the organizations and people that use it.

Still need to find better solutions for providing QoS for VoIP.

ReferencesReferences [1] Athina P. Markopoulou, Fouad A. Tobagi and Mansour J.

Karam, “Assessing the Quality of Voice Communications over Internet Backbones”, pp747-760, 2003.

[2] Xiuzhong Chen, Chunfeng Wang, Dong Xuan, Zhongcheng Li, Yinghua Min and Wei Zhao, “Survey on QoS Management of VoIP”, Proceedings of the 2003 International Conference on Computer networks and Mobile Computing (ICCNMC’ 03).

[3] Masaaki Noro 1, Takahiro KIKUCHI 1, Ken-ichi BABA 2,Hideki SUNAHARA 1,3, Shinji SHIMOJO, “QoS Support for VoIP Traffic to Prepare Emergency”, Proceedings of the 2004 International Symposium on Applications and the Internet Workshops (SAINTW’04)

[4] http://www.protocols.com/ [5] Dr. Christos Panayiotou lecture notes. [6] Siemens. Information and Communications networks. [7] HiPath 4000 V1.0, IP Distributed Architecture, Service Manual.