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Voice over Internet Protocol (VoIP) technology

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Part I - Re-cap of Basics

• What is a protocol?• Telephony• Circuit switching• Important technical terms• Public switched telephone network (PSTN)• Internet Protocol (IP) suite• Internet Protocol networks• Packet switching• What is VoIP?• What is the need for VoIP?• Growth opportunity for VoIP

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What is a Protocol?

• A protocol is a special set of rules that end points in a telecommunication connection use when they communicate

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Telephony• Telephony is “communicating at a distance”• It is “circuit switched”, i.e., there is dedicated channel for

exchange of voice and signaling throughout the conversation

• Reliable delivery• End to end

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Circuit switching

A

B

‘A’ dials ‘B’‘B’ rings

End to end path setup

Call established

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Important technical terms

• Media – refers data / audio / video• Gateway – a network element that interconnect

two disparate networks such as PSTN and IP networks

• Signaling – controls that govern how a media stream is set up, maintained, and gracefully discontinued

• TDM – Time Division Multiplexing (used in telecom networks)

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Public Switched Public Switched Telephone Network (PSTN)

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Internet Protocol (IP) Suite

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Internet Protocol (IP) networks

• Use Internet Protocol for communication of data across “packet switched” network

• Characteristics of IP are– Connectionless– Best effort– Unreliable– Out of order delivery

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Packet switching

A

B

In this IP network we shall examine how packets from computer ‘A’ travels through the IP network and reach computer ‘B’

The first packet takes the red coloured pathThe second packet takes green coloured path Therefore, path taken by a packet in an IP network changes according to conditions prevailing in the network at a particular time (eg: congestion, failure etc)

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What is VoIP?

• VoIP is the transmission of voice traffic in packets using IP as the transport protocol

• It is the merger of telephony and IP worlds together

IP network

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What is the need for VoIP?

• Integration of voice and data• Universal presence of IP• Maturation of technologies• Bandwidth consolidation• The shift to data networks

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Growth opportunity for VoIP

By 2007, internationalVoIP expected to growto 127B, representing54% of all internationaltraffic, including TDMTraffic (IDC IP Telephony Market, 2002)

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Part II - Voice processing in VoIP

• Voice signal• Digitization• Compression• Transmission• VoIP media stream• Sampling error• Sampling rate• Packet delivery in VoIP

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Voice signal

The human voice (analog in nature) impacts the diaphragm of the mouth piece of handset of the telephone.

The transducer present inside the mouth piece converts this analog sound signal to a voltage signal similar in shape, amplitude and timing as shown in figure

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Digitization

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Compression

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Transmit

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VoIP media stream

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Sampling error

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Sampling rate

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Packet delivery in VoIP

A

B

Voice signal generated at ‘A’This signal is digitizedCompressedTransmittedReception at ‘B’ – note the packets reach ‘B’ unordered

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Voice over packet data flow

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Part III - VoIP protocols

• Main types of VoIP protocols• Diagrammatic representation of VoIP protocols• H.323• MGCP / Megaco (H.248)• SIP• SIP vs H.323• VoIP signaling protocol standards compared• RTP• RTCP• Converged telephony network• VoIP protocol stack

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Main types of VoIP protocols

• Call control / signaling • H.323 by ITU-T • SIP (Session Initiation Protocol) by IETF

• Call control / signaling, Gateway control • MGCP (Media Gateway Control Protocol)• Megaco/H.248

• Bearer (carries media)• RTP (Real-Time Protocol) • RTCP (Real Time Control Protocol)

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Diagrammatic representation of VoIP protocols

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H.323

• VoIP signaling protocol• ITU standard and is a protocol suite• Takes a more telecommunications-oriented approach• 90%+ of all Service Provider VoIP networks

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H.323 components

Terminal

Video/audio/data client

MCU (Media Control Unit)

Conference control

Content mixing

Gateway

Protocol translation

Gatekeeper

Address resolution

Admission control

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H.323 call flow

PSTN

Hello

POP(Country B) Other Carrier

PSTN

PSTN

VoIPNetwork

PSTN

(2) IVR prompt

(1) User Dials Access Number

Please enter yourCalling Card Number and PIN

(4)AAA response

(3) AAA query

1st legAccess call

Billing ServerBilling Server

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H.323 call flow

PSTN

Hello

POP(Country B) Other Carrier

PSTN

PSTN

VoIPNetwork

PSTN

(5) IVR prompt

(6) User Dials Destination Number

(7) H.323 Call Setup

Two StageDialling

(8) PSTN CallSetup

1st legAccess call

Billing ServerBilling Server

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H.323 call flow

PSTN

Hello

POP(Country B) Other Carrier

PSTN

PSTN

VoIPNetwork

PSTN(10) H.323 Call Answered

(9) PSTN CallAnswered

Hello

(11) Billing Start

1st legAccess call

2nd legIP Transport

3rd legTermination call

Billing ServerBilling Server

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H.323 call flow

PSTN

Hello

POP(Country B) Other Carrier

PSTN

PSTN

VoIPNetwork

PSTN(14) H.323 CallDisconnect

(15) Disconnect

(13) Billing Stop

1st legAccess call

2nd legIP Transport

3rd legTermination call

(12) Disconnect

Goodbye

Billing ServerBilling Server

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MGCP / Megaco (H.248)

• Protocols that have been defined for communication between media gateway controllers and media gateways. Commonly used are – Media Gateway Control Protocol (MGCP)– H.248 (ITU-T) or MEGACO (IETF)

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SIP

• Another VoIP signaling protocol• IETF RFC2543• Takes an Internet-oriented approach• A text-based protocol

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SIP components

Clients: User Agent Client (UAC) / User Agent Server (UAS) Originate & Terminate SIP requests

Typically an endpoint will have both UAC & UAS, UAC for originating requests, and UAS for terminating requests

Servers: Proxy Server - relays call signaling, i.e. acts as both client

and server, operates in a transactional manner, i.e., it keeps no session state

Redirect Server - redirects callers to other servers Registrar Server - accept registration requests from users,

maintains user’s whereabouts at a Location Server Location Server

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SIP service

SIP User Agents

Registrar Redirect Location

SIP Proxy

SIP Servers/Services

REGISTER“Here I am”

INVITE“I want to talk to another UA

Proxied INVITE“I’ll handle it for

you”

“Where is this name/phone#?”

3xx Redirection“TAhey moved, try this address”

SIP User Agents

SIP-GW

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SIP methods

Basic messages sent in the SIP environment

REGISTER: UA registers with Registrar Server INVITE: request from a UAC to initiate a session ACK: confirms receipt of a final response to INVITE BYE: sent by either side to end a call CANCEL: sent to end a call not yet connected OPTIONS: sent to query capabilities outside of SDP

Answers to SIP messages

1XX – information messages (100 – trying, 180 – ringing, 183 – progress) 2XX – successful request completion (200 – OK) 3XX – call forwarding 4XX – error 5XX – server error 6XX – global failure

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Basic SIP call flow

SIP UA1 SIP UA2

INVITE w/ SDP for Media Negotiation

100 Trying

180/183 Ringing w/ SDP for Media Negotiation

200 OK

200 OK

BYE

MEDIA

ACK

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SIP registration process

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SIP operation in proxy mode

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SIP operation in redirect mode

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SIP vs H.323

SIP H.323

Encoding textual binary

Architecture

SIP is modular because it covers basic call signaling, user location, and registration.  Other features are in other separate orthogonal protocols

H.323 covers almost every service, such as capability exchange, conference control, basic signaling, QoS, registration, service discovery, and so on.

Complexity adequate: HTTP-like protocol high: ASN, use of several different protocols (H.450, H.225.0, H.245)

Extensibility the protocol is open to new protocol features

ASN.1 vendor specific 'nonstandardParam' at predefined positions only

Use in 3gpp yes no

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VoIP signaling protocol standards compared

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RTP

• The challenge for the designers of RTP, was to build a mechanism for robust, real-time media delivery above an unreliable transport layer (UDP).

• RTP was developed by the Audio/Video Transport working group of the Internet Engineering Task Force (IETF). RTP is defined by the IETF proposed standard RFC 1889 published in January 1996. It has been adopted by the International Telecommunication Union (ITU) as part of the H.323 series recommendations, and by several other standards organizations.

• In the TCP/IP model it is hard to say in which layer RTP is in. On the one hand, it looks as an application layer protocol since it runs in user space and is linked to the application program. On the other hand, it is a generic, application independent protocol that just provides transport facilities, so it looks like a transport protocol. The best description would be that RTP is a transport protocol implemented in the application layer.

• Designed to carry a wide variety of data (voice, audio, video)

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RTP message format0 1 3 8 16 31

VER P X CC M PTYPE SEQUENCE NUMBER

TIMESTAMP

SYNCHRONIZATION SOURCE IDENTIFIER

CONTRIBUTING SOURCE ID

…...

VER : Version(2 bits) P : Padding(1 bit) X : Extension header(1 bit)

CC : No. of contributing sources(4 bits) M : Periodic Marker (1 bit)

PTYPE : Payload Type(7 bits)

SEQUENCE NUMBER : Sequence no. of message(16 bits) - Is used to identify packets, and to provide an indication to the receiver of packets are being lost or delivered out of order.

TIMESTAMP : Timestamp of message(32 bits) - Denotes the sampling instant for the first octet of media data in a packet, and it is used to schedule playout of the media data.

Synchronization source identifier (SSRC): This is chosen by the participants at random when they join the session.

Contributing source identifier (CSRC) : This is chosen corresponding to the SSRC of the participant who contributed to the packet

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RTP Encapsulation

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RTCP

• RTCP provides out-of-band communication (such as periodic reporting of information such as reception quality feedback, participant identification, and synchronization between media streams) between the endpoints.

• RTCP allows senders and receivers to transmit a series of reports to one another.

• Although data packets are typically sent every few milliseconds, the control protocol operates on the scale of seconds.

• RTCP messages are encapsulated in UDP datagrams.• UDP port number used is one greater than the port

number of the associated data stream in RTP.

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RTCP message formatV P IC PT Length  

           

Format-specific information

           

           

          Padding if P=1

V – Version(2 bits) - Current version is 2.

P- Padding(1 bit) – If set indicates indicate that the packet has been padded.

IC – Item count – Indicates the number of items included in the packet.

PT - Packet type – Identifies the type of information carried in the packet (five standard packet types).

Type 200: Sender report – senders periodically send these messages to provide an absolute timestampType 201: Receiver report – receivers periodically send these messages informing the sender on the condition of receptionType 202: Source description message – provide general information about the user who owns and controls the sourceType 203: Bye message – is used by sender to end a streamType 204: Application specific message – allow applications to define their own message type (eg: subtitles)

Length – Denotes the length of the packet contents following the common header.

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Converged telephony network

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VoIP protocol stack

Physical Physical

Ethernet, PPP, FR, ATM Data Link

IP Network

TCP UDP Transport

RTP, RTCP Session

Voice Application / Presentation

TCP/IP OSI Model

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Part IV - VoIP architectures

• Centralized architecture• Distributed architecture

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Centralized architecture

• Intelligence is in the network and endpoints are relatively dumb

• Centralizes management, provisioning and call control

• Similar to PSTN• Critics claim it stifles innovation of endpoint

features

e.g. MGCP / Megaco / H.248

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Distributed architecture

• Network intelligence distributed between• Endpoints and call-control devices

Endpoints – IP phones, VoIP G/W, PCs Call control – gatekeepers (H.323) Proxy or redirect servers

(SIP)

• Flexible, easy to add new services• More complex

e.g. H.323, SIP

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Part V - Performance issues in VoIP

• Delay• Jitter• Packet Loss• Echo• Bandwidth• Reliability• Security

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Delay

• Average time a packet takes to make its way through a network end to end

• Major components include Propagation delay & Processing delay

• Packets exceeding a set delay are dropped

Coding delay

Queuing delay

Transmission delayPropagation delay

POTS POTS

Jitter buffer delay

Decoding delayIP Network

Threshold of Delay for VoIP is 150 ms Threshold of Delay for VoIP is 150 ms

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Jitter

• Jitter is variation in packet arrival time• Due to the nature of packet networks, packets can travel

from a source to a destination using different paths resulting in different travel delay

• Speech samples have to be played back at regular intervals (sampling rate). Otherwise, a severe degradation in the speech quality can take place

• A delay jitter buffer is used to reorder the packets and absorb the delay jitter caused by the network.

• The larger the buffer the better is the protection from delay jitter. However, this will result in larger delays

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Jitter buffer

in

Delay too bigRisk of overflow

Delay

out

in

Delay too smallRisk of empty

Delay

out

in

Ideal case

out

Delay

Jitter Protec-tion

Ideal Jitter Buffer Size for VoIP is 60 msIdeal Jitter Buffer Size for VoIP is 60 ms

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Packet Loss

• Packet loss is caused by buffer/queue overflow within the network or by late packet arrival at the receiver or by network failures

• For real-time interactive applications like voice, this means the signal must be output without those packets.

• Packet Loss creates gaps in voice communications, which can result in clicks, muting, or unintelligible speech.

• What can be done to minimize lost packets?

– QoS classification to expedite voice packets

– Longer jitter buffer (trade off between delay and distortion)

– Call admission control to prevent congestion

• Packet loss is caused by buffer/queue overflow within the network or by late packet arrival at the receiver or by network failures

• For real-time interactive applications like voice, this means the signal must be output without those packets.

• Packet Loss creates gaps in voice communications, which can result in clicks, muting, or unintelligible speech.

• What can be done to minimize lost packets?

– QoS classification to expedite voice packets

– Longer jitter buffer (trade off between delay and distortion)

– Call admission control to prevent congestion

Maximum Tolerable Packet Loss is 3%Maximum Tolerable Packet Loss is 3%

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Packet Loss (contd.)

• We can make voice transmission robust to small amounts of packet loss by using Packet Loss Concealment (PLC) algorithms

• These are algorithms that smooth over the gaps in the speech• Some codecs have a built-in PLC feature, while external PLC is

added to other codecs• Lost packets are handled by one of the following PLC approaches:

– Replacing lost packet by a silence packet (no speech)– Repeating the previous packet– Skipping the lost packet– Inserting a noise packet with the proper energy level & spectrum– Most vocoders have internal packet concealment techniques

that optimize the speech quality• PLC can help for short losses, not effective for long bursts (> 3 or so

packets - 40-60 ms of speech )

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Echo and echo control

• Echoes are caused by coupling between transmit and receive paths (“reflection”)

• The effect of the echo on the quality of speech depends upon the magnitude of the echo and the delay at which it occurs.

• Echoes are more problematic in VoIP due to the higher delays• Echo cancellation is critical to perceived voice quality

• Echoes are caused by coupling between transmit and receive paths (“reflection”)

• The effect of the echo on the quality of speech depends upon the magnitude of the echo and the delay at which it occurs.

• Echoes are more problematic in VoIP due to the higher delays• Echo cancellation is critical to perceived voice quality

Reflection

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Bandwidth

• Bandwidth is the raw data transmission capacity of a network

• Bandwidth required per VoIP call will depend on encoding standard used, header compression, and payload size

• For VoIP, bandwidth requirements are usually more constant e.g. G.711 VoIP average bandwidth required is 100 kb/s

• Bandwidth for voice services and associated signaling must take priority over that of best-effort Internet traffic

• Bandwidth is the raw data transmission capacity of a network

• Bandwidth required per VoIP call will depend on encoding standard used, header compression, and payload size

• For VoIP, bandwidth requirements are usually more constant e.g. G.711 VoIP average bandwidth required is 100 kb/s

• Bandwidth for voice services and associated signaling must take priority over that of best-effort Internet traffic

8

7 Voice654 File Transfer32

1

Prot. Control

Broadcast Video

Web Surfing

Strict PriorityAbsolutely goes through,Can starve other apps!

Bandwidth Reduction causes both Delay Bandwidth Reduction causes both Delay and Packet Loss in VoIPand Packet Loss in VoIP

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Reliability

• Traditional phones are powered by phone lines and continue to work during a power outage

• VoIP hardware is subject outages because it is powered by household electricity

• VoIP service outages may be caused by failures within the network

– Failover strategies are desirable for cases when network devices malfunction or links are broken e.g. redundant equipment / links

– IP recovery is slow because it uses protocol to detect and reroute traffic around failures if an alternate path exists

• Traditional phones are powered by phone lines and continue to work during a power outage

• VoIP hardware is subject outages because it is powered by household electricity

• VoIP service outages may be caused by failures within the network

– Failover strategies are desirable for cases when network devices malfunction or links are broken e.g. redundant equipment / links

– IP recovery is slow because it uses protocol to detect and reroute traffic around failures if an alternate path exists

X X

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Security

Security Threat/Attack

Multimedia

Server

IP

A B

a a

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Part VI - Our demonstration

• Web based calls

• Use of soft phones in telephony

• IP phones with PSTN routable numbers

• SMS call back

Proud to present the following research and development work carried out in-house by SLT VoIP engineers

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Offering the virtual number service to Sri Lankan people residing overseas

The demonstrations on display highlights that SLT VoIP platform is capable of offering this value added service. The customer gains the advantage of possessing a telephone service from Sri Lanka while overseas, and call his / her relatives at rates applicable to SLT local phone charges.

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Part VII – Current VoIP services offered

• International call originations from Sri Lanka to A-Z countries worldwide through MAXTALK pre-paid card – available at teleshops. The face values of such cards are LKR 200/- and LKR 400/-.

• International call terminations to Sri Lanka through local VoIP wholesale partners.