cn2 chapter 13 mmcom qos iptelprotocols
TRANSCRIPT
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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 ...
Heterogeneous IP-Network
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
Heterogeneous IP-Network
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
Heterogeneous IP-Network
PSTN
Media
(ISDN, GSM, SS#7, ...)
Gateways for Interworking
Signalling Gateways
Signalling
Services
(QoS, Firewalls, ...)
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3. Signaling: H.323
Signaling Protocols Overview
H.323 SIP
Heterogeneous IP-Network
PSTN
Media
(ISDN, GSM, SS#7, ...)
Gateways for Interworking
Signalling Gateways
Signalling
Services
(QoS, Firewalls, ...)
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)
Example of the Phases (3)
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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
Example of the Phases (4)
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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
Example of the Phases (5)
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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
Example of the Phases (6)
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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
Heterogeneous IP-Network
PSTN
Media
(ISDN, GSM, SS#7, ...)
Gateways for Interworking
Signalling Gateways
Signalling
Services
(QoS, Firewalls, ...)
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
e e
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
e e
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
d e
5. H.323 + SIP
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t t c . d Signaling Protocols Overview
H.323 SIP
Heterogeneous IP-Network
PSTN
Media
(ISDN, GSM, SS#7, ...)
Gateways for Interworking
Signalling Gateways
Signalling
Services
(QoS, Firewalls, ...)
d e
d e
H.323 + SIP (2)
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t t c . d
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
t. d e
. d e
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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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
ARQ
REGISTERTo: [email protected]
at 130.83.139.175
t . d e
c . d
e
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:]
d t . d e
c . d
e
Media Gateway Control Protocol (MGCP) (2)
Approach: to centralize intelligence
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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
d t . d e
tc . d
e
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)
d t . d e
t c . d
e
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
d t . d e
tt c . d
e
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
d t . d e
t t c . d
e
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
d t . d e
t t c . d
e
7.1 Specic Problem - Firewall Interaction
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t
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
a d t . d e
t t c . d
e
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
a d t . d e
h t t c . d
e
IP-Telephony Firewall Interaction SIP (draft-rosenberg-sip-rewalls-00.txt)
problem domain
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t a
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
ta d t . d e
h t t c . d
e
7.2 Specic Problem: Privacy & Authentication Support
Deploying security services based on Public Key Infrastructures
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t a
w w
w . h
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
c r y p te d da ta s t rea m
t a d t . d e
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e
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
t a d t . d e
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e
Value Added Services (2)
Mobility
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t
w w
w .
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
t a d t . d e
h t t c . d
e
Value Added Services (3)
Workow integration
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w w
w .
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!
s t a d t . d e
. h t t c . d
e
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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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
s t a d t . d e
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e
TAPI (Telephony Application Programming Interface) (2)
Version 3.0l l API
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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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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
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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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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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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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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
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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
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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