Transcript
VoIP 2002.7.24 한국외국어대학교 정보통신공학과 김희동 031-330-4254 016-9712-0000
[email protected]
2002-7-24
Krnet 2002
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발표내용
개요 PSTN의 이해(신호망, 지능망) VoIP관련기술(음성신호처리) VoIP 표준화 개요 Voice over IP (VoIP) signaling의 개념 VoIP 와 NGN VoIP 시스템의 구현기술 VoDSL All IP network (IMT-2000)
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전화교환망의 개관
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현재의 회선 교환망
SS7 EO Toll Toll
EO Tan EO
Tan
EO
Toll Toll
EO
Tandem/Toll Network
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교환기의 명칭
Local Switch
Transit Switch
EO (End Office) : 단국 CO (Central Office), Class 5 Switch
Tandem Switch : 중계교환기 Trunk Switch Access Tandem Class 4 Switch
기타 특수교환기
SSP (Service Switching Point) MSC (Mobile Switching Center) : 이동망의 교환기
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PSTN(전화망)의 구조 080 SCP AIN SCP LNP SCP
PBX
PSTN STP Transit Switch
Local Switch
VMS IP
DLC - Digital Loop Carrier IP - Intelligent Peripheral PBX - Private Branch Exchange
2002-7-24
Local Switch
DLC
SCP - Service Control Point STP - Signal Transfer Point VMS - Voice Mail System
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신호방식 국간 신호방식
가입자 신호방식
접속요구 발신음
다이얼
점유신호 번호송출허용 호출음
선택신호 응답신호 통
송수화기복구
호출신호 접속점유
화
종화신호
해제요구
절단신호
복구완료
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가입자 신호방식 Station Signaling
PSTN
Analog 전화기
이용자 선택번호 • Pulse Dial • DTMF
ISDN
교환기
Dedicated Network
• Busy tone • Dial tone • Ringback tone
Station Signaling
ISDN Terminals
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BRI(I.430) LAPD(I.441) Networking(Q.931)
Digital Telephone
2002-7-24
Call Progress Tone
Voice, Voice+Data Proprietary Signaling 8
국간신호방식 Trunk Signaling
Signaling
PSTN
Channel Associated Signaling : R2-MFC ISDN Signaling : PRI Common Channel Signaling : No.7
ISDN
교환기
Dedicated Network
Trunk Signaling
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공중 ISDN과 Signaling System 7 신호망 BRI
PBX1
PRI
BRI
Switch
DSS1
전송망 Transmission Network Signaling System 7
Switch
PRI
PBX2
DSS1
신호망의 도입으로
교환기와 교환기 사이의 데이터 경로가 형성됨. 신호망에 컴퓨터를 접속하여, 교환기와 컴퓨터가 상호 연동가능함. 지능망의 개념이 탄생함.
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SS7 - Protocol View
GSM MAP INAP IS-41
ISUP
TUP
TCAP SCCP MTP Level 3
MTP Level 2 MTP Level 1
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GSM MAP – Global System for Mobile Communications, Mobile Application Part IS-41 – GSM MAP like for NA wireless systems (AMPS and TDMA) INAP – Intelligent Network Application Part ISUP – Integrated Services Digital Network User Part TUP – Telephony User Part TCAP – Transaction Capabilities Application Part SCCP – Signaling Control Connection Part MTP – Message Transfer Part (Level 1 through 3) 11
SS7 신호망의 요소 SCP
Network 1
SCP
SSP
Network 2
STP
STP
STP
STP
SCP SCP
SSP SSP
SSP
SSP: Signal Switching Point Voice Trunk Signaling Link STP: Signal Transfer Point SCP: Service Control Point 2002-7-24
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SS7 ISUP
ISUP, used to establish, monitor and release circuits for calls in the PSTN Designed to allow transfer of circuit information and status between disparate ISDN networks Provides basic and supplementary services Allows automated network testing
Provides circuit testing (circuit validation and continuity) and translation testing (routing instructions to translate dialed digits into a routable address, e.g. a point code) Provides interactive and non interactive test calls
Traffic measurements and report generation
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SS7 ISUP Originating Switch
STP IAM
ACM
Terminating Switch
STP IAM ACM ANM
IAM
ACM ANM
ANM REL
RLC
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REL RLC
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REL RLC
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지능망으로의 구현기술 진화모델
Provider Control
800
교환기 기본기능 만 내장한 교환기
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교환기
교환기
AIN
NGN
Vendor 교환기 Control 교환기
다기능을 SCP 의 차세대 내장한 지원을 지능망구조로 교환기 받는 교환기 다기능을 외부로 부터 지원받는 교환기 Krnet 2002
API
교환과 제어가 완전분리된 교환기 15
지능망 서비스의 흐름 4) 5) DB 검색
SCP
SDP SS7 network 3) SCP 에 문의 2) 호전달
1) 지능망 호발신 단국
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6) SCP 에 응답
7) 호발신 SSP
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단국
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지능망 서비스의 흐름 4) 5) DB 검색
SCP
SDP SS7 network
05025005425? 1) 0502-5005425
2) 호전달
단국
2002-7-24
0313304254 7) 호발신 SSP
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031-330-4254 단국
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VoIP기술개요
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VoIP의 개요
용어
개념
VoIP ( Voice Over IP) VoP ( Voice over Packet Network) IP Telephony
패킷교환망을 통하여 음성서비스를 제공하는 기술 및 시스템
등장배경
Internet의 활성화와 함께, 패킷망이 통신망의 주류를 형성 패킷망에 음성신호를 전달하는 인터넷폰으로 통신비용절감 PC-to-Phone, Phone-to-Phone의 개념으로 확장 음성서비스를 패킷망을 통하여 전송하도록 확장 NGN(Next Generation Network)의 개념도입
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PSTN versus VoIP Today’s PSTN
VoIP
TDM 회선교환
Packet switching
QoS guarantees
Yes
No
Network resource reserved at call setup
Yes
No
Network elements
Class 4, Class 5 교환기
Gateways, gateway controllers, routers
Call processing intelligence
대부분 교환기에 위치
In separate gateway controllers
64 kb/s
Variable 8 – 32 kb/s
Signaling
DTMF, SS7
SIP, H.323
Transport
TDM in access, edge, core
ATM, FR, IP in access; ATM native IP in core
Underlying Technology
Bandwidth per call
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VoIP Issue
QoS
패킷망에 의한 음성전달기술
Carrier grade VoIP 서비스 게이트웨이의 Decomposition 신뢰성이 높은 시스템의 구현
다양한 서비스의 신속한 제공
음성압축, QoS, Delay, Jitter Signaling, Enhanced Service
대용량 게이트웨이의 구현
송신측에서의 음성압축 네트워크 측면의 기술 수신측에서의 기술
계층적 구조 개방형 구조
Next Generation Network으로 진화
Packet기술기반, 다양한 서비스의 수용, 고속접속지원
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Voice over Internet Protocol IP Transport Protocol
Uncompressed
Uncompressed
C. Speech
C. Speech
RTP C. Speech
RTP C. Speech
UDP RTP C. Speech
UDP RTP C. Speech
IP UDP RTP C. Speech
IP UDP RTP C. Speech
Medium: Ethernet, ATM, GSM, GPRS, ...
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Voice/RTP/UDP/IP
IP: Internet Protocol
UDP: User Datagram Protocol
no retransmission, no flow control <-> TCP UDP header : 8 octets (port identifier, …)
RTP: Real Time Protocol
IP header : 20 octets (IP address origination, IP address destination, …)
RTP header : 12 octets
Payload : encoded voice or multi-media stream
IP(20)
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UDP(8) RTP(12)
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C. Speech
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VoIP Protocol Stack Application
Application
Codecs
H.323, SIP
Codecs
Transport Signaling
Signaling Transport
UDP, TCP, ...
UDP, TCP, ...
IP
IP
(Wireless) Link
(Wireless) Link
Physical 2002-7-24
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Next Generation Tandem Network
SS7 EO EO
Packet
EO
EO
EO
Gateway Switch
Packet-Based Tandem Network
전화망이 모두 없어지거나, 통신망이 모두 IP phone으로 대치되기 전에는 게이트웨이에 의한 연동이 필수적임. 2002-7-24
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VoIP 기술의 요소
음성을 디지털 패킷화 및 전송
신호방식
Bearer Call Signaling Intelligent Network Call Processing
서비스 생성 및 관리
Speech Coding : 음성압축, 묵음검출 Packetization : IP 패킷의 크기, 헤더의 오버헤드 Multiplexing, Trunking
Supplementary and Multimedia Service Authentication, Mobility and Directory Service
교환 및 제어 구조
MSF SoftSwitch
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Packet Voice 요구사항
Services
Basic Services • • • •
PC-to-Phone Phone-to-PC Phone-to-Phone PC-to-PC
Supplementary Services • UMS ( Unified Messaging Service) • Internet Call Waiting • Click-to-Dial 등
Interoperability
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Packet Voice 요구사항 (계속)
Digital Signal and Packet Processing Voice Coding : 음질, 전송속도, 복잡도에 의해 결정 Trans-coding : 압축방식의 변환에도 음질 저하 최소화 Silence Removal : 묵음을 검출하여 묵음시 데이터전송 않음. Echo canceling : 전송지연에 따른 엑코현상발생 Packetization (Serialization) : 패킷화 지연시간 고려 Buffer smoothing : 음성지연
Addressing/Naming & Routing E.164, IPv4/IPv6, DNS and Directory Service IETF ENUM workgroup
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Packet Voice 요구사항
Call Processing
Signaling Mode의 수용 • CAS : Channel Associated Signaling • CCS : Common Channel Signaling
Call Response Time Call Processing Capacity
• BHCA : Busy Hour Call Attempt
QoS(Quality of Service) End-to-End Delay/Jitter Packet Loss에 대한 대처방안
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Packet Voice 요구사항 (계속)
Management
Interworking
Security Accounting/Charging/Billing
Bearer Interworking Service (IN) Interworking (080-xxxx, 700-xxxx, etc)
Mobility
Terminal and Service mobility
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패킷망에 의한 음성전달기술 음성압축, QoS, Delay, Jitter
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Voice Quality Definition 음질 평가 영향 요소: • 명료성 Clarity - The
Clarity
“Speech Quality Space” Delay
Echo
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amount of information that can be extracted out of a conversation • 반사 Echo - Reflection of the originating signal at the far end with enough strength and delay that it is perceptible to a human. • 지연 Delay - the time a signal needs to traverse the network
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User Perspective of Voice Quality
MOS (mean opinion score) quality:
Echo degradation
control : subjective perception is dependent on delay faithful reproduction of background noise
Levels :
depends on coder bit rate and complexity
gain control for optimum level MOS affected by low level
Latency
the effect of pure delay (no echo) on conversation dynamics can be detected well below 400 ms for highly interactive tasks.
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음성의 디지털 표현
x(t)
Sampler
Continuous-time Continuous-amp.
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x(n) = x(nt) Discrete-time Continuous-amp.
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Quantizer
x(n)
Discrete-time Discrete-amp.
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음성부호화 방식의 종류
Parametric (Vocoding)
Waveform Coding
PCM DPCM ADPCM DM ADM CSVD
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Hybrid Coding
APC RELP MPLPC CELP SELP
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Channel Formant LPC MBE
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Speech Coder의 형태
Waveform coding
Vocoder
Time domain Frequency domain 예 : PCM (Pulse Code Modulation) 64kbps
LPC ( Linear Prediction Coding) Low Rate 음성발생기관을 모델링
Hybrid coding
RELP ( Residual Excited Linear Prediction) CELP ( Code Excited Linear Prediction) 이동전화, 인터넷폰 응용
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음성발성기관 모델
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음성발생과 Vocoder 원리 10-30 ms 사이에는 Quasy-stationary
폐의 여기신호
입, 성도모양
음성파형
Sub optimal Solution 성모모양필터가 구해진 상태에서 여기신호를 추출함. 미지수 1개, 식 1개
음성파형으로부터 성도의 모양 필터 를 추출함. 미지수 1개, 식 1개
음성파형으로부터 성도의 모양 필터 폐의 여기신호를 추출함. 미지수 2개, 식 1개
2가지 값은 저전송속도로 수신단에 전송하고 수신단에서 합성함.
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분석-합성 방식에 의한 음성부호화
파형부호화 방식으로는 32-64 bps 정도로압축의 한계가 있음.
분석합성 방식이란
폐에서의 공기발생 + 성도필터 = 음성
프레임 구간(10-30ms)으로 구분 우선, 필터계수를 구하고 Excitation 신호를 구하여, 저전송속도로 수신단에 송신함.
수신단에서는
공기발생형태 : Excitation signal 성도필터 : Linear Prediction Filter
송신단에서 입력음성을 분석하기 위해서
음성발생모델을 이용하여 음성을 고능률 압축하는 방식
파라메터 정보를 가지고 음성을 합성함.
전송속도는 2.4 kbps 정도이나 음질이 문제임.
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Hybrid Coder
음질을 개선하기 위해서는
Excitation Signal 방식
성도필터의 계산방법은 그대로 채용하되, 유성음과 무성음을 구분하는 단순한 방법을 사용하는 대신, Excitation signal을 변화시켜가면서, 신호를 합성하여, 입력음성과 차이가 최소화 되면, Excitation signal의 값과 필터계수를 송신 Codebook을 미리 작성하고, index를 찾는 방식 신호를 산술적인 계산에 의해서 찾는 방식
Various excitation signal
MPE (multi-pulse excited) RPE (regular-pulse excited) CELP (code-excited linear prediction)
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Hybrid coders
Multi-Pulse Excitation
Regular-Pulse Excitation (RPE)
Efficient at medium bit rates. A sequence of nonuniformly spaced pulses as an excitation signal Amplitudes and positions are excitation parameters Efficient at medium bit rates. A sequence of uniformly spaced pulses as an excitation signal The position of first pulse within a vector and amplitudes are excitation parameters
Code-Excited Linear Prediction (CELP)
Efficient at low bit rates (below 8 kbps) A code book of excitation sequences Two key issues; the design and search of a codebook
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여기 신호의 종류 g3
g1
gk n4
n2 n1
a) multipulse
n3
nk
g2 0
5 g1
b) regular-pulse
g4 10
g2
15
20 g6
g4
K g5
g3 Codebook
c) Code-excited Linear Prediction
Codevector # 1 Codevector # 2 Codevector # 3 2M = N (M = 전송 bit)
Codevector # N 2002-7-24
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분석/합성 LPC coder Speech vector1
CELP
Vocal Tract Model Filter
~
S(n)
+
S(n)
e(n)
MP-MLQ
Error Minimization
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Frame Processing and Delay
G.729의 linear prediction Linear Prediction Frame Delay
LP parameter
Lookahead delay
time
120 samples (past)
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80 samples current frame
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40 samples (future)
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분석-합성 방식에 의한 음성부호화 Input speech Frame (10,20,30 ms) codebook Excitation Generation
Synthesis Filter Error Minimization
codebook
정보전송 Excitation Generation
Error Weighting
(a) Encoder Synthesis Filter
Reconstructed speech
(b) Decoder 2002-7-24
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H.323용 보코더
ITU-T G.7xx series 보코더
보코더 G.711 G.722 G.726 G.728 G.729 G.723.1
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비트율 (kbit/s) 64 64/56/48 32/24/16 16 8 5.3/6.3
지연시간 (ms) 0 3 0.25 1.25 25 62.5
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복잡도 (MIPS) <<1 10 1.25 30 20 20
대역폭 (kHz) 4 7 4 4 4 4
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ITU-T G.723.1
Dual rate (5.3/6.3 kbit/s) ITU-T 표준 보코더 1995년 말 표준확정, 멀티미디어 통신용으로 개발 대부분의 인터넷폰에서 표준으로 지원 Frame size : 30 ms, Lookahead : 7.5 ms 5.3 kbit/s higher rate
ACELP (Algebraic Code Excited Linear Prediction) USH (Univ. of Sherbrooke) contributed
6.3 kbit/s lower rate
MP-MLQ (Multi-Pulse maximum Likelihood Quantization) ACL/DSPG contributed Toll quality
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G.723.1 Extensions
Annex A
G.723.1 + VAD/SID/CNG(Comfort Noise Generation) 음성이 없는 구간에서 noise발생 Noise의 특성을 전송 - SID (Silence insertion description) • Active noise와 silent noise간의 자연스러운 연결
DTX (Discontinuous Transmission) • Noise특성을 전송할 필요가 있는지 결정
Annex B
G.723.1의 부동소수점 구현
Annex C
Scalable channel coding scheme for wireless applications H.324의 일부로 개발 Channel bit-rate: 0.7 ~ 14.3 kbit/s Channel codec: punctured convolutional codes
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Quality, Complexity and Delay DELAY/COMPLEXITY excellent
Quality
ITU4 2000+?
G.729 1996
G.728 1992
G.726 1984
good
G.711 1972
clean speech
G.723.1 1995
fair
poor 2
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16 8 bitrate (kb/s) Krnet 2002
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64
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Voice over Internet Protocol IP Transport Protocol
Uncompressed
Uncompressed
C. Speech
C. Speech
RTP C. Speech
RTP C. Speech
UDP RTP C. Speech
UDP RTP C. Speech
IP UDP RTP C. Speech
IP UDP RTP C. Speech
Medium: Ethernet, ATM, GSM, GPRS, ...
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Voice Coding Bandwidth Bytes Frame size
G.723.1
G.729a
30ms
10ms
Coder
20(G.723.1) 10(G.729a)
6.3
8.0
RTP
12
3.2
9.6
UDP
8
2.1
6.4
IP
20
5.3
16.0
Total
40
16.9
40.0
IPv6 Total
54
20.7
51.2
G.729a는 20ms 마다 패킷화하여 전송하면, 오버헤드 줄일 수 있음. Overhead를 줄이기 위한 방안 모색필요 IETF ROHC( Robust Overhead Compression) 2002-7-24
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Packet Loss Characteristics
Packet Loss 원인
Source Site - overwrite Intermediate Site - dropping Destination Site - time constraint
Packet Loss 대응방안
Packet Repetition • GSM 에서 사용하는 방식
Packet Interpolation Frame Interleaving • Bursty Error의 영향을 분산시키나 지연시간이 길어짐.
Forward Error Correction
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Delay Variation—―Jitter‖ A
B
C
Sender Transmits
t
A D1
B D2 = D1
C
Sink Receives
D3 = D2
t
Queueing delay 의 변화에 의해 발생 Hold time과 Jitter Buffer size의 비례관계 Jitter가 전체 Delay에 미치는 영향이 더 큼. 50-100ms 사이를 사용 Delay에 따라 Buffer Size가 적응되도록 함.
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DTMF Transfer
DTMF 문제
음성부호화방식은 음성의 특성에 최적화하여 압축비를 높임. Tones: fax tones, dialing tone, busy tone DTMF digits for two-stage dialing or voice-mail
DTMF전송방식
In-band 방식 (G.729에서 사용하는 방식) • DTMF tone을 음성처럼 부호화하여 전송함. • 음성부호화 방식이 tone도 정확히 코딩, 디코딩 할 수 있어야 함.
Out-of-band 방식 (G.723.1 의 방식) • • • • • •
2002-7-24
DTMF는 음성특성과 달라 DTMF의 전달특성이 나쁨 Encoder에서 DTMF 검출 제어패킷으로 만들어 별도로 TCP를 통하여 전송 수신부(Decoder)에서 DTMF톤 발생 검출전에 DTMF톤의 일부가 전송됨 데이타망의 상태에 따라 지연이 발생됨. Krnet 2002
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Out of Band 방식
The ingress gateway needs to intercept
The tones and DTMT digits Use an external signaling system Easy at the start of a call Difficult in the middle of a call
Encode the tones differently form the speech
Send them along the same media path An RTP packet provides the name of the tone and the duration Or, a dynamic RTP profile; an RTP packet containing the frequency, volume and the duration RFC 2198 • An RTP payload format for redundant audio data • Sending both types of RTP payload
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Echo Echo - Reflection of the originating signal at the far end with enough strength and delay that it is perceptible to a human.
Phone
PSTN Network Hybrid
Echo Cancel. Echo Cancel
Gateway
IP Network
H.323 Terminal
Line Echo Acoustic Echo
Echo tail ( 8 - 120 mS)
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Echo의 원인 및 대책 Central Office
2 Wire Local Loop Rx & Tx Superimposed
2w-4w Hybrid
EC
Receive Direction
인터넷 Transmit Direction
Echo는 반사에 의해서 생김. 2w-4w hybrid 에서의 Impedance mismatch가 엑코의 발생원인임. Echo는 항상 발생하지만, Echo의 지연시간과 Echo의 크기에 따라 문제로 대두됨.
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VoIP 표준
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VON 관련 표준화기관
ITU - International Telecommunications Union SG 16
IETF - International Engineering Task Force
multimedia 단말과 security에 관한 연구 H.323 web services 와 PSTN/ISDN의 상호연동 Iptel : IP telephony PINT: PSTN-Internet service Sigtran : Telephony signaling transport SIP : Session Initiation Protocol
ETSI - European Telecomm. Standards Institute
TIPHON (Telephony and IP Harmonization Over Networks) SPS5 (UNI/NNI signaling aspects)
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표준화기관의 관계 SG16 SG2 Multimedia Numbering
SG11 Signaling
SG13 Architecture
권고안
표준개발요구
Integrated Keeper spec Global country code
T1
ITU-T
Naming/Addressing QoS Interoperability
TIPHON Collaboration RFCs
Megaco
2002-7-24
PINT
IPTEL
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Requirement
SIP
IETF
60
VoIP Protocols
H.323 (ITU-T)
SIP (IETF)
initial push for VON interoperability Multimedia session control - call control Session Initiated Protocol – Call Control SIP/ SAP/ SDP SIP+ (MGC to MGC communication)
MGCP (IETF : Media Gateway Control Protocol)
Decomposed gateway model • media gateway controller (MGC) • media gateway (MG) • signaling gateway (SG)
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VoIP Protocols
MEGACO (IETF)
H.248 (ITU-T) [formerly H.GCP]
ITU variant of MEGACO
SIP-T (IETF, Cable Labs)
MEdia GAteway Control; Successor to MGCP
Signaling between media gateway controllers
SCTP (IETF Sigtran)
Simple Control Transmission Protocol Transport PSTN signaling (SS7, Q.931) over IP
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H.323
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H.323
ITU-T Recommendation H.323 Version 4
Describes terminals and other entities that provide multimedia communications services over Packet Based Networks (PBN) which may not provide a guaranteed Quality of Service. H.323 entities may provide real-time audio, video and/or data communications.
H.323 elements
Terminals Gateways Gatekeepers MCUs
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H.323 H.323 Terminal H.323 Gatekeeper
N-ISDN
Speech Terminal
H.320 Terminal
Scope of H.323 Non-Guaranteed QoS LAN H.323 Gateway
H.323 Terminal
Guaranteed QoS LAN
B-ISDN
H.322 Terminal
H.321 Terminal
H.321 Terminal
H.310 terminal operating in H.321 mode 2002-7-24
H.323 MCU
Krnet 2002
H.323 Terminal
V.70 Terminal (DSVD)
GSTN
H.324 Terminal
Speech Terminal (H.324M)
65
H.323 Functional Block Diagram Scope of H.323 Video I/O Equipment
Video Codec H.261, H.263
Audio I/O Equipment
Audio Codec G.711, G.722 G.723, G.728 G.729
User Data Applications T.120 etc.
Receive Path Delay
System Control
H.225.0 Layer
H.245 Control System Control User Interface
Local Area Network Interface
Call Control H.225.0 RAS Control H.225.0
•RAS - registration, admission, status 2002-7-24
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H.323 표준의 범위
H.323 defines
H.225 defines
System-level characteristics of terminals, gateways, gatekeepers, and MCUs Call signaling procedures (phases A-E) Refers to H.225 and H.245 specs Packetization and use of RTP/RTCP Use of Q.931/2 and RAS messages
H.245 defines
Procedures for use of Control Protocol messages
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H.323 is an “Umbrella” Specification Media
Signaling
Audio Codec
G.711 G.723 G.729
Call Control and
Video Codec H.261
RTCP
H.263
H.225
H.225
Q.931
RAS
TCP
UDP
H.245
RTP UDP
TCP
IP
2002-7-24
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Overview of H.323 Singaling
Audio codecs (G.711, G.723.1, G.728, etc.) Video codecs (H.261, H.263) Media streams transported on RTP/RTCP
RTP carries actual media RTCP carries status and control information
RTP/RTCP carried unreliably on UDP
Signaling is transported reliably over TCP
RAS - registration, admission, status (over UDP) Q.931 - call setup and termination H.245 - capabilities exchange
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통신과정
H.323의 통신과정 5단계
Call setup. Initial communication and capabilities exchange. Audio/video communication establishment. Call services. Call termination.
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단순한 H.323 Call Setup Gatekeeper GK 1. ARQ 2. ACF 3. SETUP 4. Call Proceeding 5. ARQ 6. ACF 7.Alerting 8.Connect
H.245 Messages RTP Media Path RAS messages Call Signaling Messages 2002-7-24
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Simplified H.323 Call Setup
Both endpoints have previously registered with the gatekeeper. Terminal A initiate the call to the gatekeeper. (RAS messages are exchanged). The gatekeeper provides information for Terminal A to contact Terminal B. Terminal A sends a SETUP message to Terminal B. Terminal B responds with a Call Proceeding message and also contacts the gatekeeper for permission. Terminal B sends a Alerting and Connect message. Terminal B and A exchange H.245 messages to determine master slave, terminal capabilities, and open logical channels. The two terminals establish RTP media paths.
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H.323 basic protocols Annex G
Gatekeeper RAS
Gatekeeper
Q.931/H.245
Q.931/ H.245
Q.931/ H.245
RAS
Signalling (Q.931) Endpoint
H.245 RTP/RTCP
Endpoint
Gatekeeper Routed Signaling Direct Routed Signaling 2002-7-24
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H.323 Components and Signaling H.225/RAS messages over RAS channel
H.225/Q.931 (optional)
H.225/RAS messages over RAS channel
Gatekeeper
H.225/Q.931 (optional) H.245 messages (optional)
H.245 messages (optional)
H.225/Q.931 messages over call signaling channel
Terminal
H.245 messages over call control channel
PSTN Gateway
H.245 – A protocol for capabilities advertisement, media channel establishment and conference control. H.225 - Call Control. RAS – Registration, admission and status protocol 2002-7-24
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H.225 RAS Signaling
RAS(Registration, Admissions, Status)시그널링
GRQ(Gatekeeper Request) RRQ(Registration Request) LRQ(Location Request) ARQ(Admission Request) IRQ(Inforamation Request) BRQ(Bandwidth Request)
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RAS Messages
GRQ/GCF/GRJ (Discovery)
Endpoint alias/IP address binding Endpoint authentication Destination Address Resolution Initial bandwidth request Call Routing Call Authorization
Get rid of call state
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Ping during active calls Resource information for gateways
BRQ/BCF/BRJ (Bandwidth)
Stateless name - IP address resolution Inter gatekeeper communication
IRQ/IRR (Information Request)
DRQ/DCF/DRJ (Disconnect)
LRQ/LCF/LRJ (Location)
ARQ/ACF/ARJ (Admission)
Unicast - Multicast Find a gatekeeper
RRQ/RCF/RRJ (Registration)
Ask for more/less bandwidth during call
URQ/UCF/URJ (Unregistration)
Get rid of registration state
76
RAS - Direct Call Signaling Gatekeeper A
LRQ
Gatekeeper B
LCF ACF
ACF
IP Network RRQ / RCF
RRQ / RCF ARQ
H.225 (Q.931) Setup
ARQ
H.225 (Q.931) Alert & Connect H.245
V
RTP
Gateway A
V Gateway B
Phone B
Phone A
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RAS - GK Routed Call Signaling Gatekeeper A
LRQ
Gatekeeper B
LCF H.225 (Q.931) Setup H.225 Alert & Connect H.245
ACF A&C
Setup
ACF
IP Network
H.245
H.245
ARQ Setup
V
ARQ
RTP
Gateway A
Phone A
2002-7-24
Alert & Connect
V Gateway B
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Phone B
78
H.225
호제어 시그널링(Call Control Signaling)
H.225표준, Q931 signal Message Setup Call Proceeding Alerting Release Complete Faclity 호제어 처리방식 종류 – DECS, GKRCS
2002-7-24
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Q.931 Signaling 절차 UNI
UNI
ATM 망
SET UP SET UP CALL PROCESSING CALL PROCESSING CONNECT CONNECT
CONNECT ACK CONNECT ACK
Signaling Cell(ATM Forum) VPI=0, VCI=5 2002-7-24
VCC 설정
ATM Cell Header
Krnet 2002
Signaling Message
80
H.245
H.245 미디어 제어 및 전송(Media Control and Transport) H.245 제어 동작과정
capabilities 교환 – 단말기의 음성, 비디오 및 데이터를 위한 송수신 능력 Master-Slave 결정 Round –Trip : 송신단말기에서 수신단말기까지의 지연을 결정. 로지컬 채널 시그널링
2002-7-24
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H.245 System Control
Capabilities Exchange
Master/Slave Determination Open Logical Channel/Ack
Codec negotiation - Transmitter/Receiver capabilities Mode Request - Request specific transmission format
Create new audio/video/T.120 stream
Commands and Indications
out of band DTMF
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82
H.323 End-to-End
Direct Signaling with Slow Start
1. H.225 RAS ARQ/ACF location/admission (UDP)
GK
3. H.225 RAS ARQ/ACF admission (UDP)
2. H.225 Q.931 signaling-setup, proceed, alert, connect (TCP) 323
4. H.245 media negotiation-TCS, MSD, OLC, OLCA (TCP)
323
5. RTP media (UDP)
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83
H.323 End-to-End
Gatekeeper-Routed Signaling with Fast Start 2. H.225 Q.931 signaling with fastStart media negotiation-setup, proceed, alert, connect (TCP)
GK
1. H.225 RAS ARQ/ACF location/admission (UDP) 323
2002-7-24
3. H.225 Q.931 signaling with fastStart media negotiation-setup, proceed, alert, connect (TCP) 4. H.225 RAS ARQ/ACF admission (UDP)
5. RTP media (UDP)
Krnet 2002
323
84
RTP/RTCP—RFCs 1889/1890
End-to-end network transport function
Sequence numbering Time Stamping Delivery monitoring Payload type identification - Voice, Video, Compression Type
RTCP (Real-Time Control Protocol) provides feedback on the quality of the distribution
V 4bytes E R
CC
M
Payload Type
Sequence Number
4bytes
RTP Timestamp
4bytes
Synchronization Source (SSRC) ID
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85
Session Initiation Protocol
2002-7-24
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Relevant IETF Working Groups
Audio/Video Transport (avt) - RTP Differentiated Services (diffserv) – QoS in backbone IP Telephony (iptel) – CPL, GW location, TRIP Integrated Services (intserv) – end-to-end QoS Media Gateway Control (megaco) – IP telephony gateways Multiparty Multimedia Session Control (mmusic) – SIP, SDP PSTN and Internet Internetworking (pint) – mixt services Resource Reservation Setup Protocol (rsvp) Service in the PSTN/IN Requesting InTernet Service (spirits) Session Initiation Protocol (sip) – signaling for call setup Signaling Transport (sigtran) – PSTN signaling over IP Telephone Number Mapping (enum) – surprises ! Instant Messaging and Presence Protocol (impp)
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What is SIP?
IETF RFC 2543 Session Initiation Protocol
Session Initiation Protocol - An application layer signaling protocol that defines initiation, modification and termination of interactive, multimedia communication sessions between users.
Components
User Agents Proxy Server Location Server Redirect Server Registrar Server
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SIP Distributed Architecture SIP Components
Location Server
Redirect Server
Registrar Server
PSTN User Agent
Gateway Proxy Server
2002-7-24
Proxy Server Krnet 2002
89
SIP Components
User Agents
An application that initiates, receives and terminates calls. • User Agent Clients (UAC) – An entity that initiates a call. • User Agent Server (UAS) – An entity that receives a call. • Both UAC and UAS can terminate a call.
Proxy Server
An intermediary program that acts as both a server and a client to make requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. Interprets, rewrites or translates a request message before forwarding it.
Location Server
A location server is used by a SIP redirect or proxy server to obtain information about a called party’s possible location(s).
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SIP Components
Redirect Server
A server that accepts a SIP request, maps the address into zero or more new addresses and returns these addresses to the client. Unlike a proxy server, the redirect server does not initiate its own SIP request. Unlike a user agent server, the redirect server does not accept or terminate calls.
Registrar Server
A server that accepts REGISTER requests. The register server may support authentication. A registrar server is typically co-located with a proxy or redirect server and may offer location services.
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SIP Messages – Methods
SIP Methods:
INVITE – Initiates a call by inviting user to participate in session. ACK - Confirms that the client has received a final response to an INVITE request. BYE - Indicates termination of the call. CANCEL - Cancels a pending request. REGISTER – Registers the user agent. OPTIONS – Used to query the capabilities of a server. INFO – Used to carry out-of-bound information, such as DTMF digits.
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SIP Messages –Responses
SIP Responses:
1xx - Informational Messages. 2xx - Successful Responses. 3xx - Redirection Responses. 4xx - Request Failure Responses. 5xx - Server Failure Responses. 6xx - Global Failures Responses.
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통신설정 과정
SIP을 이용한 6가지 통신설정 단계
Registering, initiating and locating the user. Determine the media to use – involves delivering a description of the session that the user is invited to. Determine the willingness of the called party to communicate – the called party must send a response message to indicate willingness to communicate – accept or reject. Call setup. Call modification or handling – example, call transfer (optional). Call termination.
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SIP - Session Initiation Protocol
IETF RFC2543, March 1999 Function
SIP Redirect Server
2
Conveyed as message body
Based on IP Interoperability Scalability Platform for applications
5
4 1
6 11
12
Benefits
Location Service
3
Independent of session
Invite user to interactive sessions Terminate and modify sessions
7
SIP Proxy
1 0 8 9
SIP Client
SIP Proxy
Request
SIP Client (User Agent Server)
Response
2002-7-24
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95
SIP/ SAP/ SDP
SIP – Session Initiation Protocol
SAP – Session Announcement Protocol
Application layer signaling protocol Text, HTTP-like Extensions being drafted for Calling party privacy, distributed call signaling (DCS), local number portability (LNP) SIP+ is now SIP-T (or SIP-BCP) for Telephony Interworking Multicast multimedia session directory Announcers and listeners
SDP – Session Description Protocol
Not really a protocol Also used by MGCP
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H.323 and SIP
H.323
SIP
Widely deployed (in gateways and PC’s) Full backwards compatibility Multimedia ready Complex protocol (bulky code size) Firewall friendly User friendly Shorter call setup time Easy implementation (smaller code size) 새로운 표준으로 우세한 상황으로 전개되고 있음.
Hot Topic: SIP and H.323 Interworking
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97
H.323 vs. SIP
H.323 ITU Standard Binary-based ASN.1 encoding Utilizes other ITU standards such as ISDN (Q.931) and Q.SIG model Defines call signaling, call transfer, codec negotiation Network services are provided by Gatekeepers Direct endpoint or Gatekeeper routed call signaling Uses RTP for media
2002-7-24
Krnet 2002
SIP IETF Standard ASCII text-based Leverages other Internet standards such as HTTP, DNS, MIME, etc. Defines only session initiation and termination (call signaling) Network services provided by SIP servers (Proxy, Redirect, and Registrar) Endpoint signaling or “record-route” signaling Uses RTP for media 98
Why SIP and H.323 Must Co-exist
Established base of large-scale H.323 networks
SIP is the new “buzz” word
ITSPs have been around for several years Equipment vendors only supported H.323 until recently Gaining industry momentum (embedded SIP clients in future operating systems and PDAs) Protocol based on Internet standards Concept of presence in the network brings with it new and exciting services
As a result, existing ITSPs are now beginning to offer SIP services along with their existing H.323 services
2002-7-24
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99
System Decomposition
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100
ITU H.323 Decomposition Service Agent User GK
Gatekeeper
Network GK
H.323 gateway
분산시스템 개방형 인터페이스 독립적 발전가능
Media Gateway Controller MGCP Media Gateway
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101
스위치의 구현기술 진화모델 Provider Control 800
교환기 기본기능 만 내장한 교환기
2002-7-24
교환기 다기능을 내장한 교환기
교환기
AIN
교환기
NGN
Vendor 교환기 Control
SCP 의 차세대 지원을 지능망구조로 받는 교환기 다기능을 외부로 부터 지원받는 교환기
Krnet 2002
API
교환과 제어가 완전분리된 교환기
102
IP Gateway 의 기능요소
제어정보 ( Control )
Telephony Protocols IP Call Control Protocols
ISUP
• H.323 (ITU) • SIP(IETF)
이용자정보 (Media)
PCM streams
GSTN Side
• 64kbps
IP Packets • Jitter • Delay • Sequence 변경
2002-7-24
Voice Stream PCM
Krnet 2002
Signaling Gateway
Media Gateway Controller Media Gateway
IP Side
Voice Stream IP Packet
103
Gateway Decomposition IETF SG: Signaling Gateway MGC: Media Gateway Controller MG: Media Gateway
Gateway Function SG
SS7, Q.931
SSP
Sigtran
MGC
PSTN
IP Network
Megaco
Bearer Channels
2002-7-24
IP Signaling (e.g. H.323, SIP)
MG Krnet 2002
Packetized Media (e.g., RTP) 104
VoIP Gateway의 구성(IETF) SG: Signaling Gateway MGC: Media Gateway Controller MG: Media Gateway
IP Device
PSTN
PSTN
SS7, SSP Q.931 SG
SCTP
MGC
MGCP, MEGACO, H.GCP/H.248
Gateway
Bearer Channels
2002-7-24
SIP+
MGC
MG
RTP/RTCP
IP Network Krnet 2002
SCTP
SG
SS7, SSP Q.931
Gateway
MG
Bearer Channels
105
VoIP Standards Applications
(…)
(…) Call Control & Service Control
H.225 Annex G
SIP-T or BICC
Call Control & Service Control
H.225
H.225
Bearer Control
H.245 SDP
Bearer Control
MGCP/Megaco MGCP/Megaco /AAL1/2/5 RTP
Media
2002-7-24
Media
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106
Gateways
Media Gateway (MG)
미디어 변환기능 수행 e.g., TDM-to-RTP No call states
Gateway의 종류
Trunking gateways • digital circuits으로 PSTN접속
Business gateways • Digital PBX interface, soft-PBX interface
Residential gateways • Set-top boxes, xDSL devices, cable modem equipment
Access Gateways • 소용량 VoIP gateways, MG, CA, MGC
Voice over ATM gateways • ATM network에 접속
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Media Gateways Controller
Media Gateway Controller (MGC)
MGC 는 망내의 지능이 분산가능하도록 함. Call processing, routing, billing, protocol conversion etc. Media gateway의 제어프로토콜을 사용 • SGCP, MGCP • Megaco protocol, H.248
Controls MG by sending MGCP commands to create, modify, delete connections Receives PSTN and Internet signalling Maintains overall call state Monitors MG resources Makes call routing decisions Records call detail records
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108
VoIP Controllers
ITU H.323 Gatekeeper
IETF SIP server
Name Address translation : Alias IP @ Admission control : CAC, authorisation of access to a LAN Bandwidth control Zone management : registration of terminals, gateways and MCUs Invitation to multimedia conferences Name Address Translation, Capabilities negotiation, Gateway service invocation, ...
IETF MGCP : Media Gateway Control protocol
It provides call control for multiple media gateways It interfaces with signalling gateways
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What is MGCP?
IETF RFC 2705 Media Gateway Control Protocol
Components Call agent or media gateway controller
Media Gateway Control Protocol - A protocol for controlling telephony gateways from external call control elements called media gateway controllers or call agents.
Provides call signaling, control and processing intelligence to the gateway. Sends and receives commands to/from the gateway.
Gateway
Provides translations between circuit switched networks and packet switched networks. Sends notification to the call agent about endpoint events. Execute commands from the call agents.
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Components Call Agent or Media Gateway Controller (MGC)
Call Agent or SIP Media Gateway H.323 Controller (MGC)
MGCP
MGCP
Media Gateway (MG)
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Media Gateway (MG)
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단순화한 호흐름(Call Flow)
Phone A 가 오프후크하면, Gateway A 는 Call agent에서 신호를 전송 Gateway A는 발신음을 송출하고, 이용자가 입력한 번호를 수신함. 이 번호는 call agent로 전달됨. call agent는 호 라우팅을 결정함. call agent가 Gateway B에 명령을 내림. Gateway B 는 phone B에 호출음 송출 call agent 는 양측 게이트웨이에 RTP/RTCP sessions을 개설하도록 명령을 내림.
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Call Flow
MGCP
A user picks up analog phone and dials a number. The gateway notifies call agent of the phone (endpoint) event. The Call agent determines capabilities, routing information, and issues a command to the gateways to establish RTP/RTCP session with other end.
2
Call Agent/ Media Gateway Controller
1 Analog Phone
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RTP/ Gateway A
RTCP
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Gateway B
Analog Phone 113
MGCP Commands
Call Agent Commands:
EndpointConfiguration NotificationRequest CreateConnection ModifyConnection DeleteConnection AuditEndpoint AuditConnection
Gateway Commands:
Notify DeleteConnection RestartInProgress
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Characteristics of MGCP
MGCP:
A master/slave protocol. • Assumes limited intelligence at the edge (endpoints) and intelligence at the core (call agent). • Used between call agents and media gateways. • Differs from SIP and H.323 which are peer-to-peer protocols.
Interoperates with SIP and H.323.
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Decomposed Gateway Components
Signaling Gateway (SG)
대역외 신호방식(SS7, Q.931)의 신호를 종단하여, MGC로 전달 (Sigtran) 신호변환기능은 수행하지 않음. (i.e., no ISUP-H.323 interworking) SS7 SSP
Signaling Gateway
App
MGC/ Softswitch App
IWF
ISUP
ISUP
MTP-3
MTP-3
M3UA
M3UA
MTP-2
MTP-2
SCTP
SCTP
IP
IP
IWF = Inter-working Function M3UA = MTP-3 Adaptation Layer 2002-7-24
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116
SCTP Protocol Features
Connection oriented
Reliable packet delivery
Fail-over to a redundant link(s), but no load sharing Supports IPv4 and IPv6 addressing
TCP-like flow control and congestion avoidance
Selective retransmission of lost/corrupted packets Detection of duplicate packets Packet integrity protected by checksum and verification tags
Explicit support for multi-homed hosts
Multiple control “streams” per port In-sequence delivery within a “stream”, if desired
Internet friendly
Fragmentation of long packets on network MTU boundaries Bundling of multiple small chunks into single transmission unit
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VoIP와 Next Generation Network
2002-7-24
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네트워크 발전추세
음성과 데이타망의 통합
광대역 전송 기술의 도입
XoIP – 모든 미디어가 IP로 전송됨. QoS 만족을 위한 전송망의 진화 유선분야 : xDSL, Cable, Optical network 무선분야 : W-CDMA, W-LAN, Bluetooth
Content 및 application을 망에서 제공
Personal rich media Storage networks Application Services
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차세대통신망의 진화경로
PSTN 및 지능망
지능망서비스를 인터넷에서 제어
Hybrid Networks Internet 및 데이타망
Internet Telephony
VoP Networks
Next Generation Networks
ALL IP 핵심망 무선통신망
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서비스 제공구조
현재의 voice 서비스 구조
현재의 IP network architecture
PSTN 에 기반, Services Nodes 와 SCP 사용 제한된 customer data features Back end provisioning – no customer interaction Limited customer control of new features or services 데이터 서비스 중심이며, 제한된 음성서비스 제공 Easy customer or end user provisioning - Web Flexible environment for new features and services Powerful platform for integrated data, voice and video services
통신사업자의 목표는 음성과 데이터 서비스를 통합하고, 공통의 IP 기반망 플랫폼에서 서비스를 제공하고자 함.
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Hybrid Network 개념
PSTN의 지능망과 IP 데이타망/VoIP망과 연동하는 망서비스 Web 기반 서비스 관리
데이터 저장 및 접근
LDAP(Lightweight Directory Access Protocol) 이용
Service Invocation and Call Control
PSTN의 지능망 서비스의 내용을 웹에서 변경 적용가능분야 ; 평생전화번호서비스의 번호변경 예 : 대만의 청화통신
PINT(PSTN and Internet Internetworking) SPIRITS(Services In the PSTN/IN Requesting Internet service)
New Applications for specialized resources
NTT의 L-mode
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Network Architecture STP Call Agent
Call Agent
MGCP RGW
RGW
TCAP/SS7 SS7 Gateway ISUP/SS7 MGCP
Internet
Trunking Gateway
PSTN
Voice
Residential Gateway(RGW) Trunking Gateway(TGW) Call Agent - Media gateway controller SS7 Gateway
2002-7-24
SCP
Krnet 2002
MGCP : Media Gateway Control Protocol SIP : Session Initiation Protocol SCP : Signaling Control Point STP : Signaling Transfer Point ISUP : ISDN User Part TCAP : Transaction Capabilities Application Part
123
Protocol Support Application Specific
Application Layer • Call intelligence • Service creation/ execution • Mgmt of provisioning
Call-Control Layer (Softswitch)
AS or SCP
AS or SCP SIP TCAP
IP: SIP, SIP-T
MGC
• Resource mgmt • Bearer control • Call routing/translation • IN/AIN
MGC
Control Protocols M3UA/SCTP (Sigtran) VoATM: BICC TDM: ISUP, Q.931, CAS
Signaling Layer • Signal processing • Signal control
Inter-App Protocols
Signaling Protocols
SG
MEGACO/H.248
SG
Media Layer • Media processing • Media control • Tone & announcement • QoS 2002-7-24
TDM, IP/RTP, ATM
MG
Media Transport
Krnet 2002
MG
124
Next Gen Network
One Network for all Service 특성
모든 형태의 서비스를 제공하기 위해서 새로운 제어, 관리 및 신호기술을 사용하는 packet 기반 네트워크 음성과 데이타망이 분리되지 않음. 다른 서비스에 대해서 Overlay망의 형태로 제공되는 것이 아님. 음성, 데이터, 멀티미디어가 IP 중심으로 전달되며, 각 미디어 특성에 맞는 QoS가 제공됨. 개방형 구조를 가진 플랫폼상에서 구현됨. 이때, 전송과 교환이 분리된 형태를 가짐.
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NGN Layered Architecture
separating control/services from transport functions single packet based transport infrastructure (voice + data)
Softswitch
Network Service Layer
Control Layer Media Layer
Media Gateway
Access and Transport Layer
Enterprise Customers Remote Office/SOHO
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Mobile
Residential Users
126
개방형 네트워크 기술표준화동향
MSF(Multiservice Switching Forum)
Telcodia, Cisco, MCI WorldCom등 14개 통신관련 회사를 주축으로 98년8월에 구성 ATM, Frame Relay, IP, 영상 서비스 등을 모두 하나의 전송 및 교환 인프라로 수용할 수 있는 방안에 대해서 reference architecture를 제시하고 구성기능 및 기능간의 개방형 인터페이스에 대한 정의
ISC(International Soft Switch Consortium)
Voice over IP를 중심으로 Cisco, Lucent등과 같은 대형 통신업체와 Venture 기업까지 포함한 조직으로서 MGC(Call Agent)의 architecture, 상호 연동성 등에 대한 활동 주로 Soft Switch를 개발 하거나 제품을 가진 업체들이 중심이 되어 활동
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MSF VoP Architecture Service Service Agent
Applications Plane
SS7 Signaling Gateway Control Plane
Call Agent
Call Agent
Signalling Controller
Proxy Signaling
Switching Plane
Switching Fabric Adaptation Plane
TGW
TGW
Voice TDM 2002-7-24
Voice TDM Krnet 2002
128
An NGN VoP Architecture Service Agent
Call Connection Agent
SCP
IP Phones, PCs, H.323 Terminals
STP SS7 Gateway
Customer Gateway
2002-7-24
Access Network
Access Gateway
Core Packet Network Voice Streams
Trunk Gateway
NB Trunks
Krnet 2002
LE
129
International Softswitch Consortium
목적 :
to support rapid advancement of application development for the evolving Internet protocol networks which support both voice and multimedia communications. Internet protocol networks are built on distributed call control servers generally called "call agents," "media gateway controllers,” "softswitches," and "media gateways".
구성 :
1999.05. 창립, 2000.07. 현재 15여 기관 참여 중 • Cisco, HP, Lucent, Nortel, Telcordia 등
WG : Application, Architecture, Device Control, SIP, Session Mn.
활동 사항 :
IP 망을 기반으로 기존 망과의 연동 기능에 대한 구조 제시 • VoIP
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Basic Softswitch Concepts
A conventional voice switch has three components:
Switching matrix Line cards Intelligence
In new voice networks:
The switching matrix is now the IP network The line cards are now gateways The intelligence is a softswitch There is a single virtual distributed switch The new distributed switch must communicate with legacy switches
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SoftSwitch Model Circuit Switch Model
SoftSwitch Model Services
Services
Controller *GCP
T D M
Time Slot Interchange
Line Card
2002-7-24
T D M
T D M
Trunk Card
*GCP
T D M
Packet Data Backbone Media Gateway
Krnet 2002
Soft Switch
Media Gateway
132
Classic Softswitch SS7 Network
SCP
SG
Local Carrier
SG Local Carrier
IMT
MGC Service Routing/Call Control/RM
IMT
MGCP/ MEGACO
Managed Packet Network
MG
ISDN PRI MRS
FS App Server
MG
ISDN PRI
ISP B
PBX Third Party Apps
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133
Softswitch의 연동 Softswitch1 STP SoftSwitch
Softswitch2 BICC SIP-T
SoftSwitch
SS7 SIP ISUP ANSI/ITU
LE C
MEGACO/H.248
Trunk MG
Trunk MG
IP/ATM
Access MG
LE C
DLC Fax SIP
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Fax
134
IP 기반 통합 서비스망 구조
wireless network wireless access
sub mgmt messaging security billing network admin
sub mgmt messaging security billing network admin
sub mgmt messaging security billing network admin
Today’s Business Environment
wireline network
Internet
wireline access
data access
Next Generation Communications Networks intelligent unified central sub mgmt message security
central billing
network admin
Internet / packet network (ATM / IP)
wireless
wired
data
Content-driven environments Seamless, integrated service Access distinctions minimized wireless and wireline voice and data
서비스 내용의 중복 특정망에 적용된 컨텐츠
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Service Intelligent IP Network Vision Application Layer
Programmability 제3자 응용
응용 Hosting
메시징 응용
• 제3자 응용 • Web access • Open APl's
Web 기반 서비스 선택
Application Mediation Layer
Service Control Layer
Network Layer
IP Service and Application Control • • • •
Policy Server LDAP based
Soft Switch
QoS Authentication Security Multicast Metro Optical RAS DSL
Multimedia Call Control • VoIP and Video • H.323/SIP • MGCl (H.248)
IP Service Switch lP/MPLS Core
Aggregation Network
Media GW
PSTN
Cable Frame/ATM
ASP
Wireless
2002-7-24
Internet
Enterprise
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NexGen Service Architecture: Service provider / ASP application servers
Utility Services
Communications Services
Messaging Services Commerce Services
Customer Profiles
Internet
Directory Servers
Content Services
Secure Intranets
IP-based Transport and Signaling Wireless voice/data
IN/AIN/WIN Packet Core
PSTN
2002-7-24
Wireless Data
Wireless data
Wireline voice Web appliances
Entertainment Services
Dialup/broadband
Packet voice
Krnet 2002
137
VoIP 응용분야
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VoIP 응용분야의 주요이슈
One Pipe access for all comm. Services Unified IP infrastructure
VoDSL case
xDSL VoDSL
Wireless Internet Case
GSM+GPRS IMT-2000 ALL IP Network
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139
고속 인터넷 접속서비스 IAS: Internet Access Server DSLAM: DSL Access Multiplexer PSTN/ISDN
IAS
Modem
Data Network DSLAM
ISP-1
POP
ISP-2
POP
ISP-3
음성과 데이터의 결합 - 접속망은 Hybrid - 핵심망은 분리
ADSL Modem
2002-7-24
POP
Krnet 2002
140
VoDSL (Voice over DSL)
음성과 데이터를 다중화하여, ATM over xDSL의 형식으로 전송 복수의 음성채널을 하나의 AAL2 VC로 다중화함. 데이터 채널은 AAL5 VC로 전송
Power Spectrum POTS
상 향 100 kHz
하향 Freq 1 MHz
ADSL frequency usage 2002-7-24
Voice Data AAL2 AAL5 ATM xDSL Frequency
Non-voiceband frequency Krnet 2002
141
VoDSL (Voice over DSL) 연결
V - IF
Transmitted by AAL2(I.366.2)
Subscriber Switch
VoDSL CLAD IAD DSLAM
LAN
ATM-SW
Router
ISP
VC for Voice VC for IP
2002-7-24
VC for Voice and VC for IP are separated by ATM Switch
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142
제2단계 VoDSL ATM-SW IAD
DSLAM
Media GW Controller
Media Gateway
PSTN Switch
Router LAN ISP
Access Gateway가 Media GW와 MGC로 대체됨. MGC에서 교환기능을 수행함.
2002-7-24
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VoDSL Solution Characteristics
VoDSL Solution Characteristics
Based on ATM, not on IP
Today supports up to 16 lines of toll-quality voice plus data on as little as 384 kbps DSL connection Leverages existing DSL deployments (both ADSL and SDSL) Voice packetization based on native ATM using AAL2
Why ATM?
ATM is there: most DSL is ATM-based 대역폭이 효율성이 우수함. QoS가 제공되는 것이 입증되었음. No security concerns (using PVCs) Almost no local service providers looking for a VoIP solution
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144
Current VoDSL Network Voice Gateway
IAD
GR 303
Class 5 Switch
PSTN
xDSL
DSLAM
Packet Backbone
xDSL IAD
Broadband Access Server
ISP1
ISP2 IAD = Integrated Access Device DSLAM = DSL Access Multiplexer 2002-7-24
Krnet 2002
145
GSM Architecture GMSC
PSTN
A-bis
BTS
2002-7-24
„A‟
MSC MAP
BSC
Krnet 2002
GPRS Register H/VLR
HLR/AUC GPRS Register
146
GPRS
GPRS(General Packet Radio Service)
3GPP packet data services standards a packet-mode technique to transfer data and signaling Applications based on standard data protocols and SMS Interworking defined with IP networks and X.25
By adding GPRS functionality to the PLMN
• Optimizes the use of network and radio resources • Value added services - internet • Reasonable charging: amount of translated data, QoS, connection time
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무선망에서의 인터넷접속구조(GPRS) GMSC
PSTN Circuit Switched Traffic A-bis
BTS
„A‟
MSC
MAP
Peer Elements
BSC
GPRS Register H/VLR
PCU Gb
MSC : Mobile Switching Center GMSC : Gateway MSC PCU : Packet Control Unit SGSN: serving GPRS support Node GGSN: gateway GPRS support Node
2002-7-24
Packet Traffic
MAP
SGSN Gn
GGSN
Krnet 2002
HLR/AUC GPRS Register
Gi
Public Switched Packet Network
148
GSM/GPRS Network Configuration MAP
BTS BSC
HLR
MSC/ VLR
PSTN
BTS
BTS
Intranet
BSC SGSN
BTS
GGSN
BSC
Internet
Radio Link GTP Tunnel TCP/IP connection .GPRS is a IP packet network for P-to-P packet data transportation 2002-7-24
Krnet 2002
149
VoIP over GPRS End-to-end IP codec RTP UDP IP
codec RTP UDP IP
회선교환 network
GW BSS
codec RTP UDP IP
2002-7-24
SGSN
IP
GGSN
Public IP
Protocol stack in the MS
codec RTP UDP IP
Krnet 2002
150
VoIPoGPRS : 이용자 평면
CODEC RTP UDP IP
IP Relay
SNDCP
SNDCP
GTP
GTP
LLC
LLC
UDP / TCP
UDP / TCP
IP
IP
Relay
RLC
RLC
BSSGP
BSSGP
MAC
MAC
Network Service
Network Service
L2
L2
GSM RF
GSM RF
L1bis
L1bis
L1
L1
Um
MS 2002-7-24
Gb
BSS
Gn
SGSN Krnet 2002
Gi
GGSN 151
VoIPoGPRS의 개념정리
VoIPoGPRS 은 GSM voice services을 대체하는 것은 아님.
VoIPoGPRS는 다음의 기능을 제공함.
No bandwidth gain on the radio interface Lower speech quality (lower Codec quality, delay 와 loss 증가) 단말기가 변경되어야 하고, 게이트웨이 및 Call Agent 가 요구됨. “click-and-talk”, “browse and phone” 등과 같은 멀티미디어 서비스와 결합기능제공 Particular long-distance tariff distortions Integrated end-to-end multi-media 통신으로의 진화경로
실시간 패킷음성전달에 표준의 개발이 요구됨.
QoS renegotiation or multiple QoS per PDP-Context Mobile constrained Coding-Scheme New SNDCP features such as RTP/UDP/IP compression
2002-7-24
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152
IMT-2000의 표준환경 Modular IMT-2000 Harmonization (Terrestrial Component) IMT-DS (Direct Spread)
IMT-MC (Multi-Carrier)
IMT-TC (Time-Code)
IMT-SC (Single Carrier)
IMT-FT (Frequency-Time)
조각 2분기 1 3분기 4분기
Flexible connection between Radio modules & Core Networks based on operator needs
Core Networks
Evolved GSM (MAP)
Inter-Network Roaming
2002-7-24
Evolved ANSI-41
IP-based Networks
Network-to-Network Interfaces Krnet 2002
153
IMT-2000의 All IP Network
ALL IP 망
IMT-2000 이동통신망을 IP기술로 구성하여 패킷망을 기반으로 하여 실시간, 비실시간 데이터를 동시에 서비스할 수 있도록 함. 패킷음성서비스, 멀티미디어 서비스 제공
All IP network의 특성
이동통신망과 인터넷의 결합 IP protocols을 사용하여 패킷전송 IP 기반의 호처리 프로토콜사용 IP 수용하는 단말사용 실시간 IP 기능을 가진 radio access network 및 Cable과 HIPERLAN을 포함한 다른 접속방식을 지원함.
2002-7-24
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154
ALL IP Network의 목표
Goals of ALL IP Network
Seamless High-Speed Network, supporting a variety of End to End Mobile Multimedia Services with IPv6 functions High Quality (IP QoS) Cost Reduction Simple Mobility Management, only through IP Addressing Open and Rapid Services Development Environment
Approach
Start point is 2G : separate voice and data with internet access 3G Release 99 : gives a higher performance and some important additional service capabilities 3G Releases 4 and 5 : gives the opportunity to transition to integrated multimedia service
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155
ALL IP 망에서의 주요 이슈
IP Transport
이동성 관리
GPRS(3GPP) 및 PPP-MIP(3GPP2)에 기반한 망에서 진화함에 따른 망의 구조를 도출하고, 망기능요소를 정의하며 이들 요소들 사이에 인터페이스를 설계 무선접속망에 IP의 도입 (IP BTS) 전송기능과 제어기능을 분리하여 독립적으로 구현 음성 및 실시간 서비스에 패킷기술을 도입함으로써, Mobile IP와 같은 이동성 관리 기능의 성능개선 핸드오버의 성능개선
QoS
end-to-end QoS를 위한 백본망의 개선 QoS 구조로서 RSVP 나 DiffServ 등의 도입방안
2002-7-24
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156
ALL IP 망에서의 주요 이슈
VoIP
VoIP의 기능을 무선통신망에 도입(multimedia domain) 새로운 호제어, 미디어 제어기능을 핵심망의 요소로 포함
신속한 서비스의 도입
개방형 구조를 취함으로서 신속한 서비스의 생성 및 진화가 가능토록함.
2002-7-24
Krnet 2002
157
3GPP All IP의 개념 Release 99 Architecture - Hybrid PSTN/ISDN 회선모드단말
Circuit domain
Iu-cs UTRAN 패킷모드단말
Iu-ps
packet domain Internet
•hybrid circuit/packet architecture •2 separate domains - circuit and packet
Pure “All IP” Architecture
GW
회선모드단말
UTRAN /
PSTN/ISDN
packet domain Internet
Iu-ps
GERAN packet mode terminal
2002-7-24
long term vision •call control H.323 or SIP • no support for circuit mode terminals Krnet 2002
158
3GPP All IP의 개념(2) Pure “All IP” Architecture
GW
회선모드단말
UTRAN /
PSTN/ISDN
packet domain Internet
Iu-ps
GERAN packet mode terminal
Pure “All IP” CS option Architecture 회선모드단말
GW UTRAN /
GW
PSTN/ISDN
packet domain Internet
Iu-ps
GERAN packet mode terminal
2002-7-24
Krnet 2002
159
3GPP All IP Network Architecture Legacy mobile signaling Network
Applications & Services SCP Alternative Access Network
R-SGW Ms
Mh
Multimedia IP Networks
CSCF
Mw Mm
HSS
CAP
CSCF
Cx Gr TE
TE
Um
Iu-ps' Iu UTRAN
MT R
Gc
Uu Gn SGSN
2002-7-24
SGSN
GGSN Gn
Gp
Mg
MRF Gi
ERAN
MT R
Mr
Gi
Gf
Gi MGCF
Gi
T-SGW
Mc MGW
PSTN/ Legacy/External
EIR
GGSN
GPRS Based
Other PLMN
Signalling Interface Signalling and Data Transfer Interface
Krnet 2002
160
New Functional Elements
Call State Control Function (CSCF)
Home Subscriber Server (HSS)
유선망과 신호연동
Roaming Signalling Gateway Function (R-SGW) Media Gateway Control Function (MGCF) Media Gateway Function (MGW) Multimedia Resource Function (MRF)
HLR
Transport Signaling Gateway Function (T-SGW)
Call Agent
Intelligent Peripheral
MSC Server(for Option 2) Gateway MSC Server(for Option 2
2002-7-24
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161
IMT-2000 ALL IP – 망 참조모델 Legacy mobile network MSC/VLR HLR/HSS
TE
MT R
Um
BS BSC/ RNC
TE
Feature server
legacy Gateway
다양한 기술에 의한 무선접속망 BTS/Node B
MSC server Signaling Gateway ISDN PSTN
다양한 무선접속 기술에 대한 투명한 접속 인터페이스
Media Gateway Controller
IP Network
Media Gateway Gateway router
Session Control/ multimedia call server
Other IP Network
Operation/ Management/ Accounting/ Billing /authentication
IP based Network / IP Multimedia Domain 2002-7-24
Krnet 2002
162
차세대 무선망의 개념 Services and applications
download channel
New radio interface
DAB DVB
IP based core network
return channel: e.g. GSM
cellular GSM
IMT-2000 UMTS
Wireline xDSL WLAN type
other entities short range connectivity
2002-7-24
Krnet 2002
163
Beyond IMT-2000 System Overview 동기/ 비동기 3G Access
30Mbps 급 Radio Access
Um
cell access node
Multi-service IP MM Subsystem
IP기반 3G/3.5G Network
RAN Servers
Legacy Mobile Network
UE
PSTN/ISDN
All IP RAN (high mobility)
Intelligent Edge for 4G core
cell Access node
IPv6 기반 광대역 backbone network
IWU
4G Integrated WLAN
AP
(low mobility) AP 11Mbps 급 Radio Access
100Mbps 급 Radio Access
2002-7-24
Krnet 2002
광/IP 기반 초고속 기간망
Conventional WLAN (Internet Access) 164
감사합니다. 한국외국어대학교 정보통신공학과 김희동
2002-7-24
Krnet 2002
165
