voice over ip 與 ip telephony 簡介資策會網路及通訊實驗室 conference over ip team...

Voice over IP Voice over IP 與與 IP Telephony IP Telephony 簡介簡介

資策會網路及通訊實驗室資策會網路及通訊實驗室Conference over IP TeamConference over IP Team

楊政遠博士楊政遠博士[email protected]

2003/07/26

mailto:[email protected]



Review - PSTNReview - PSTN• PSTN(Public Switch Telephone Network)

– Signaling: System Signal No: 7 (SS7)

– Carrier: T1 主幹 and successors …...

STP

Local loop

DTMF

Signaling plane

Bearer plane

客戶端 (CPE)

局端 (CO)

Review - Voice ConferenceReview - Voice Conference

• Basic issues of voice conference setup

1. ?

3. Digital voice packets

2. Conference setup

AD/DA compress/decompress AD/DA

compress/decompress

phonebookserver

4. Conference terminate

Review - basic issuesReview - basic issues

• Telephony Issues (PSTN v.s. VoIP)– Signaling

• Addressing / Control– PSTN - SS7 (ITU E.164)– VoIP - H.323 、 SIP 、 MGCP 、 Megaco/H.248

• Capability exchange– PSTN - Analog voice / -law 、 A-law PCM– VoIP - Digital voice / G.711 、 G.723.1 、 G.729

Review - basic issuesReview - basic issues

• Telephony Issues (PSTN v.s. VoIP)– Bearer

• Transport– PSTN - TDM (Time Division Modulation) Trunk – VoIP - RTP over UDP/IP

• Delay and Jitter– PSTN - circuit switching / propagation delay– VoIP - packet switching / unbounded delay and jitter

• Internetworking between the existent PSTN, GSM/GPRS and future 3G all IP network.

Review - short conclusionReview - short conclusion

• Signaling– Addressing: find call party– Call control: control the call progress – Capabilities exchange: negotiate the media

types of this call

• Media transport (bearer)– media processing– media transmission

Media TransportMedia TransportMedia ProcessingMedia ProcessingMedia Transmission

Process of digital voice transmissionProcess of digital voice transmission

Low-pass filterSampling &A/D convert

Silentdetection Compression

RTP packetencapsulation

RTP packetdecapsulation

DecompressionTiming reconstruct

D/A convert

Internet

VoIP Endpoint FunctionalityVoIP Endpoint Functionality

AD/DA converter

DSP coding

Buffering andpacketization

Jitter buffer

TCP/IPprotocol stack

Network interface

PCM

copper wire

IP

framesPhone

interface

Digital SignalProcessing

Packet handling

Digitalization SpeechDigitalization Speech

• Digitalization Speech– Low Pass Filter (LPF) 300 Hz ~ 3000 Hz– Sampling and Quantization

VoIP VoIP Endpoint FunctionalityEndpoint Functionality

AD/DA converter

DSP coding


Jitter buffer


Network interface

PCM

copper wire

IP

framesPhone

interface


Packet handling


• PCM (Pulse Code Modulation)– digital quantization introduces distortions


• main speech coding techniques – waveform codec, source codec and hybrid codec

VoIP VoIP Endpoint FunctionalityEndpoint Functionality

AD/DA converter

DSP coding


Jitter buffer


Network interface

PCM

copper wire

IP

framesPhone

interface


Packet handling

R,G,B bitmap

Y,Cb,Crmatrix

ColorTransform

frequencymatrix Quantizer

DiscreteCosine

Transform

Huffmanencoder

RTP packetencapsulation

Quantizationtable

Huffmantable

Media TransportMedia TransportMedia ProcessingMedia Transmission Media Transmission

Voice Quality of ServiceVoice Quality of Service

• Interactive Voice QoS factors– Packet lost– Delay– Jitter

Voice QoS - Packet LostVoice QoS - Packet Lost

0%5%

10%15%20%25%30%35%40%45%

Per

fect

Exc

elle

nt

Go

od

Acc

epta

ble

An

no

yin

g

Bad

Un

usa

ble

G.711

G.723.1InternetIntranet

• Minimize one-way delay, keep it below 150ms ITU G.114 states one-way delay <= 150 msec ~200 msec is acceptable

GPRS BackboneIP Network

IP based network

variable delay 20~300 or more ms

Fixed delay 1. Framing: 20~30 ms 2. Processing: 15 ms 3. Transmission: 10ms 4. Decompress/buffer: 25 ms

Variable delay 1. Buffer: 5~20 ms 2. Network: 20 ~ ? ms

Framing (algorithm): 20 ~ 30 msCompress (H/W DSP): 5 msProcessing (packetize): 10 ms

Receiving buffer: 20 msDecompress delay: 5 ms

Voice QoS - DelayVoice QoS - Delay

• Codec algorithm delay ( Ex. G.729 )– serialize the frame ( 10 ms)– look ahead (5 ms)

total algorithm delay = 15 ms

Frame

next sample Sampling & A/D converter

8000 Hz


0100200300400500600700800900

Per

fect

Exc

elle

nt

Go

od

Acc

epta

ble

An

noy

ing

Bad

mill

isec

ond

s

G.711Intranet

Internet


Voice QoS - JitterVoice QoS - Jitter

Jitter (Delay Variation)Jitter (Delay Variation)

0

100

200

300

400

500

600

Per

fect

Exc

elle

nt

Go

od

Acc

epta

ble

An

no

yin

g

Bad

Un

usa

ble

mill

isec

on

ds

G.711Intranet

Internet

Packet Handling LatencyPacket Handling Latency

• Jitter– variability in the arrival rate of data is called jitter

H i H o w are yo u

H i H o ...w a re yo u

J itte r

sender

rece ive

• Jitter bufferH i H o w are yo u

H i H o ...w a re yo u

J itte r

sender

rece ive

H i H o w are yo u< 1 5 0 ~2 0 0 m s

J itte r b u ffe r/ S m o o th e r

p layback

Voice QoS - JitterVoice QoS - Jitter

DefinitionsDefinitions

• Voice over IP (VoIP)– Voice over Internet Protocol

• voice packet over well controlled IP network !

– does not imply Voice over Internet

• IP Telephony– Telephony system based on Internet Protocol– Inter-operabilities

• standards• compatibility

Voice packets transmissionVoice packets transmission

• TCP(reliable) or UDP(unreliable) ?– The characteristics of interactive voice/video

• on-the fly (realtime)– retransmission is none-sense

• human physiology– tolerate few information lost independently

• isochronal– timing information snapshot and re-construct

– media frame encapsulated in RTP/UDP/IP

IP header(20 bytes)

UDP header(8 bytes)

RTP header(12 bytes)

media payload

RTP: A Transport Protocol for Real-Time RTP: A Transport Protocol for Real-Time Applications (RFC 1889) Applications (RFC 1889)

http://www.ietf.org/html.charters/avt-charter.htmlhttp://www.ietf.org/html.charters/avt-charter.html

• The simplest RTP fixed header

RTP (RFC1889)RTP (RFC1889)

IP header UDP header RTP header RTP payload

Fields of RTP HeaderFields of RTP Header

• V (version):– RFC 1889 RTP version 2, V=2

• P (padding): – padding bytes ?

• X (extension):– RTP header extension ?

• CC (count of contributor):– number of media contributors (for mixer)

• M (marker):– media specified

• audio: the begin of talk spurt• video: begin of end of video frame

• PT (payload type):– Defined by RFC 1990

Fields of RTP HeaderFields of RTP Header• Sequence number:

– increment by one– initial value is random

• Timestamp:– reflect the sampling instant of the 1st data bytes– format depends on application– initial value is random, increments monotonically

• Sync SRC:– synchronization source ID– random choice– RTP session global uniquely

RTP Header profile (RFC1900)RTP Header profile (RFC1900)

PT encoding nameaudio/video

(A/V)clock rate

(Hz)Channels(audio)

0 PCMU A 8000 11 1016 A 8000 12 G721 A 8000 13 GSM A 8000 15 DVI4 A 8000 16 DVI4 A 16000 17 LPC A 8000 18 PCMA A 8000 19 G722 A 8000 1

10 L16 A 44100 211 L16 A 44100 115 G728 A 8000 125 CelB V 9000026 JPEG V 9000028 nv V 9000031 H261 V 9000032 MPV V 90000

SignalingSignaling AddressingAddressing Call control Call control Capabilities Capabilities exchangeexchange

ReviewReview

• The milestone of Voice over IP– the 1st experiment of voice packet over IP

• 1974 Network Voice Protocol (RFC741)

– the 1st commercial Internet telephony AP, Windows 3.1• Vocaltec, 1995

– the 1st version of H.323• ITU, 1996

– the 1st widely deployed H.323 AP • Microsoft NetMeeting, May, 1996

– the 1st commerical Internet Telephony Service• Delta Three, 1996

VoIP signaling protocol standardVoIP signaling protocol standard

• ITU-T H.323– http://www.itu.int/rec/recommendation.asp?

type=folders&lang=e&parent=T-REC-H.323

• IETF MGCP – RFC2705

• IETF SIP– RFC3261– http://www.ietf.org/html.charters/sip-charter.html

• IETF/ITU-T Megaco/H.248– RFC3015

Session Initiation ProtocolSession Initiation Protocol

• SIP Architecture– RFC3261

SIP UserAgent

SIP UserAgent

SIP UserAgent

RegistrarProxyServer

RedirectServer

SIP Server

INVITE SIP:[email protected] SIP/2.0…….

180, Ringing

200, OK

ACKRTP (voice)

BYE

ACK

Caller CalleePickup & dial

ringing

pick up

on-hook

SIP BASIC Call flow

ringback

VoIP protocol standard - SIPVoIP protocol standard - SIP

Request MethodsRequest Methods

INVITE The user is begin invited to participate in a session.

ACK The client has received a final response to an INVITE.

OPTIONS The server is begin queried as to its capabilities.

BYE The user wishes to release the call.

CANCEL It cancels a pending request (not completed request).

REGISTER It conveys the user’s location information to a SIP server.

Response Status LineResponse Status Line

• SIP-Version SP Status-Code SP Reason-Phrase CRLF– Status-Code =

– SIP/2.0 SP 180 SP Ringing CRLF

1xx Informational

2xx Success

3xx Redirection

4xx Client-Error

5xx Server-Error

6xx Global-Failure

SIP Request ExampleSIP Request Example

INVITE sip:[email protected] SIP/2.0 Method type, request URL and SIP version

Call-ID:[email protected] Globally unique ID for this call

Content-type:application/sdp The body type, an SDP message

Cseq:1 INVITE Command Sequence number and type

From:sip:[email protected] User originating the request

To:sip:[email protected] User being invited into the call

Via:SIP/2.0/UDP 140.92.61.55:5060 IP Address and port of previous hop

Blank line separates header from body

v=0 SDP version

o=smayer 280932498 280932498 IN IP4 140.92.62.105

Owner/creator and session identifier

s=sip session The name of session

p=+886-2-25643588 Phone number of caller

c=IN IP4 140.92.61.105 Connection information

t=0 0 Time the session is active

m=audio 49170/1 RTP/AVP 1 media name and transport

SIP RegistrationSIP Registration

SIP Registrar(domain: iptel.org)

Location Server

jiri@

195.3

7.7

8.1

73

REGISTER sip:iptel.org SIP/2.0

From:sip:[email protected]

To:sip:[email protected]

Contact:<sip:195.37.78.173>

Expires:3600

SIP/2.0 200 OK

SIP Operation in Proxy ModeSIP Operation in Proxy Mode

SIP Proxy Server

Location Server

[email protected]

7.7

8.1

73

INVITE

sip:[email protected]



Call-ID:[email protected]

SIP/2.0 200 OK

jiri

?

SIP/2.0 200 OK

INVITE sip:[email protected]



Call-ID:[email protected]

ACK sip :[email protected]

[email protected]@195.37.78.1

73

SIP Operation in Redirect ModeSIP Operation in Redirect Mode

SIP RedirectServer

Location Server

Calle

e@

hom

e.co

m

Calle

e

?

INVITE

sip:[email protected]

[email protected]

[email protected]

302 Moved TemporarilyContact: [email protected]

ACK sip:[email protected] sip:[email protected]

SIP/2.0 200 OK

ACK sip:[email protected]

RTCPRTP

SIP, H.323 and MGCPSIP, H.323 and MGCP

IP

MGCP

Call Control and Signaling Signaling and Gateway Control

Media

H.225

Q.931

H.323

H.323 Version 1 and 2 supports H.245 over TCP, Q.931 over TCP and RAS over UDP.

H.323 Version 3 and 4 supports H.245 over UDP/TCP and Q.931 over UDP/TCP and RAS over UDP.

SIP supports TCP and UDP.

TCP

RAS

UDP

SIPH.245

Audio/Video

RTSP

Protocol wars - Protocol wars - Viewpoint from CISCOViewpoint from CISCO

Projected Port (DS0) Protocol Transition Rates

Q1CY99 Q1CY00 Q1CY01 Q1CY02 Q1CY03 Q1CY04

Calendar Quarters

% P

ort

Un

it S

ales

MGCP / H.248 - DS0s

SIP - DS0s

H.323 - DS0s

MixedH.323 & SIP

20%

40%

60%

80%

100%

Next Generation Converged Network Next Generation Converged Network andand

IP Telephony systemIP Telephony system

0

1

2

3

4

5

6

7

8

9

10

1997 1998 1999 2000 2001 2002 2003 Year

Rel

ativ

e tr

affi

c

Total

Data

Telephony

Siemens

• Telecommunication deregulation– Investment reward : Data network > voice network– Cost - single network architecture– Cost - open standards / short time-to-market

• Open VoIP and supplemental standards– H.323 、 MGCP 、 Megaco/H.248 、 SIP

• Bandwidth is no more a critical issue– DWDM ， xDSL / cable ， Fast/Giga Ethernet

• Quality of Service guarantee

Next Generation Converged NetworkNext Generation Converged Network


T ru n kg a te w a y

PSTN

M e d iag a te w a y

c o n tro llo r

PSTNInternet

T ru n kg a te w a y

S ig n a lin gg a te w a y

M G C PM E G A C O /H .2 4 8

STP

SCP

SSPSSP

S ig n a lin gg a te w a y

M e d iag a te w a y

c o n tro llo r

STP

SCP

PO TS

PO TS

M G C P /S IPp h o n e

S o ftsw itc h

softsw itchsoftsw itch

ana logyphone se t

R e s id e n tia lg a te w a y

S IP ¡B M G C P ¡BM E G A C O /H .2 4 8

M G C P ¡BM E G A C O /H .2 4 8

S IP -T

S IP -T

S IP -T


• Residential Gateway / Integrated Access Device

Call Control and Switching

Operation System Support

Feature and Application Creation

• IP Telephony System must support

IP Telephony SystemIP Telephony System

SIP based IP Telephony SystemSIP based IP Telephony System

CDR Server(s)

Feature Server(s)

Provisioning Server(s)

3rd Party Billing System

RADIUS

SNMP NetworkManager

ClearingHouse

Internet

SIP proxy ServerSIP proxy Server

PSTN

Gateway

SIP proxy Server

SIP IP Phone MGCP Device

MGCP/SIPTranslator

SIP proxy Server

H.323/SIP Translator

SIP proxy Server

H.323 Terminal

SIP based

VOCAL System [http://www.vovida.org/]

H.323 Translator: Acts as a Gatekeeper to control H.323 endpoints.Talks SIP to the rest of the network for routing and features.


MGCP Translator: Acts as a call agent to control MGCP end points. Talks SIP to the rest of the network for routing and features.


SIP based IP Telephony SystemSIP based IP Telephony SystemSIP proxy Server: Acts as a trusted boundary for calls entering or leaving a network. Provides authentication and collects billing information for the CDR server.

CDR Server: Collects billing information from Marshal Servers and interfaces with billing systems using the RADIUS accounting protocol.


Provisioning Server: Used to provision, configure and manage subscribers and servers from a GUI.


Feature Server: Provide CPL based or XML scripts that run basic telephony features.


VoIP Feature ServicesVoIP Feature Services• Feature services are the value-added functions of the

phone system– Core features

• Calling Information– Calling Number Delivery (CND) or Calling Line Identification (CLID) /

Calling Party Identity Blocking (CIDB)• Calling Forwarding

– Forward All Calls (CFA) / Forward - No Answer Mode (CFNA) / Forward - Busy Mode ( CFB )

• Call Blocking / Call Screening– Set features

• Call transfer / Call Return / Call waiting / Cancel Call Waiting ( CCW )– Scriptable features

• Call Processing Language (CPL)

IP Telephony - SoftswitchIP Telephony - Softswitch

Softswitch

Cellular Station

Media GatewaysIAD with DSL/Cable Modem

Digital Cross Connect

SS7 Gateway

SS7

Application Servers

Q.931/Q.2931

CPL

SIP

MGCP

MEGACO

SIPH.323

MGCP

33GPP Network ModelGPP Network Model

Endpoints with voice driving Endpoints with voice driving converged IP infrastructureconverged IP infrastructure

VideoTelephony

VoicePortals

PC toPhone

IP PhonesPDA

UnifiedMessaging Voice-enabled

Websites

InstantMessenger

Voice Service FocusVoice Service Focus

PSTN

IPSec orMPLS

SOHO

Internet

SS7

BranchOffice

HQ

Ent/SMB B

Messaging,ACD, IVR

HQBranch Office

Soft Switches

Enterprise A

Enterprise B

Ent/SMB AIOS Telephony

Services

CallManager

1. 1. Managed IP Managed IP TelephonyTelephony

3. 3. IP Centrex and IP Centrex and Hosted AppsHosted Apps

2. 2. Voice-Enabled Voice-Enabled Data VPNData VPN

4. 4. Integrated Integrated AccessAccess

V

V V

All IP NetworkAll IP Network

3G/4G Wireless Coverage

Home WLAN

Restaurant WLAN

Office LAN Hotel WLAN/LAN

Airport WLAN

LAN, WLAN hot spots LAN, WLAN hot spots

and 3G/4G wireless mobilityand 3G/4G wireless mobility

Wireless LAN Voice MobilityWireless LAN Voice Mobility

The Big Technical Challenge: The Big Technical Challenge: 802.11 VoIP Mobility802.11 VoIP Mobility

• Two Types of mobility:– Macro Mobility is the change of domain/administration

• Between “hotspots”

• Between Cellular (wide area) and WLAN (local area)

– Micro Mobility is the change of sub-net attachment (Campus, Enterprise)

Hotspot A(AP AP

Hotspot B

AP AP

Internet

Micro-Mobility Macro-Mobility Micro-Mobility

Call Control an Mobility ProtocolsCall Control an Mobility Protocols

• Two protocol approaches to support mobility– Support mobility at Network Layer: Mobile IP– Support Mobility at the Application Layer: SIP– H.323 is not expected to play a significant role in VoIP mobility

• SIP is widely supported in PC market and applications– Microsoft has included SIP as part of Windows XP release– Sip Handles Proxy server, NAT and Firewall issues– Ideal For HOME/SOHO/Consumer Market

• Mobil IP is desired but requires significant infrastructure investment

An Example: Loosely Coupled An Example: Loosely Coupled Cellular GPRS-WLAN IntegrationCellular GPRS-WLAN Integration

Operators IP

Network

Internet

HLR - AuC

CG

Dual ModeMN

SGSN GGSN

Billing Mediator

GPRS CORE

APBSS-1

WLAN Network

APBSS-2

APBSS-N

GPRS/UTRANNetwork

AP: WLAN Access PointBSS: Basic Service SetCG: Charging GatewayHLR: Home location registerAuC: Authorization centerSGSN: Serving GPRS support nodeGGSN: Gateway GPRS support nodeCAG: Cellular access gatewayFA: Foreign AgentHA: Home Agent

Billing System

FA/AAA

HA

CAG

SIP Roaming SupportSIP Roaming Support

• Logging into different IP networks away from home

• Basic Steps:1.Get an IP address

• Use DHCP

2.Register with local proxy• For firewall transversal for UDP

3.Register with home Registrar• For calls routing


Home.comVisit.com

From:[email protected]:166.1.2.3 From:[email protected]

Contact:[email protected]

INVITE

INVITEINVITE

• Remote registration

From:[email protected]:166.4.5.6

Move


• Precall mobilityHome.comDHCP

IP Address INVITE

INVITE

OK

302 moved temporarily

ACK

ACK

MEDIA


• Midcall mobility

INVITE

OK

ACK

MEDIA

MEDIA

中斷 ?

voice over ip 與 ip telephony 簡介 資策會 網路及通訊實驗室 conference over ip team...

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voice over ip 與 ip telephony 簡介資策會網路及通訊實驗室 conference over ip team...