Multimedia networking Voice/data integration

(1)

Presentation_ID 1

1 Eric Vyncke

Eric Vyncke

Distinguished

Engineeer

Cisco Systems

[email protected]

Multimedia networking

Voice/data integration

ULg VoIP

3 Agenda



‘XXth Century’ voice = Analog thenTime Division Multiplexing (TDM)



‘XXIst Century’ voice

packetization

Quality of service

Signalling

Issues with NAT

Security

(2)

5 Loop

(Local or Station)

+

48v

–

Station

PBX or Central Office

Switch

Loop Start Signaling

T

R

On-hook,

open loop

BELL

+

–

DC Current

Switch

48v

Off-hook,

close loop

BELL

+

–

AC

Ringing

_Switch

Switch

BELL !!

48v

BELL

Ring on-hook

Ans off-hook

Current

sense

Echo in Voice Networks

Delay in

the network

Listener Echo

Talker

Listener

Talker Echo

(3)

ULg VoIP

7 Echo Loss

(dB)

Echo Path Delay

(ms)

Echo Is Unnoticeable

Echo Is Always Present

~20

~200

- 10

- 50

Echo Is a Problem

…

Too Much Echo Is Bad,

but No echo is also bad!!

High Loss

Low Loss

ULg VoIP

8 Speech and

the Telephone Network

300Hz

_4kHz

_16kHz

Po

w

er

/ Vo

lu

m

e

Frequency / Pitch

Human Ear

Response

Telephone

Network

3700Hz voice bandwidth

3400Hz

(4)

ULg VoIP

9 Mean Opinion Score

Source

Impairment

Codec ‘X’

Channel Simulation

“Nowadays, a chicken leg is

a rare dish”

1

2

3

4

5

1

2

3

4

5 Rating

Speech Quality

Level of Distortion

5 Excellent

Imperceptible

4 Good

Just perceptible but not annoying

3 Fair

Perceptible and slightly annoying

2 Poor

Annoying but not objectionable

1 Unsatisfactory

Very annoying and objectionable

MOS of 4.0 = Toll Quality

Summary



Analogue voice technology dates

back to the late 1800s;



Analogue information exchange is based on voltage, current

sense, grounding;



Echo is a fundamental component of Analogue voice and

(5)

ULg VoIP

11 Agenda



‘XX Century’ voice



‘XXI Century’ voice

packetization

Quality of service

Signalling

Issues with NAT

Security

ULg VoIP

12 IP Phones



QoS in phones - standard 802.1p/q



Integrated Ethernet switching



Easy access to new world features

IPv6

GigaEthernet

Video

IEEE 802.1x

(6)

ULg VoIP

13 Inline Power: IEEE 802.3AF

Provides DC Power over Standard Category-5 Ethernet

Inline Power

10/100 Ethernet without Inline Power



IP phone are power hungry and you do not want to have a 220V power

cable



=> get power through the UTP cable

Agenda



‘XXth Century’ voice



‘XXIst Century’ voice

Packetization

Quality of service

Signalling

Issues with NAT

Security

(7)

ULg VoIP

15 Sample rate = 2 x highest frequency

Sample rate = 2 x highest frequency

Sampling

Stage

Analogueue

Audio

Source

Pulse Code Modulation—Nyquist Theorem

...

0010010

1 1110110

01001

...

8,000 samples per second

1 sample = 8 bits;

8000 samples/sec = 64,000

bit/s

Digital Audio Stream

Analogue to Digital Voice

B/W = 300 to 4000Hz

ULg VoIP

16 Speech Compression Techniques

Overview

Waveform Coding

• PCM

Differential Waveform Coding

• DPCM, ADPCM

Source algorithms

(8)

ULg VoIP

17

Mean Opinion Scores

5

4

3

2

1

2

4

8

16

32

64 Kbps

Su

b

je

ct

iv

e

Q

u

al

it

y

(M

O

S)

Hybrid Coders

(LD-CELP &

CS-ACELP)

Vocoders

(Older Technology)

Waveform Coders

(ADPCM)

Score

Quality

Description of Impairment

5

4

3

2

1 Excellent

Good

Fair

Poor

Bad

Imperceptible

Just Perceptible, not Annoying

Perceptible and Slightly Annoying

Annoying but not Objectionable

Very Annoying and Objectionable

Source: A.M. Kondoz, “Digital Speech Coding for Low Bit-Rate Communications Systems”, 1995

4 Bytes

RTP Timestamp

Synchronization Source (SSRC) ID

Sequence Number

Payload

Type

M

CC

V

E

R

RTP/RTCP—RFCs 1889/1890



End-to-end network transport function

Payload type identification—voice, video, compression type

Sequence numbering

Time stamping

Delivery monitoring

(9)

ULg VoIP

20 Header is 40 bytes

IP Header (20) UDP (8)

RTP (12)

_{PAYLOAD : 20}

26 kbps of bandwidth

per call

Compressing RTP Header gives

4-5

_{PAYLOAD : 20}

11 kbps of bandwidth per call

Bandwidth Per IP Call

20ms @ 8kbit/s of compressed voice

ULg VoIP

21 Summary



All voice over the telephone network is somewhat

compressed;



DSPs allow very high compression rates while producing

good quality speech



Silence suppression can deliver additional bandwidth

(10)

ULg VoIP

22 Agenda



‘XXth Century’ voice



‘XXIst Century’ voice

Packetization

Quality of service

Signalling

Issues with NAT

Security

A

First Bit

Transmitted

Last Bit

Received

Network

Sender

Receiver

t

Network

Transit

Delay

Processing

Delay

Processing

Delay

End-to-End Delay

Delay and Voice

PBX

(11)

24 Delay Variation—“Jitter”

t

Sender Transmits

B Receives

C

B

A

C

B

A

SenderA

ReceiverB

Network

d1

d2

D1 = d1

D2 = d2

Jitter

ULg VoIP

25 Delay and Jitter



Delay and jitter are generated when a packet is stored and

forwarded:

by router and switches



Delay is also generated by links

1 microsecond every 200 Km



Jitter is also caused by burst



Jitter requires play-back buffers

(12)

26 Reserve 1

Mbps BW

on this line

I need 1 Mbps

BW and 200

msec delay

This app needs

1 Mbps BW and

200 msec delay

Reserve 1

Mbps BW

on this line

Integrated Services QoS Model

Resource Reservation Protocol

Main use: Call Admission Control

Is the network ready for a new call?

Campus

Backbone

Multimedia

Training

Servers

Order Entry,

Finance,

Manufacturing

Finance

Manager

Remote

Campus

Differentiated Services

Classification

Enforcement

(13)

ULg VoIP

28 Packet Classification Layers

PT

DATA

FCS

PREAM. SFD DA

SA

Layer 2

802.1Q/p

TAG

4 Bytes

3 bits used for COS

(user priority)

Version

Length

ToS

1 Byte

Len

ID

offset

TTL Proto

FCS IP-SA

IP-DA

Data

3 bits called IP Precedence for differentiated services

(DiffServ may use 6 D.S. bits plus 2 for flow ctrl)

Layer 3

IPV4

Version

Length

Traffic

Class

1 Byte

Hop

Limit

IP-DA

Flow

Label

Len

Hdr

IP-SA

Data

6 diff serv code points + 2 for flow control

Layer 3

IPV6

ULg VoIP

29 Evolving Business Requirements

Business Requirements Will Evolve and Expand over Time

Time

Critical Data

Realtime

4-Class Model

Best Effort

Signaling / Control

Call Signaling

Critical Data

Interactive Video

Voice

8-Class Model

Scavenger

Best Effort

Streaming Video

Network Control

Network Management

Realtime Interactive

Transactional Data

Multimedia Conferencing

Voice

12-Class Model

Bulk Data

Scavenger

Best Effort

Multimedia Streaming

Network Control

Broadcast Video

Call Signaling

http://www.cisco.com/en/US/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND_40/QoSIntro_40.html#wp61135

(14)

ULg VoIP

30 ML-PPP queueing algorithm



Fragment large packets



Let small packets:

Use “normal” encapsulation

Interleave with fragmented traffic

Jumbogram

Voice 2

Voice 1

Fragment 4

Fragment 3

Fragment 2

Fragment 1

Voice 2

Voice 1

Collaboration & Presence



Presence augmented Instant Messaging

Who is on-line

Are they busy?

Where are they?



All of this pieces of information

Can be automated

(15)

ULg VoIP

32 Collaboration & Teleconference



High-speed, ubiquitous Internet allows

Cheap (Internet based) communications

Visual interaction

Sharing slides, documents

Seeing others on video

Working on the same document

décembre 12

32

ULg VoIP

33 Collaboration and Telepresence

(16)

ULg VoIP

34 Collaboration and Telepresence



Next step: large HDTV screens, smooth video



Next next step: HDTV replacing walls



Best seen over Youtube



http://www.youtube.com/watch?v=J0jrmTf_0tE

(commercial)



http://www.youtube.com/watch?v=rcfNC_x0VvE

(start at 1

minute, 3D)

décembre 12

34 New Application Requirements

The Impact of HD on the Network



User demand for HD video has a major impact on the network

(H.264) 720p HD video requires twice as much bandwidth as (H.323) DVD

(H.264) 1080p HD video requires twice as much bandwidth as (H.264) 720p

(17)

ULg VoIP

36 Agenda



‘XXth Century’ voice



‘XXIst Century’ voice

Packetization

Quality of service

Signalling

Issues with NAT

Security

ULg VoIP

39 SIP: Session Initiated Protocol



SIP is another VoIP signaling protocol



Web like



Text format messages

Similar to HTTP



Fast call setup



Run over UDP or TCP

(18)

ULg VoIP

40 SIP Basics



SIP is a peer-to-peer protocol where end-devices (User Agents - UAs) initiate

sessions



SIP defines the signaling mechanism



SIP works for voice, video, instant messaging



SIP uses IETF protocols

HTTP 1.1

Session Description Protocol (SDP)

media (RTP)

name resolution & mobility (DHCP & DNS)

application encoding (MIME)



SIP is ASCII text-based:- implementation & debugging

Internet or

private IP network

VoIP Architecture

Based on Session Initiation Protocol

décembre 12

Technologies

SIP Proxy

SIP Trunk

Old Phone

network

Extensio

n

IP Address

2000

192.168.0.1 6000

2001:db8::abba:babe

SIP Clients

1) SIP registration

Ext: 2000

IP: 192.168.0.1

IP: 2001:db8::abba:babe

Ext: 6000

2) Voice

3) External

(19)

ULg VoIP

42 VoIP Pricing...



SIP: Session Initiation Protocol

Used to allow only authenticated device

SIP Proxy Register the IP address of a phone extension

SIP Trunk: gateway to classical analog voice



SIP proxy: free software (Asterisk) on an existing server



SIP trunk: cheap calls fixed price for Europe 5 EUR/month



SIP client on mobile/PC: free



SIP physical phones: 100 EUR

décembre 12

Technologies

- 42

ULg VoIP

43 SIP Commands/Responses



INVITE



CONNECTED



BYE



UNREGISTER



REGISTER



1XX Information



2XX Success



3XX Redirection



4XX Client Error



5XX Server Error



6XX Global Failure

Commands

Responses

(20)

44 SIP Phone

SIP UA / GW

Redirect

Server

Or SIP proxy

INVITE

3xx Redirect

INVITE to Address Returned in Contact: of 3XX response

100 Trying

180 Ringing

200 OK

ACK

BYE

200 OK

SIP Call Flow

What Is 9-1-1 (or 1-1-2 or 9-9-9)?



A simple, easy to remember telephone number that

allows automated call routing to the

local

public safety

agency, based on where you are calling

from



In some jurisdictions (North America) there are many

different destinations;

source

routed



Mostly ubiquitous for residential service



Varying degrees of deployment globally

Enhanced 9-1-1 in North America

European Union current efforts to converge on 1-1-2

India currently has country-wide rollout of 1-0-8

(21)

ULg VoIP

46 Residential 9-1-1 Call-Flow (US view)



“Plain Old Telephone Service” (POTS) line dials 9-1-1 (fixed ANI)



CO forwards to SR and includes ANI



SR determines proper PSAP and forwards call including ANI

911 Tandem Switch

Home

555-1234

Class 4

CO Switch

Class 5

CO Switch

PSAP

#002

PSAP

#003

PSAP

#001

(Selective Router)

ULg VoIP

47 Legacy Architecture

PhoneCompany, Inc.

The End Device

OSI Model

Layer 1/2

Mywires

Layer 3

Mynetwork

Locat

ion

PhoneCompany, Inc.

Layer 7

Mydialtone

(22)

ULg VoIP

48 Internet Architecture

Last Mile, Inc.

ISP, Inc.

Location/Presence.com

Common

Point—The End

Device

OSI Model

Locat

ion

Locat

ion

Layer 3

Network

Layer 7

Application

Layer 2

Access

I Think I’ll

Advertise My

Location

Dumb Network—Smart Endpoints

Problem: The Global Road Warrior

Internet

Chicago

PSAP

Hotel in Chicago

Corporate

HQ in Paris

VPN to Cor

_porate

This issue Must be solved!

112, What’s That?

Chicago,

Where’s That?

How Do I Route

This One?

(23)

ULg VoIP

50 SIP Routing Based on UAC’s Location

Alice

Outbound Proxy

--0a0

Content-Type: application/sdp

v=0

o=alice 2890844526 2890844526 IN IP4 atlanta.com

c=IN IP4 10.1.3.33

t=0 0

m=audio 49172 RTP/AVP 0

a=rtpmap:0 PCMU/8000

--0a0

**Content-Type: application/pidf+xml (short form*)**

<gml:location>

<gml:coordinates>28.44N 81.46W </gml:coordinates>

</gml:location>

<method>802.11</method>

<provided-by>www.cisco.com</provided-by/>

--0a0--

SIP Routing based on Location

urn:service:sos is not globally unique

INVITE

w/ SDP and Location

INVITE sips:urn:service:sos SIP/2.0

Via: SIP/2.0/TLS pc33.atlanta.com;branch=z9hG4bK74

Max-Forwards: 70

From: Alice <sip:[email protected]>;tag=9fxced76sl

To: <sip:urn:service:sos>

Call-ID: [email protected]

CSeq: 31862 INVITE

Geolocation: <cid:[email protected]>

Route: <sips:[email protected];lr>

Contact: <sip:[email protected]>

Content-Type: multipart/mixed; boundary=0a0

Content-Length: 311

Proxy MUST learn UAC’s location,

determine where UAC is, then

Route the call to the proper Public Safety

Answering Point (PSAP)

* “Short form” means not enough room here

If LoST query done by UA, may be as a

Route header

Though not sure yet

ULg VoIP

51 Agenda



‘XXth Century’ voice



‘XXIst Century’ voice

Packetization

Quality of service

Signalling

Issues with NAT

(24)

ULg VoIP

52 Network Address Translation: IP at Home



IPv4 addresses are scarce and close to exhaustion



Network Address Translation helps

WiFi ʻRouterʼ

Multiplex all inside

Hosts over the ISP address

ADSL Modem

Internet

192.168.1.1 ADSL or Cable modem:

1 IPv4 address

192.168.1.2 Different NAT Behaviors...



Mainly for stateless UDP sessions like RTP streams

Symmetric NAT:

one entry only for a specific 5-uple

<udp, global address, global port, remote address, remote port>

Full-Cone NAT:

one entry only a for a 3-uple

<udp, global address, global port>

Restricted-Cone NAT

: one entry only a for a 4-uple

<udp, global address, global port, remote address>

Port-Restricted-Cone NAT

: one entry only a for a 4-uple

<udp, global address, global port, remote port>

(25)

ULg VoIP

54 Symmetric NAT

ULg VoIP

55

(26)

ULg VoIP

56 What is STUN/ICE?



STUN

Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NAT)

STUN (RFC3489) is a request/response protocol

Response contains IP address and UDP port of request

Allows client behind a NAT to find out its public address, the type of NAT it is behind and the

internet side port associated by the NAT

Example application: Googletalk



ICE

Interactive Connectivity Establishment

Defines a standardized method for SIP-enabled clients to determine a set of IP addresses where

clients can establish contact behind firewall

Leverages STUN to collect IP addresses

Example: MSN Live Messenger

STUN Overview

Simple Traversal of UDP through NAT



RFC 3489



Client-server protocol



Allows a client behind a NAT

find out its public address

the internet side port associated by NAT with a particular local port

type of NAT it is behind



This information is used for UDP communication between two hosts that

are both behind NAT routers.



Free implementation of STUN client/server

(27)

ULg VoIP

58 STUN Operation



STUN server located on the public Internet.

Using 2 addresses and 2 ports.



STUN usages

– binding discovery,

– NAT keepalives



STUN messages are sent on the very same

ports that RTP will use latter

– First 2 bits allow to differentiate between STUN

and RTP

STUN

STUN Server

NAT2

NAT1

STUN Client

Public Internet

Private Net 2

Private Net 1

ULg VoIP

59 Interactive Connectivity Establishment (ICE)

Overview



offer-answer model for media streams through NAT.



use of STUN and its relay extension TURN

in a specific methodology which avoids many of the pitfalls of using any one

alone.

Each agent can have its own STUN server, or they can be the same



ICE agents (endpoints) discover their topologies to find a path or paths

by which they can communicate.



Agents L and R are capable of engaging in an offer/answer exchange

SDP messages to set up a media session between L and R. Exchange

will occur through a SIP server...

(28)

ULg VoIP

60 Gathering Candidate Addresses



each agent has a variety of candidate transport addresses:

directly attached network interface

A translated address on the public side of a NAT (a "server reflexive"

address)

The address of a media relay the agent is using

Could be IPv4 or IPv6 or both

Example

Stun Srvr

Binding discovery usage

192.0.2.2:3478

Agent L

10.0.1.1 Agent R

192.0.2.1 NAT

192.0.2.3

(29)

ULg VoIP

62 Connectivity Checks



Local

Order highest to lowest priority candidates

Sends them to R over the signaling channel

in the SDP offer.



When R receives the offer:

same gathering process

responds with its own ordered list of candidates.

sorts the candidate pairs in priority order.

Sends checks on each candidate pair in priority order.



Both acknowledge checks received from the other agent.

ULg VoIP

63 Agenda



‘XXth Century’ voice



‘XXIst Century’ voice

Packetization

Quality of service

Signalling

Issues with NAT

Security

(30)

ULg VoIP

64 Voice and Data Threat Models Merge



IP Telephony

inherits IP data network threat

models:

Reconnaissance, DoS, host vulnerability exploit, surveillance,

hijacking, identity, theft, misuse, etc.



QoS requirements of IP Telephony

increase exposure to

DoS attacks

that affect:

Delay, jitter, packet loss, bandwidth



PC endpoints typically require user authentication,

phones

typically allow any user

(exceptions: access/billing codes,

Class of Service)

IPT Servers



They are essential to IPT



Protected by

Strict security policy enforcement (firewall, …)

Host security: IPS, AV, …

Applying security fixes

RBAC management

(31)

ULg VoIP

66 Design a Secure IP Network

Data And Voice Segmentation



Physical separation is of course giving the best security but has investment

constraints



Use the same physical access, core, and distribution layers for the two segments

but segment logically



Segmentation also provides easier QoS configuration, scalability, and manageability