CN2 Chapter 13 MmCom QoS IPTelProtocols

8/3/2019 CN2 Chapter 13 MmCom QoS IPTelProtocols

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eCommunication Networks II

Prof. Dr.-Ing. Ralf Steinmetz

TU Darmstadt - Darmstadt University of Technology,

Dept. of Electrical Engineering and Information Technology, Dept. of Computer Science KOM - Multimedia Communications Lab Merckstr. 25, D-64283 Darmstadt, Germany, [email protected] Tel.+49 6151 166151, Fax. +49 6151 166152

httc - Hessian Telemedia Technology Competence-Center e.V

Merckstr. 25, D-64283 Darmstadt, [email protected]

Multimedia Communications /QoSSpecic Topics:Internet and Intranet TelephonyProtocols (H.323, SIP, MGCP)


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Scope

KN III (Mobile Networking), Distributed Multimedia Systems ( MM I and MM II),Telecooperation II,III. ...; Embedded Systems

L5Applications

T e r m

i n a l

a c c e s s

F i l e

a c c e s s

E - m a i

l

W e b

P e e r -

t o -

P e e r

I n s t . -

M s g .

IP-Tel.

Application Layer

(Anwendung)

SIP &

H.323

L4 Transport Layer(Transport)Internet:

UDP, TCP, SCTP

N e t w . T

r a n s i

t i o n s

S e c u r i

t y

A d d r e s s

i n g

TransportQoS - RTP

L3 Network Layer(Vermittlung)Internet:

IPNetwork

QoS

L2 Data Link Layer(Sicherung)LAN, MAN

High-Speed LAN

L1 Physical Layer(Bitbertragung) Queueing Theory & Network Calculus

Introduction

Legend: KN I KN II


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Overview

1. Introduction

1.1 Introduction: Motivation and Expectations1.2 Introduction: Origin1.3 Internet+Intranet Telephony: What do we mean?

2. Scenarios, Basics & Principles, Building Blocks

2.1 Enhanced Scenarios: IP-Telephony Gateways2.2 Building Blocks

2.3 Driving Forces and Related Protocol Families2.4 IP Telephony - Interaction of Protocols


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Overview (2)

3. Signaling: H.323

3.1 H.323 - Architecture3.2 H.323 - Media Coding3.3 H.323 - Basic Elements3.4 H.323 - Functions3.5 Phases3.6 H.323 - A Scenario

4. Signaling: Session Initiation Protocol - SIP

4.1 Session Description Protocol - SDP4.2 SIP - Components4.3 SIP - Example - Control and Data Flow4.4 SIP - Requests/Methods, Responses, Header e.g.

5. H.323 + SIP

6. Media Gateway Control Protocol (MGCP)



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Overview (3)

7. Operation

7.1 Specic Problem - Firewall Interaction7.2 Specic Problem: Privacy & Authentication Support

8. Value Added Services8.1 Interfaces8.2 Communication Workow

9. Products and Prototypes

9.1 Software Only Systems9.2 Major joint IP-Tel-PABX Infrastructure Project at Darmstadt9.3 Open Source Developments9.4 Darmstadt Example: Interaction with Firewalls9.5 Darmstadt Example: MBone2Tel-Gateway - Concept9.6 Darmstadt Example: A Virtual PBX9.7 Darmstadt Example: QoS & Charging via RSVP



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Overview (4)

10. Future Trends: Today & Near future

11. Annex

11.1 References11.2 Glossary11.3 Standardization: Some relevant groups


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1. Introduction

Based on Internet Real-time and Multimedia Protocols

Signaling Quality of Service Media transp.

H.323 SIP RTSP RSVP RTCP

H.261,MPEG

RTP

TCP UDP

IPv4, IPv6

PPP AAL3/4 AAL5 PPP

Sonet ATM Ethernet Modem


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Introduction (2)

And in some more detail

Application Call Controll and Resource Management

MG-CP

SIP

A/VApplications

H.323 Terminal Controland Management

DataApplicatio

ns

ISDN UserPart

AINAPI

G.nnnH.261H.263

R T C P

H . 2

2 5

. 0

H . 2

2 5

. 0

H . 2

4 5

T .

1 2 4

T . 1

2 5

TCAP RTP

SCCP UDP UDP TCP

T . 1

2 3

MTP3 SS7 NetworkLayer IP IP Layer

MTP2 SS7 Link Layer Link Layer

MTP1 SS7 Phys Layer Physical Layer


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1.1 Introduction: Motivation and Expectations

Trend:

shift towards dominance of data trafc (telephony volume


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1.2 Introduction: Origin

IP-Telephony evolved over many years from

early voice over networks, e.g. Etherphone early experiments at Xerox with audio over Ethernet

Computer Telephony Integration CTI control of hardphones or Modem/ISDN-cards in a PC

and more recently from Video Conferencing

as a special case (1 to 1 communications) Mbone

multicasting audio data over the Internet Audio in many PCs

Softphones


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1.3 Internet+Intranet Telephony: What do we mean?

What we do NOT mean: just the simplied users point of view (out of date)

like user interfaces to phone on a PC, soundcards & softphones i.e. just to surf and to talk

VoIP: Voice over IP just voice packets over an IP protocol i.e. no other media (as voice allowed)

Internet telephony (as known in massmedia) only as a means for cheap phoning i.e. it is much more

What we do mean: Professional view

integration of telecommunications and data networks,

related services and applications simplied management and operation, PBX and data network enhancements or replacements Value Added Services


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IP-Telephony: What do we mean?

What we do NOT only mean: The well-known point of view

Characteristics

cost saving cheaper call atrate is needed (hardly any) hardware costs just a soundcard and a headset only one line/cabling is needed (surf&talk)

quality low quality is accepted if its for free

services participants lists my telephone book conguration (skins) for Softphones

Heterogeneous IP-Network

GW

CTI


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Internet+Intranet Telephony: What do we mean?

The professional view: Scenarios (apart from PC to PC) deployment of dial-in and dial-out facilities

IP-based backbone (as a part of future telephone system)

Characteristics cost saving quality services convergence

IP networkPSTN

Internet Telephony Gateway

Phone to PC

IP networkPSTN PSTN

Internet Telephony Gateway Provider A Internet Telephony Gateway Provider B

Phone to Phone


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Traditional Telephony vs. IP-Telephony

Circuit switched analog and digital networks internal sophisticated SS7 protocol

Well known basic call model: nite state machine:

off-hook, tone, ring, ..Endsystem easy and cheap Q.931 protocol to access switch

High reliability of individual switches 99.9999% (less than 5 min/year)

Centralized network intelligence centralized service control points SCP for services access through add-on to switches: service switching points SSP


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2. Scenarios, Basics & Principles, Building Blocks

Calling Party to contact partner using

direct IP-addressing, personal telephone book or directory service to use Internet telephony service provider gateway

in order to reach phone partners on POTS (plain old telephone system) to receive call processing information (ring, busy)

to perform full- or half-duplex conversation

Called Party possible registration with variety of directories application (standalone / applet) or

start-up mechanism listening on (proprietary / well-known) port ring - predetermines handling

audible / visible information,

accept, reject, store, forward, answering machine ...


GW GW

Caller Callee


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2.1 Enhanced Scenarios: IP-Telephony Gateways

Goals

to extend the range of reachable users to have means for

gathering experience & gaining widespread acceptance transition period

Scenarios (apart from PC to PC) deployment of dial-in and dial-out facilities

IP-based backbone (as a part of future telephone system)

IP networkPSTN

Internet Telephony Gateway

Phone to PC

IP networkPSTN PSTN

Internet Telephony Gateway Provider A Internet Telephony Gateway Provider B

Phone to Phone


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IP Telephony - Comprehensive View

H.323 SIP


PSTN

Media

(ISDN, GSM, SS#7, ...)

Gateways for Interworking

Signalling Gateways

Signalling

Services

(QoS, Firewalls, ...)


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2.2 Building Blocks

Basic Connectivity bandwidth technology (IP

over/vs. ATM) tetwork planning QoS support ...

Signaling(telephony related)

to end systems in the backbone

Operation migration

strategymanagement

maintenance security

infrastructure ...

Value AddedServices

new applications gateways surveillance unied

messaging

IP Telephony demands expectations limitations


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Building Blocks (2)

Basic Connectivity

interoperability of end systems adaptive encoding

Signaling

standardized infrastructure to locate and address communication partners nd out about their specics

locating (ingress / egress) gateways towards POTS efcient setup and use of multipoint communication mobility (identity attached to person, not to a line jack)

approaches: H.323 and H.450.x for supplementarys services

Session Initiation Protocol (SIP) Media Gateway Control Protocol (MGCP) / Megaco

Trend towards umbrella standard architecture


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Building Blocks (3)

Additional Functions for Operating and Maintenance

network and facility management pricing, billing, charging (in the Internet) security issues like

privacy for end users authentication and protection for infrastructure interaction with other rewalls

Value Added Services vision:

Ubiquitous Seamless Communication integration with other services

e.g. MBone, Unied Messaging Workow, conferencing, Computer Supported Cooperative Work

E-commerce surveillance entertainment distributed games


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2.3 Driving Forces and Related Protocol Families

Agreement on data transport

but difference on how to deal with control

Internet

Telecommunications

MGCP

IPTel, Pint

ITU-T SG 16ETSI- Tiphon

H.3xx

SS#7

SIP

IETF Working Group and Mailing Lists

DrivingForces

experience gatheredfrom e.g. multimediaconferencing

combination andextension of protocols

more lightweight andadaptable

H.323 specication extensive and complex

standards POTS & ISDN know-how industry acceptance existence of reference

implementations

ECTF, IMTC, ITC


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2.4 IP Telephony - Interaction of Protocols

Signaling Quality of Service Media transp.

H.323 SIP RTSP RSVP RTCP

H.261,MPEG

RTP

TCP UDP

IPv4, IPv6

PPP AAL3/4 AAL5 PPP

Sonet ATM Ethernet V.34


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PSTN/IP Network

Application Call Controll and Resource Management

MGCP

SIP

A/VApplications

H.323 Terminal Controland Management

DataApplicatio

ns

ISDN UserPart

AINAPI

G.nnnH.261H.263

R T C P

H . 2

2 5

. 0

H . 2

2 5

. 0

H . 2

4 5 T.

124T.

125TCAP RTP

SCCP UDP UDP TCP

T . 1

2 3

MTP3 SS7 NetworkLayer IP IP Layer

MTP2 SS7 Link Layer Link Layer

MTP1 SS7 Phys Layer Physical Layer


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Signaling Protocols

Candidates - Tasks - Basic Principles ITU H.323 Session Initiation Protocol SIP

S/MGCP and MeGaCo

H.323 SIP


PSTN

Media



Signalling Gateways

Signalling

Services



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3. Signaling: H.323

Signaling Protocols Overview

H.323 SIP


PSTN

Media



Signalling Gateways

Signalling

Services


l ( )


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Signaling: H.323 (2)

ITU-T Recommendation

industry driven packet-based multimedia conferencing over LAN no inherent quality-of-service guarantees

Properties

stateful binary session as ASN.1 notation: Packet encoding rules

Protocol suite H.225.0 / Q.931

registration, authentication, status (RAS) call control

H.245 logical channel capabilities negotiation

dynamic port negotiation ( problem with rewalls) H.450.x - H.323 Supplementary Services

supplementary services

H 323 P f H 320 h H 32 F il


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H.323 as Part of H.320, the H.32x Family

H.320 species (as overview) videophone for ISDN

H.310 adapt MPEG 2 for communication over B-ISDN (ATM)

H.321 dene videoconferencing terminal for B-ISDN (instead of N-ISDN)

H.322 adapts H.320 for guaranteed QoS LANs (like ISO-Ethernet)

H.323 original: videoconferencing over non-guaranteed LANs

H.324 terminal for low bit rate communication (over V.34 Modems)

3 1 H 323 A hit t


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3.1 H.323 - Architecture

Source: H.323 Standard Document

interoperable and industry standard may form building block for wide-spread deployment somehow complex and developed mainly for local environments

V.70Terminal H.324Terminal SpeechTerminal H.322Terminal SpeechTerminal H.320Terminal H.321Terminal H.321Terminal

H.323Terminal

H.323MCU

H.323Gatekeeper

H.323Gateway

H.323Terminal

H.323Terminal

GSTN GuaranteedQOSLAN

N-ISDN B-ISDN

H.323

H.310 terminaoperating inH.321 mode

H 323 Architect re (2)


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H.323 - Architecture (2)

Video I/O Equip.

Audio I/O Equip.

User Data Appl.

System Control

Video CodecH.261, H.263

Audio CodecG.711, G.722, G.723G.728, G.729

ReceivePathDelay

H.225.0Layer

LANInterface

RAS GK Interface

H.245 Control

Q.931 Call Setup

3 2 H 323 Media Coding


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3.2 H.323 - Media Coding

Specied Audio Coding

Specied Video Coding H.261: Video codec for audiovisual services at p x 64kbit/s

CCITT standard from 1990 for ISDN with p=1,..., 30

technical issues: real-time encoding/decoding max. signal delay of 150ms constant data rate implementation in hardware (main goal) and software

ITU-Rec. Description

G.711 Audio codec, 3.1 KHz at 48, 56, and 64 Kbps (normal telephony)

G.722 Audio Codec, 7 KHz at 48, 56, and 64 Kbps

G.728 Audio Codec, 3.1 KHz at 16 Kbps

G.723 Audio Codec, for 5.3 and 6.3 Kbps modes

G.729 Audio Codec, 8 kbps audio codec

H 323 Media Coding (2)


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H.323 - Media Coding (2)

Specied Video Coding

H.263 extension to H.261 max. bitrate: H.263 approx. 2.5 x H.261

H.261 and H.263 source image formats

Format Pixels H.261 H.263Encoder Decoder Encoder Decoder

SQCIF 128 x 96 optional required

QCIF 176 x 144 required required

CIF 352 x 144 optional optional4CIF 704 x 576

not dened optional16CIF 1408 x 1152

3 3 H 323 Basic Elements


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3.3 H.323 - Basic Elements

Terminal client endpoints provide real-time two-way communication

Gatekeepers (GK) brain of a H.323-zone but optional

zone = logical grouping of devices only one GK per zone no GK means, i.e. no zone exists

provides address translation assigns bandwidth to connections

H.323Terminal

H.323MCU

H.323Gatekeeper

H.323Gateway

H.323Terminal

H.323Terminal

H 323 - Basic Elements (2)


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H.323 - Basic Elements (2)

Multipoint Control Units (MCU) support for multipoint conferences resource management codec negotiation

Gateways (GW) translation functions between

transmission formats, AV codecs communication procedures

H.323Terminal

H.323MCU

H.323Gatekeeper

H.323Gateway

H.323Terminal

H.323Terminal

3 4 H 323 - Functions


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3.4 H.323 Functions

Call processing divided into

registration with the gatekeeper using REGISTRATION AUTHENTICATION AND S TATUS PROTOCOL RAS

call signaling using H.225.0 may be

gatekeeper routed - and then H.245 gatekeeper routed gatekeeper routed - and then H.245 directly routed direct i.e. H.245 directly routed

media capability exchange

using H.245 usually end-to-end

3.5 Phases


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3.5 Phases

1. Call set-up

e.g. point-to-point or multipoint request2. Initial comm. between endpoint & capability exchange

master/slave relation is dened one terminal is dened to be the master with the "highest" media capabilities resolves conicts of media capability exchange

3. Establishment of audio/visual communication between endpoints4. Media data ow5. Request and negotiation of Call Services

e.g. request for additional bandwidth may occur any time after having established audio/visual communications

6. Call termination

Example of the Phases


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Example of the Phases

1. H323-CallSetUp-1-init

registration confirmed

B - Called partyGatekeeperA - Calling party

request registrationRAS RCF

RAS RRQ

request admissionARQ

admission confirmed

ACF

registration confirmed

request registration

RAS RCF

RAS RRQ

optional: initiallyGatekeeper-Discovery usingMulticast

Signaling can be "relayed" viathe Gatekeeper (not mandatory)H.225.0 - Q.931 Setup (listening H.245 A)

H.225.0 - Q.931 Call Proceeding

H.225.0 - Q.931 Alerting

H.225.0 - Q.931 Connect (H.245 B)

GK keepstrack ofressources

ARQ

ACF

Example of the Phases (2)


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p ( )

2. H323-CallSetUp-2-capability

B - Called partyA - Calling party

H.245 Terminal Capability Set (Called Party)

Master / Slave Determination

H.245 Terminal Capability Set (Called Party)

H.245 Terminal Capability Set ACK

Gatekeeper

Master / Slave Determination

Master / Slave Determination ACK (B=Slave)

H.245 Terminal Capability Set ACK

Master / Slave Determination ACK (A=Master)



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p ( )

3. H323-CallSetUp-3-avcomm

B - Called partyA - Calling party

H.245 Open Logical ChannelData Type, Multiplex Parameters (RTP "listen")

H.245 Open Logical ChannelData Type, Multiplex Parameters (RTP "listen")

H.245 Open Logical Channel ACK

Gatekeeper

RTP ports again

H.245 Open Logical Channel ACK

RTP ports again



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p ( )

4. H323-CallSetUp-4-ow

Media Flows over RTP using the UDP portsopened by the Open Logical Channel Commands To UDP Port

To UDP Port

Calling party A Called party B



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5. H323-CallSetUp-5-renegotiate

Called party BCalling party A

BCF

BCR

Need more bandwidth

H.245 Close Logical Channel

H.245 Close Logical Channel ACK

whole multi-step H.245 Open Logical Channel proc.

Optional Request foradditional bandwidth,if required, BCR/BCF-BRJ can occur anytime an endpointwishes to adjust its callbandwidth

Gatekeeper



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6. H323-CallSetUp-6-terminate

Called party BCalling party A

Disengage from GK

DRQ

Video, Data and Audio Channels are closed

H.245 CloseLogicalChannel (1)

H.245 endSessionCommand (2)

H.245 CloseLogicalChannel (3)

H.245 CloseLogicalChannel ACK (4)H.245 CloseLogicalChannel ACK (5)

H.225.0 ReleaseComplete (6)

May be sent before the CloseLogicalChannel ACK is received

DCF

causes

Gatekeeper

Disengage from GK

DRQ

DCF

GK gets ressourcesback

3.6 H.323 - A Scenario


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i.e. has been in focus of the industry with its extensions for supplementary services

Internet

PSTN

Firewall

Gateway

GATEKEEPERMCU Terminal BTerminal

Terminal A

H.323 Zone

Intranet

H.323 Supplementary Services - H.450.x


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H.450 Protocols

Description

H.450.1 Generic functional protocol for the support of supplementaryservices in H.323

H.450.2 Call transfer supplementary service for H.323

H.450.3 Call diversion supplementary service for H.323H.450.4 Call hold supplementary service for H.323

H.450.5 Call park and call pickup supplementary services for H.323

H.450.6 Call waiting supplementary service for H.323

H.450.7 Message waiting indication supplementary service for H.323

H.450.8 Name identication supplementary service for H.323

H.450.9 Call Completion Supplementary Service

H.450.10 Call Offer Supplementary ServiceH.450.11 Call Intrusion Supplementary Service

4. Signaling: Session Initiation Protocol - SIP


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Signaling Protocols Overview

H.323 SIP


PSTN

Media



Signalling Gateways

Signalling

Services


Signaling: Session Initiation Protocol - SIP (2)


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Properties

IETF driven peer-to-peer signaling protocol RFC 2543 text-based signaling protocol

machine independence header / body comparable to HTTP

Based (most often) on UDP, but TCP allowed as well simplicity, speed

Session

location server

Site A

User with

1. Invite

7. OK

symbolic name8. ACK

2 . W h e r e ?

3 . @

. . . .

Site B

4. Invite

6. OK

5. Ring

calls another

9. ACK

Proxy Mode


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4.1 Session Description Protocol - SDP


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intention: to describe multimedia sessions

session announcement, session invitationproperties: abbreviated message format as ASCII-text

SDP in SIP messages includes description of: media to use (codec, sampling rate)

media destination, IP address and port number session name and purpose time / period the session is active contact information

example (SIP)v=0o=mgoertz 28944526 2890842807 IN IP4 130.139.83.68s=ICW SIPi=A session for voice transmission using SIPu=http://www.kom.e-technik.tu-darmstadt.de/[email protected] (Manuel Goertz)c=IN IP4 130.233.154.68t=2873397496 2873404696m=audio 49170 RTP/AVP 0

m=video 51372 RTP/AVP 31

4.2 SIP - Components


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UAC: User-agent client

UAS: User-agent serverredirect server:to redirect requests

proxy server: to fork requests to multiple servers

registrar: to track user locations

Usally: user agent = UAC + UAS

Often: registrar and proxy or redirect server combined

Registrar / Location Server

Proxy Server Redirect Server

Media GatewayIP Network

User Agents

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4.3 SIP - Example - Control and Data Flow


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INVITE: sip:[email protected] SIP/2.0To: sip:[email protected]

From: sip:[email protected]: [email protected]: 1 INVITEContact: sip:[email protected]=IN IP4 192.168.1.1m=audio 4567 RTP/AVP 0

Proxy

SIP/2.0 100 Trying

SIP/2.0 180 Ringing

SIP/2.0 200 OK

INVITE sip:[email protected]: sip:[email protected]

ACK sip:sip:[email protected]

ACK: sip:[email protected]: , tag=11

SIP/2.0 302 Moved temporarilyContact sip:[email protected]: , tag=11

INVITE sip:[email protected]: sip:[email protected]

SIP/2.0 180 RingingSIP/2.0 200 OKFrom: sip:[email protected]: , tag=21Call-Id: [email protected]: 1 INVITEContact: sip:[email protected]

SIP/2.0 OK

BYE sip:[email protected] SIP/2.0

RTP Audio

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4.4 SIP - Requests/Methods, Responses, Header e.g.


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Requests/Methods

REGISTER registers a user address at a registrar binds a permanent address to the current location registrations can timeout => refresh

INVITE usually begin of a session body contains session description re-Invite to change session state

OPTIONS

returns user capability to the requestor ACK

only used in session initiation process end of 3-way handshake

CANCEL terminates a uncomplete request

BYE terminates an open session

e e

SIP - Responses


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t t c . d Characteristics

Similar to HTTP - syntax xxx explanatory text

Codes response codes from HTTP/1.1 extended with new codes

Numbering

1xx : Informational 100 Trying 180 Ringing

2xx : Success 200 OK

3xx : Redirect 302 Moved temporarily

4xx : Client error 400 Bad Request

5xx : Server error 6xx : Global failure

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SIP - Header


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t t c . d E.g. Invite Request

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP gw.proxy.comFrom: Ralf To: ManuCall-ID: [email protected]: 1 INVITE

Subject: IP Telephony Tutorial at ACM MM 2001

Content-Type: application/sdpContent-Length: 400

v=0o= gw 881300437 IN IP4 192.168.0.2c=IN IP4 gw.proxy.dem=audio 3456 RTP/AVP 0 3 4 5

blank

Start Line

SIP Header

Request Header

Entity Header

SDP Data

Header

Body...

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SDP - Example within SIP Header


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t t c . d Structure of a SDP message, e.g.

v=0o=rac 28944526 2890842807 IN IP4 130.139.83.68 s=testi=A session for voice transmission using SIPu=http://www.kom.e-technik.tu-darmstadt.de/[email protected] (Ralf Ackermann)c=IN IP4 130.233.154.68t=2873397496 2873404696m=audio 49170 RTP/AVP 0m=video 51372 RTP/AVP 31

d e

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5. H.323 + SIP


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t t c . d Signaling Protocols Overview

H.323 SIP


PSTN

Media



Signalling Gateways

Signalling

Services


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H.323 + SIP (2)


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Mapping and Gatewaying: H.323/SIP Gateway

SIP

H.323

H.323 SIPFirewall

CTI

RTP

H.323 Gatekeeper

* Registration* Admission Control* Suppl. Services

H.323 User AgentSignaling

Media Streaming

SignalingMedia Streaming

SIP User Agent

H.323 Signaling

H.323SIP

RTPMedia

SIP Signaling

H.323

H.323

d e

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H.323 + SIP: Cases (2)

ll


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t t c . Case: H.323 Participant calls SIP participant

using Gatekeeper telephone number or alias (e.g. 100) mapping to SIP URL is done

at the gateway based on a loadable mapping / registration list

Case: SIP Participant calls H.323 participant has to address the Gateway at port 5060

physically by means of IP address or symbolic name

uses SIP URL:

sip:@:5060 mapping to H323 participant by using the heading part of the SIP URL

other mappings could be integrated

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H.323 + SIP: Some Experiences (3)

B k d


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t t c . Background

based on own implementation

Limitations, i.e. to be considered .. capability set limited to G.711 64k law, due to

limited mapping of H.323 capabilities to SIP AVP proles and limited capabilities of example SIP client

SIP INVITE for H.323 originated calls based on empty (concerning description of endpoint IP address and RTP ports) initial

INVITE

transmission of this information as part of the ACK after adequate information has been received

from the H.323 side (as described in Gateway Draft) does not actively register with a SIP server SIP and H.323 messaging characteristics

based on internal (OpenH323, Vovida)

t . d e

. d e

Gateways between Signaling Protocols - SIP/H323

W k i g


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s t a d t

w w w . h

t t c Work in progress

SIP integration SIP - H.323 interaction

i.e building SIP-terminals integrating

SIP with conventional PABXs

bridging towards H.323

integrating comprehensive

signaling (QoS, security)into the process

RCFOK

SIP UA 130.83.139.175

GWGK2 GK1 H.323

Terminal

[email protected]

ARQ

[email protected]

REGISTERTo: [email protected]

[email protected]

at 130.83.139.175

t . d e

c . d

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6. Media Gateway Control Protocol (MGCP)

Goal


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t t c Goal

to bridge current circuit-based PSTN with IP-based protocols

Characteristics stateless functions in form of sequences of commands header and session descriptor session descriptor

set of parameters set up endpoints to produce or recognize media formats uses SDP (like SIP)

example: RQNT packet formatIP Header

UDP Header

MGCP Header

Packet Type = CommandCommand = Notification Request [RQNT]Transaction Identifier = 68Endpoint = aaln/[email protected] = MGCP 0.1Parameter = Request Events [R:]

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Media Gateway Control Protocol (MGCP) (2)

Approach: to centralize intelligence


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t a d

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t t c Approach: to centralize intelligence

in a call agent not in the Truncing (TGW) or Residential (RGW) Gateways

Source: MeGaCo Working Group

Features

(simple) UDP based protocol (evolved from SGCP) for managing endpoints connections (which are described using SDP) between endpoint new services may be implemented by augmenting the Call Agent

CallAgent

H.323/SIP

TruncingGateway

ResidentialGateway

IP Network

MGCP /UDPMGCP/ UDPSS7 GW

PSTN

Audio Dataon RTP

TCAP/SS7ISUP/SS7

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Media Gateway Control Protocol (MGCP) (3)

Megaco/H 248


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t t c Megaco/H.248

joint effort by IETF and ITU-T media gateway control protocol for use in distributed switching environments set up media (for example, voice trafc) paths through the distributed

network.

Properties signaling logic is located on Media Gateway Controllers (MGCs) media logic is located on Media Gateways (MGs)

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7. Operation

Security Issues


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t t Security Issues

Condentiality ensures privacy (partial) encryption of

control and mediastreams

Integrity

ensures correctness ofcontrol information

e.g. alerting, call routing,bill especially necessary for

signaling, control trafc

cryptographicprotection of message

diggest

Authentication

identity of com. partners

e.g. both, subscribers &service providers prevents spoong

(e.g. PIN, TAN)

cryptographicprotocols,Public KeyInfrastructures (PKI)

Access Control who is allowed to access

what part(s) of a systemsystem specicmechanisms,(media) rewalls

Non Reputationof Origin

basic mechanisms forprovider/customerrelationships

see above

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Security Issues (2)

Customer requirements


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t t Customer requirements accurate and veriable accounting condentiality and privacy protection of anonymity

Threats through one network for

(in-band) signaling voice data management of network components

signaling and service provided by decentral systems

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Security Issues (3)

Service provider requirements


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t p q mainly protection against corruption and fraud

Necessity for: IP-telephony specic security security for services

Standardization Effort condentiality

H.3xx / H.235 RTP data encryption (DES, Triple DES, RC2) key exchange in signaling messages (H.245)

IETF / SIP RTP data encryption (DES, Triple DES, RC2) key exchange is part of the SIP message bodies

authentication H.3xx / H.235

TLS or IPSec SIP

authorization header eld - digital signature

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7.1 Specic Problem - Firewall Interaction


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Firewall Functions permit/deny dedicated data ows

assignment of dedicated data ows to users/systems hiding internal structures monitoring, logging and alerting

Firewall components and architectures components: lters, proxies, stateful lters, ... architectures: DMZ, inbound lters, dual homed gateway, ...

Well dened functions, but many possible architectures

intern Fire wall extern

FW-function

owFilter Proxy Stateful Filter

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Firewall Interaction

Facts


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(all) organisations use rewalls IP-telephony applications/protocols differ from standard applications rewalls have problems to support IP-telephony applications/protocols IP-telephony is emerging to a carrier grade service

Firewalls have to support IP-telephony to allow the deployment

Failure prevents deployment

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IP-Telephony Firewall Interaction SIP (draft-rosenberg-sip-rewalls-00.txt)

problem domain


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w w

w . h problem domain

getting SIP and RTP through rewalls IP address for the media streams in SIP bodies

solution suggestions application layer rewall that understands SIP packetlter with SIP proxy

H.323 (draft-shore-h323-rewalls-00.txt) problem domain

dynamic port negotiation very complex structure due to ASN.1 and PER

solution suggestions realm specic IP (RSIP) packetlter with H.23. parser

vendor specic solutions/proposals complete solutions are missing

IETF foglamps put intelligence where the knowledge about the used protocols is:

in the endsystems

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7.2 Specic Problem: Privacy & Authentication Support

Deploying security services based on Public Key Infrastructures


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from Trust by Wire to Trust by Authentication

Local

Remote

Originator TerminalDestination

SignalingServer

CerticationCertication

S i g n Re q ue s t w i t h P r i v

a te Ke y

Exchange Certicates

Validate Public Key Validate Public Key

Get Certicate

SignalingServer

of Originator

Validate Public Key

e xc ha nge s es s ion ke ys

X.509 CAsCerti-cates

CA CA

CA

Networksystem

Gateway

CAsCerti-

cates

End-system

End-system

CAsCerti-cates

X.509

Networksystem

UserCerti-cates

X.509

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8. Value Added Services

Basic Connectivity > ..... > Value added services


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Major difference to POTS

telephony service e.g., H.450 like features click to dial & Web callback

new services, e.g. unied messaging Web assistance voice access to content application sharing electronic whiteboard

interoperating terminalsIP Telephony in and between INTRANETS

Interoperable and interacting gateways

IP Telephony enhanced / based PBXes

IP Telephony as common service in the INTERNETNEW SERVICES and related applications?

time

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Value Added Services (2)

Mobility


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wireless (moving while communicating) nomadic (go off-line and move, then go on-line anywhere) be mobile

Personalization user proles context awareness (i.e. situation denes/provides for facilities) group communication & multiple media functions support for various stati

reachable

temporary busy permanent absent

Protection of investment independence of

endsystems, terminal equipment destination medium (fax, sms, eMail) location physical number

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Value Added Services (3)

Workow integration


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well-known in working processes integration of communication methods in workows gateways to third party products (SAP R/3)

Service-Gateways coexistence

transition to convergence bridge between PSTN and IP-networks integration into workow

It is more important to reach to the partnerthan to know where he/she is!

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8.1 Interfaces

TAPI (Telephony Application Programming Interface)TAPI d d d j i l b Mi f d I l


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w w

w .

TAPI standard was created jointly by Microsoft and Intel. in a TAPI environment, the physical connection is made at the desktop

level only for WinX platforms packaged in a dynamic link library (DLL)

mainly for CTI hardware independence

Telephony Service Provider Interface interface for hardware drivers (= service provider) access for TAPI applications to the hardware

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TAPI (Telephony Application Programming Interface) (2)

Version 3.0l l API


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w w w user-level API

enhanced CTI methods H.323 based QoS support COM model

built-in Internet Protocol Telephony (IPT) components for creating new-world call center applications


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JTAPI (Java Telephony API) object orientated interface only the description not an implementation


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platform independent located between application and proprietary API

applets Web callback predictive calling

Java Applet Java Program

Java Implementation

Java Runtime

MS TAPI TSAPI Dialogic other

Telephone hardware

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JTAPI (Java Telephony API) (2)

Architecture modular


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w w w modular

extendable

JTAPI 1.3 class packages

Modul Packet

Core javax.telephony

Call Control javax.telephony.callcontrol

Call Center javax.telephony.callcenterMedia javax.telephony.media

Mobile javax.telephony.mobile

Phone javax.telephony.phone

Private Data javax.telephony.privatedata

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JAIN (Java API for Integrated Networks) integration of Internet and intelligent networks


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new abstraction layer removal of proprietary roadblocks set of software component library

JAIN Application LayerCallModel

Coordination &Transaction Model

Packet BasedWireless

PSTN

H.323/SIP

JAIN Protocol Layer

IN/AINMGCP

m s

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CSTA Computer Supported TelecommunicationApplications CSTA is a CTI interface


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provides access to telecommunication functions can be used with APIs, such as TAPI, JTAPI, JavaPhone, CallPath, TSAPI may be used by 3rd party applications

Task Group, ECMA TC32-TG11 development and renement a standardized Computer-Telephony Integration (CTI) interface interactions between computer applications and the telecommunications

network


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Communication Workow:Example of a Communication Process

Communication Workow:


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9. Products and Prototypes


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Broad range diverse functionality

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Communication Workow: Experiences

IP-Tel means to work in research


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development deployment

And available / research Software Only SystemsAnd e.g., in Darmstadt actual

Major joint IP-Tel-PABX Infrastructure Project at Darmstadt

Open Source Developments completed projects

Darmstadt Example: Interaction with Firewalls Darmstadt Example: MBone2Tel-Gateway - Concept Darmstadt Example: A Virtual PBX Darmstadt Example: QoS & Charging via RSVP

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9.1 Software Only Systems

E.g. Skypehttp://www.skype.com


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w like Instant Messaging KaZaA, P2P (peer-to-peer) technology

to connect you to other users to talk and chat

very good audio quality

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9.2 Major joint IP-Tel-PABX Infrastructure Projectat Darmstadt

Goals


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u - d a w to make extensive use of IP Telephony (eld test on a broader basis)

to operate & to research on PABX(es) and attached IP Telephony to install IP-phones with the following expected amount

starting with few islands of 30 to 100 phones via 10% IP-phones (600) towards 33% (2000)

Customers (about 6000 phones) University TUD and attached institutions (FHD, ...) research institutes (Fraunhofer-IPSI , Fraunhofer-SIT)

Constraints (some) 1-2 years timeframe of replacement operational and R&D (continuous enhancements) larger scale eld trial

Status replacement of the PBX for the scientic institutions on the way 6000 telephones to be replaced and a planned

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Test-Scenario

classical PBX classical PBX4S0 4S0


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IP-PhoneIP-Soft-Phone

GatekeeperProxy-Server

MCP-Server

IP-PhoneIP-Soft-Phone

GatekeeperProxy-Server

MCP-Server

PSTN-Phone

PSTN-Phone

Firewall Router

Router

PSTN

IP-Network

IP-Network

IP-Network

to Internet

GW

GW

ATM 155 Mbit


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9.4 Darmstadt Example: Interaction with Firewalls

Open Source - KOM

Contact: Utz Roedig ([email protected])


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www.kom.e-technik.tu-darmstadt.de/KOMproxyd/

set of modules for the integration of external proxies in a distributed rewall system

based most often on standard components for H.323

tested and in operation

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9.5 Darmstadt Example: MBone2Tel-Gateway - Concept


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Basic Idea: deploy gateways as point of interaction

intentionally use specics / advantages of the connected areas deploy Value-Added Services

Basic Functionality:

users may passively or actively take part in MBone audio conferences usingtheir conventional telephone

MBone MBone2TelGateway

PSTN

variety of users,using conventional, ISDN

or mobile phones

Internet

could have access facility via PSTN too -ISP

but usually with limited connectivityMBone Multimedia

Terminals

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Example: MBone2Tel-Gateway - Components

A di I f A di I fd MB


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Architecture

modular based on well-evaluated / evolving components using a simple graphical user-interface building blocks for Value Added Services and Gateway Farm

configurationdata

WWW-Interface

Audio Interface

Text-to-Speech

Audio Interfacetowards MBone

IP network(Internet)

multicast data

unicast data

Component

controlcomponent

to PSTN

DTMF

generatedaudio

Session Dir.Interface

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9.6 Darmstadt Example: A Virtual PBX

Gateway 1

G t 2

Prototype-Implementation


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Basic idea: have working scenario and testbed / incorporate standard protocols ! enhance existing PBX

locally distributedMBone Session(s)

as Single Connections oras Virtual MCU

Central Control Components:- User Location- Gateway Location and Selection- User-specific Configuration(s)

enhanced withControl Component(s)

- Value Added Services

Point of Presence (PoP)of a User:(MBone)-ConferencingApplication

Control informationPersistentDatabase

Gateway 2

Gateway n

MediaData


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10. Future Trends: Today & Near future

Approaches device and technology driven


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u - d technology driven

Network engineering reservation and (not vs.) overprovisioning

Comprehensive umbrella developments (and standards) more than simple IP-Telephony more than simple basic call processing

Voice quality is not perfect

Delays are still too long signaling audio transfer

Security new security issues arise with IP Telephony

rewalls should communicate with other IP-Tel components

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11. Annex

The annex shall help toe plore in more detail the e plained topics


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o m . t

u - explore in more detail the explained topics

nd out about related topics

Hence only references to relevant information is provided

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11.1 References

URLs, please look at General

http://www itu ch


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u http://www.itu.ch

http://www.ietf.orghttp://www.computertelephony.org

H.323http://www.h323.org

SIPhttp://www.sipforum.org/

Books, please look at

IP Telephony - The Integration of Robust VoIP Services,Bill Douskalis,

IP Telephony: Packet-Based Multimedia Communications Systems,Olivier Hersent, et al.

Converged Networks and Services: Internetworking IP and the PSTN,

Igor Faynberg, et al. IP Telephony with H.323, M. Korpi

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References (2)

Papers, please look athttp://www.kom.e-technik.tu-darmstadt.de/Research/Publications/publications html


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u Publications/publications.html

http://www.cs.columbia.edu/~hgs/sip/papers.html

Conferences, please look athttp://www.pulver.com

http://www.upperside.fr/basip.htm

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11.2 Glossary

please have a look athttp://www.kom.e-technik.tu-darmstadt.de/Teaching/de/literatur/glossare html


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u de/literatur/glossare.html

http://www.ucc.ie/info/net/acronyms/index.html

http://www.gateway.de/knowledge/lexikon/

RFC 1983: Internet Users Glossary (IETF)

http://www.uni-koeln.de/allgemeines/glossar/

http://www.springer.de/iuk/glossar.htm

http://whatis.com

http://www.oreilly.com/reference/dictionary

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11.3 Standardization: Some relevant groups

ETF Working Groups and Mailing Lists IP Telephony (iptel) - Call Processing Service Model Gateway Attribute


97/98


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u IP Telephony (iptel) Call Processing, Service Model, Gateway AttributeDistribution Protocol

PSTN and Internet Internetworking (pint) - Architecture, Services andManagement (CTI vs. IN, Click-to-Dial, Click-to-Fax, Access to voicecontent, ...)

MMusic - Companion Protocols - SAP, SDP, SIP, RTSP, SCCP, (RTP/RTCP) Mailing List SGCP (Simple Gateway Control Protocol)

but, Transition to WG MEdia GAteway COntrol (megaco)

ITU-T SG 16 standardization of

multimedia communications between terminals, network equipment andservices in LAN and WAN environments

without explicit QoS support - H.323 series of standards

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Standardization: Some relevant groups (2)

ETSI TIPHON - Telecommunications and Internet Protocol Harmonizationover Networks


98/98


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3 phases: IP->GSTN, GSTN ->IP, GSTN->GSTN over IP working groups on:

requirements, security, charging

architecture and Reference Conguration call control procedures, information ow, protocols numbering, addressing and naming QoS interoperability testing

International Multimedia Conferencing Consortium (IMTC) -Voice over IP Forum (VoIP)

Internet Telephony Interoperability Consortium (ITC) - architecture, resource allocation, business strategies

CN2 Chapter 13 MmCom QoS IPTelProtocols

Documents

Transcript of CN2 Chapter 13 MmCom QoS IPTelProtocols