VoIP 2002.7.24 한국외국어대학교정보통신공학과김희동 031-330-4254 016-9712-0000 kimhd@hufs.ac.kr 2002-7-24 Krnet 2002 1
발표내용 개요 PSTN의이해 ( 신호망, 지능망 ) VoIP관련기술 ( 음성신호처리 ) VoIP 표준화개요 Voice over IP (VoIP) signaling의개념 VoIP 와 NGN VoIP 시스템의구현기술 VoDSL All IP network (IMT-2000) 2002-7-24 Krnet 2002 2
전화교환망의개관 2002-7-24 Krnet 2002 3
현재의회선교환망 SS7 EO EO Toll Toll Toll Tan EO EO Tan Toll EO Tandem/Toll Network 2002-7-24 Krnet 2002 4
교환기의명칭 Local Switch EO (End Office) : 단국 CO (Central Office), Class 5 Switch Transit Switch Tandem Switch : 중계교환기 Trunk Switch Access Tandem Class 4 Switch 기타특수교환기 SSP (Service Switching Point) MSC (Mobile Switching Center) : 이동망의교환기 2002-7-24 Krnet 2002 5
PSTN( 전화망 ) 의구조 080 SCP AIN SCP LNP SCP STP Transit Switch PSTN PBX Local Switch IP VMS Local Switch DLC DLC - Digital Loop Carrier IP - Intelligent Peripheral PBX - Private Branch Exchange SCP - Service Control Point STP - Signal Transfer Point VMS - Voice Mail System 2002-7-24 Krnet 2002 6
접속요구 다이얼 발신음 신호방식 국간신호방식 점유신호 번호송출허용 가입자신호방식 호출음 선택신호응답신호 호출신호 접속점유 통 화 송수화기복구 종화신호절단신호 해제요구 복구완료 2002-7-24 Krnet 2002 7
가입자신호방식 Station Signaling 교환기 Station Signaling PSTN ISDN Dedicated Network Analog 전화기 이용자선택번호 Pulse Dial DTMF Call Progress Tone Busy tone Dial tone Ringback tone ISDN Terminals BRI(I.430) LAPD(I.441) Networking(Q.931) Digital Telephone Voice, Voice+Data Proprietary Signaling 2002-7-24 Krnet 2002 8
국간신호방식 Trunk Signaling PSTN Signaling Channel Associated Signaling : R2-MFC ISDN Signaling : PRI Common Channel Signaling : No.7 ISDN 교환기 Dedicated Network Trunk Signaling 2002-7-24 Krnet 2002 9
공중 ISDN 과 Signaling System 7 신호망 BRI BRI PBX 1 PRI Switch 전송망 Transmission Network Switch PRI PBX 2 DSS1 Signaling System 7 DSS1 신호망의도입으로 교환기와교환기사이의데이터경로가형성됨. 신호망에컴퓨터를접속하여, 교환기와컴퓨터가상호연동가능함. 지능망의개념이탄생함. 2002-7-24 Krnet 2002 10
SS7 - Protocol View GSM MAP IS-41 TCAP SCCP INAP ISUP MTP Level 3 MTP Level 2 MTP Level 1 TUP 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) 2002-7-24 Krnet 2002 11
SS7 신호망의요소 SCP Network 1 Network 2 SCP SCP STP STP SCP SSP STP STP SSP SSP SSP Voice Trunk Signaling Link SSP: Signal Switching Point STP: Signal Transfer Point SCP: Service Control Point 2002-7-24 Krnet 2002 12
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 2002-7-24 Krnet 2002 13
SS7 ISUP Originating Switch STP STP Terminating Switch IAM ACM ANM IAM ACM ANM IAM ACM ANM REL RLC REL RLC REL RLC 2002-7-24 Krnet 2002 14
지능망으로의구현기술진화모델 800 AIN Provider Control NGN API 교환기교환기교환기교환기 기본기능만내장한교환기 다기능을내장한교환기 SCP 의지원을받는교환기 차세대지능망구조로다기능을외부로부터지원받는교환기 Vendor Control 교환기 교환과제어가완전분리된교환기 2002-7-24 Krnet 2002 15
지능망서비스의흐름 SCP 4) 5) DB 검색 SDP SS7 network 3) SCP 에문의 1) 지능망호발신 2) 호전달 6) SCP 에응답 7) 호발신 단국 SSP 단국 2002-7-24 Krnet 2002 16
지능망서비스의흐름 SCP 4) 5) DB 검색 SDP 1) 0502-500- 5425 SS7 network 05025005425? 0313304254 2) 호전달 7) 호발신 031-330-4254 단국 SSP 단국 2002-7-24 Krnet 2002 17
VoIP 기술개요 2002-7-24 Krnet 2002 18
VoIP 의개요 용어 VoIP ( Voice Over IP) VoP ( Voice over Packet Network) IP Telephony 개념 패킷교환망을통하여음성서비스를제공하는기술및시스템 등장배경 Internet의활성화와함께, 패킷망이통신망의주류를형성 패킷망에음성신호를전달하는인터넷폰으로통신비용절감 PC-to-Phone, Phone-to-Phone의개념으로확장 음성서비스를패킷망을통하여전송하도록확장 NGN(Next Generation Network) 의개념도입 2002-7-24 Krnet 2002 19
PSTN versus VoIP Today s PSTN VoIP Underlying Technology TDM 회선교환 Packet switching QoS guarantees Yes No Network resource reserved at call setup Network elements Call processing intelligence Yes Class 4, Class 5 교환기 대부분교환기에위치 No Gateways, gateway controllers, routers In separate gateway controllers Bandwidth per call 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 2002-7-24 Krnet 2002 20
QoS 송신측에서의음성압축네트워크측면의기술수신측에서의기술 VoIP Issue 패킷망에의한음성전달기술 음성압축, QoS, Delay, Jitter Signaling, Enhanced Service 대용량게이트웨이의구현 Carrier grade VoIP 서비스게이트웨이의 Decomposition 신뢰성이높은시스템의구현 다양한서비스의신속한제공 계층적구조개방형구조 Next Generation Network 으로진화 Packet 기술기반, 다양한서비스의수용, 고속접속지원 2002-7-24 Krnet 2002 21
Voice over Internet Protocol IP Transport Protocol RTP Uncompressed Uncompressed C. Speech C. Speech 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,... 2002-7-24 Krnet 2002 22
IP: Internet Protocol Voice/RTP/UDP/IP IP header : 20 octets (IP address origination, IP address destination, ) UDP: User Datagram Protocol no retransmission, no flow control <-> TCP UDP header : 8 octets (port identifier, ) RTP: Real Time Protocol RTP header : 12 octets Payload : encoded voice or multi-media stream IP(20) UDP(8) RTP(12) C. Speech 2002-7-24 Krnet 2002 23
VoIP Protocol Stack Application Application H.323, SIP Codecs Transport Signaling UDP, TCP,... IP (Wireless) Link Physical Codecs Signaling Transport UDP, TCP,... IP (Wireless) Link 2002-7-24 Krnet 2002 24
Next Generation Tandem Network EO SS7 EO Packet EO EO Gateway Switch Packet-Based Tandem Network EO 전화망이모두없어지거나, 통신망이모두 IP phone 으로대치되기전에는게이트웨이에의한연동이필수적임. 2002-7-24 Krnet 2002 25
VoIP 기술의요소 음성을디지털패킷화및전송 Speech Coding : 음성압축, 묵음검출 Packetization : IP 패킷의크기, 헤더의오버헤드 Multiplexing, Trunking 신호방식 Bearer Call Signaling Intelligent Network Call Processing 서비스생성및관리 Supplementary and Multimedia Service Authentication, Mobility and Directory Service 교환및제어구조 MSF SoftSwitch 2002-7-24 Krnet 2002 26
Services Basic Services Packet Voice 요구사항 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 2002-7-24 Krnet 2002 27
Packet Voice 요구사항 ( 계속 ) Digital Signal and Packet Processing Voice Coding : 음질, 전송속도, 복잡도에의해결정 Trans-coding : 압축방식의변환에도음질저하최소화 Silence Removal : 묵음을검출하여묵음시데이터전송않음. Echo canceling : 전송지연에따른엑코현상발생 Packetization (Serialization) : 패킷화지연시간고려 Buffer smoothing : 음성지연 Addressing/Naming & Routing E.164, IPv4/IPv6, DNS and Directory Service IETF ENUM workgroup 2002-7-24 Krnet 2002 28
Call Processing Signaling Mode의수용 Packet Voice 요구사항 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 Loss 에대한대처방안 2002-7-24 Krnet 2002 29
Packet Voice 요구사항 ( 계속 ) Management Security Accounting/Charging/Billing Interworking Bearer Interworking Service (IN) Interworking (080-xxxx, 700-xxxx, etc) Mobility Terminal and Service mobility 2002-7-24 Krnet 2002 30
패킷망에의한음성전달기술음성압축, QoS, Delay, Jitter 2002-7-24 Krnet 2002 31
Voice Quality Definition Echo Clarity Speech Quality Space Delay 음질평가영향요소 : 명료성 Clarity - The 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 2002-7-24 Krnet 2002 32
User Perspective of Voice Quality MOS (mean opinion score) quality: depends on coder bit rate and complexity Echo degradation control : subjective perception is dependent on delay faithful reproduction of background noise Levels : 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. 2002-7-24 Krnet 2002 33
음성의디지털표현 Sampler Quantizer x(t) x(n) = x(nt) x(n) Continuous-time Continuous-amp. Discrete-time Continuous-amp. Discrete-time Discrete-amp. 2002-7-24 Krnet 2002 34
음성부호화방식의종류 Waveform Coding PCM DPCM ADPCM DM ADM CSVD Hybrid Coding APC RELP MPLPC CELP SELP Parametric (Vocoding) Channel Formant LPC MBE 2002-7-24 Krnet 2002 35
Speech Coder 의형태 Waveform coding Time domain Frequency domain 예 : PCM (Pulse Code Modulation) 64kbps Vocoder LPC ( Linear Prediction Coding) Low Rate 음성발생기관을모델링 Hybrid coding RELP ( Residual Excited Linear Prediction) CELP ( Code Excited Linear Prediction) 이동전화, 인터넷폰응용 2002-7-24 Krnet 2002 36
음성발성기관모델 2002-7-24 Krnet 2002 37
음성발생과 Vocoder 원리 10-30 ms 사이에는 Quasy-stationary 폐의여기신호입, 성도모양음성파형 성모모양필터가구해진상태에서여기신호를추출함. 미지수 1 개, 식 1 개 Sub optimal Solution 음성파형으로부터성도의모양필터를추출함. 미지수 1 개, 식 1 개 음성파형으로부터 - 성도의모양필터 - 폐의여기신호를추출함. 미지수 2 개, 식 1 개 2 가지값은저전송속도로수신단에전송하고수신단에서합성함. 2002-7-24 Krnet 2002 38
분석 - 합성방식에의한음성부호화 파형부호화방식으로는 32-64 bps 정도로압축의한계가있음. 분석합성방식이란 음성발생모델을이용하여음성을고능률압축하는방식 폐에서의공기발생 + 성도필터 = 음성 공기발생형태 : Excitation signal 성도필터 : Linear Prediction Filter 송신단에서입력음성을분석하기위해서 프레임구간 (10-30ms) 으로구분우선, 필터계수를구하고 Excitation 신호를구하여, 저전송속도로수신단에송신함. 수신단에서는 파라메터정보를가지고음성을합성함. 전송속도는 2.4 kbps 정도이나음질이문제임. 2002-7-24 Krnet 2002 39
음질을개선하기위해서는 Hybrid Coder 성도필터의계산방법은그대로채용하되, 유성음과무성음을구분하는단순한방법을사용하는대신, Excitation signal 을변화시켜가면서, 신호를합성하여, 입력음성과차이가최소화되면, Excitation signal 의값과필터계수를송신 Excitation Signal 방식 Codebook 을미리작성하고, index 를찾는방식 신호를산술적인계산에의해서찾는방식 Various excitation signal MPE (multi-pulse excited) RPE (regular-pulse excited) CELP (code-excited linear prediction) 2002-7-24 Krnet 2002 40
Hybrid coders Multi-Pulse Excitation Efficient at medium bit rates. A sequence of nonuniformly spaced pulses as an excitation signal Amplitudes and positions are excitation parameters Regular-Pulse Excitation (RPE) 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 2002-7-24 Krnet 2002 41
여기신호의종류 g 1 g 3 n 4 g k a) multipulse n 1 g 2 n 2 n 3 g 4 nk 0 5 10 15 20 g 1 g 2 g 4 g 6 b) regular-pulse K g 3 Codebook g 5 c) Code-excited Linear Prediction Codevector # 1 Codevector # 2 Codevector # 3 2 M = N (M = 전송 bit) Codevector # N 2002-7-24 Krnet 2002 42
분석 / 합성 LPC coder Speech CELP vector1 Vocal Tract Model Filter ~ - S(n) + S(n) e(n) MP-MLQ Error Minimization 2002-7-24 Krnet 2002 43
Frame Processing and Delay G.729 의 linear prediction Linear Prediction LP parameter Frame Delay Lookahead delay 120 samples (past) 80 samples current frame time 40 samples (future) 2002-7-24 Krnet 2002 44
분석 - 합성방식에의한음성부호화 codebook Input speech Frame (10,20,30 ms) Excitation Generation Synthesis Filter Error Minimization Error Weighting codebook 정보전송 (a) Encoder Excitation Generation Synthesis Filter (b) Decoder Reconstructed speech 2002-7-24 Krnet 2002 45
ITU-T G.7xx series 보코더 H.323 용보코더 보코더 비트율 (kbit/s) 지연시간 (ms) 복잡도 (MIPS) 대역폭 (khz) G.711 64 0 <<1 4 G.722 64/56/48 3 10 7 G.726 32/24/16 0.25 1.25 4 G.728 16 1.25 30 4 G.729 8 25 20 4 G.723.1 5.3/6.3 62.5 20 4 2002-7-24 Krnet 2002 46
ITU-T G.723.1 Dual rate (5.3/6.3 kbit/s) ITU-T 표준보코더 1995년말표준확정, 멀티미디어통신용으로개발 대부분의인터넷폰에서표준으로지원 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 2002-7-24 Krnet 2002 47
Annex A G.723.1 Extensions G.723.1 + VAD/SID/CNG(Comfort Noise Generation) 음성이없는구간에서 noise 발생 Noise 의특성을전송 - SID (Silence insertion description) Active noise 와 silent noise 간의자연스러운연결 DTX (Discontinuous Transmission) Noise 특성을전송할필요가있는지결정 Annex B G.723.1 의부동소수점구현 Annex C Scalable channel coding scheme for wireless applications H.324 의일부로개발 Channel bit-rate: 0.7 ~ 14.3 kbit/s Channel codec: punctured convolutional codes 2002-7-24 Krnet 2002 48
Quality Quality, Complexity and Delay excellent DELAY/COMPLEXITY good ITU4 2000+? G.723.1 1995 G.729 1996 G.728 1992 G.726 1984 G.711 1972 clean speech fair poor 2 4 8 16 32 64 bitrate (kb/s) 2002-7-24 Krnet 2002 49
Voice over Internet Protocol IP Transport Protocol RTP Uncompressed Uncompressed C. Speech C. Speech 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,... 2002-7-24 Krnet 2002 50
Voice Coding Bandwidth Bytes G.723.1 G.729a Frame size 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 마다패킷화하여전송하면, 오버헤드줄일수있음. Overhead를줄이기위한방안모색필요 IETF ROHC( Robust Overhead Compression) 2002-7-24 Krnet 2002 51
Packet Loss Characteristics Packet Loss 원인 Source Site - overwrite Intermediate Site - dropping Destination Site - time constraint Packet Loss 대응방안 Packet Repetition GSM 에서사용하는방식 Packet Interpolation Frame Interleaving Bursty Error 의영향을분산시키나지연시간이길어짐. Forward Error Correction 2002-7-24 Krnet 2002 52
Delay Variation Jitter A B C Sender Transmits t A B C D 1 D 2 = D 1 D 3 = D 2 Sink Receives t Queueing delay 의변화에의해발생 Hold time과 Jitter Buffer size의비례관계 Jitter가전체 Delay에미치는영향이더큼. 50-100ms 사이를사용 Delay에따라 Buffer Size가적응되도록함. 2002-7-24 Krnet 2002 53
DTMF 문제 DTMF Transfer 음성부호화방식은음성의특성에최적화하여압축비를높임. Tones: fax tones, dialing tone, busy tone DTMF digits for two-stage dialing or voice-mail DTMF 전송방식 In-band 방식 (G.729 에서사용하는방식 ) DTMF tone 을음성처럼부호화하여전송함. 음성부호화방식이 tone 도정확히코딩, 디코딩할수있어야함. Out-of-band 방식 (G.723.1 의방식 ) DTMF 는음성특성과달라 DTMF 의전달특성이나쁨 Encoder 에서 DTMF 검출 제어패킷으로만들어별도로 TCP 를통하여전송 수신부 (Decoder) 에서 DTMF 톤발생 검출전에 DTMF 톤의일부가전송됨 데이타망의상태에따라지연이발생됨. 2002-7-24 Krnet 2002 54
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 2002-7-24 Krnet 2002 55
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 Gateway IP Network H.323 Terminal Hybrid Echo Cancel. Echo Cancel Line Echo Acoustic Echo Echo tail ( 8-120 ms) 2002-7-24 Krnet 2002 56
Echo 의원인및대책 2 Wire Local Loop Central Office Receive Direction Rx & Tx Superimposed 2w-4w Hybrid Echo 는반사에의해서생김. EC Transmit Direction 인터넷 2w-4w hybrid 에서의 Impedance mismatch 가엑코의발생원인임. Echo 는항상발생하지만, Echo 의지연시간과 Echo 의크기에따라문제로대두됨. 2002-7-24 Krnet 2002 57
VoIP 표준 2002-7-24 Krnet 2002 58
VON 관련표준화기관 ITU - International Telecommunications Union SG 16 multimedia 단말과 security에관한연구 H.323 IETF - International Engineering Task Force web services 와 PSTN/ISDN의상호연동 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) 2002-7-24 Krnet 2002 59
표준화기관의관계 SG16 Multimedia SG2 Numbering SG11 Signaling SG13 Architecture ITU-T T1 권고안 Integrated Keeper spec Global country code Collaboration RFCs TIPHON 표준개발요구 Naming/Addressing QoS Interoperability Requirement Megaco PINT IPTEL SIP IETF 2002-7-24 Krnet 2002 60
VoIP Protocols H.323 (ITU-T) initial push for VON interoperability Multimedia session control - call control SIP (IETF) 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) 2002-7-24 Krnet 2002 61
VoIP Protocols MEGACO (IETF) MEdia GAteway Control; Successor to MGCP H.248 (ITU-T) [formerly H.GCP] ITU variant of MEGACO SIP-T (IETF, Cable Labs) Signaling between media gateway controllers SCTP (IETF Sigtran) Simple Control Transmission Protocol Transport PSTN signaling (SS7, Q.931) over IP 2002-7-24 Krnet 2002 62
H.323 2002-7-24 Krnet 2002 63
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 2002-7-24 Krnet 2002 64
H.323 H.323 Terminal Scope of H.323 Non-Guaranteed QoS LAN H.323 MCU H.323 Gatekeeper H.323 Gateway H.323 Terminal H.323 Terminal N-ISDN Guaranteed QoS LAN B-ISDN GSTN Speech Terminal H.320 Terminal H.322 Terminal H.321 Terminal H.321 Terminal V.70 Terminal H.324 Terminal Speech Terminal H.310 terminal operating in H.321 mode (DSVD) (H.324M) 2002-7-24 Krnet 2002 65
H.323 Functional Block Diagram Scope of H.323 Video I/O Equipment Audio I/O Equipment User Data Applications T.120 etc. Video Codec H.261, H.263 Audio Codec G.711, G.722 G.723, G.728 G.729 System Control H.245 Control Receive Path Delay H.225.0 Layer Local Area Network Interface System Control User Interface Call Control H.225.0 RAS Control H.225.0 RAS - registration, admission, status 2002-7-24 Krnet 2002 66
H.323 defines H.323 표준의범위 System-level characteristics of terminals, gateways, gatekeepers, and MCUs Call signaling procedures (phases A-E) Refers to H.225 and H.245 specs H.225 defines Packetization and use of RTP/RTCP Use of Q.931/2 and RAS messages H.245 defines Procedures for use of Control Protocol messages 2002-7-24 Krnet 2002 67
H.323 is an Umbrella Specification Media Audio Video Codec Codec G.711 H.261 G.723 H.263 G.729 RTP Call Control and Signaling RTCP H.225 H.225 H.245 Q.931 RAS UDP TCP UDP TCP IP 2002-7-24 Krnet 2002 68
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 2002-7-24 Krnet 2002 69
통신과정 H.323의통신과정 5단계 Call setup. Initial communication and capabilities exchange. Audio/video communication establishment. Call services. Call termination. 2002-7-24 Krnet 2002 70
단순한 H.323 Call Setup Gatekeeper GK 1. ARQ 2. ACF 3. SETUP 4. Call Proceeding 7.Alerting 8.Connect H.245 Messages RTP Media Path 5. ARQ 6. ACF RAS messages Call Signaling Messages 2002-7-24 Krnet 2002 71
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. 2002-7-24 Krnet 2002 72
H.323 basic protocols Annex G Gatekeeper Q.931/H.245 Gatekeeper RAS Q.931/ H.245 Q.931/ H.245 RAS Endpoint Signalling (Q.931) H.245 RTP/RTCP Gatekeeper Routed Signaling Direct Routed Signaling Endpoint 2002-7-24 Krnet 2002 73
H.323 Components and Signaling H.225/RAS messages over RAS channel H.225/RAS messages over RAS channel H.225/Q.931 (optional) Gatekeeper H.225/Q.931 (optional) H.245 messages (optional) H.245 messages (optional) Terminal H.225/Q.931 messages over call signaling channel H.245 messages over call control channel Gateway PSTN 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 Krnet 2002 74
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) 2002-7-24 Krnet 2002 75
RAS Messages GRQ/GCF/GRJ (Discovery) Unicast - Multicast Find a gatekeeper RRQ/RCF/RRJ (Registration) Endpoint alias/ip address binding Endpoint authentication ARQ/ACF/ARJ (Admission) Destination Address Resolution Initial bandwidth request Call Routing Call Authorization DRQ/DCF/DRJ (Disconnect) Get rid of call state LRQ/LCF/LRJ (Location) Stateless name - IP address resolution Inter gatekeeper communication IRQ/IRR (Information Request) Ping during active calls Resource information for gateways BRQ/BCF/BRJ (Bandwidth) Ask for more/less bandwidth during call URQ/UCF/URJ (Unregistration) Get rid of registration state 2002-7-24 Krnet 2002 76
RAS - Direct Call Signaling Gatekeeper A ACF LRQ LCF IP Network Gatekeeper B ACF RRQ / RCF RRQ / RCF ARQ H.225 (Q.931) Setup ARQ H.225 (Q.931) Alert & Connect V Gateway A H.245 RTP V Gateway B Phone B Phone A 2002-7-24 Krnet 2002 77
RAS - GK Routed Call Signaling ACF Gatekeeper A A & C LRQ LCF H.225 (Q.931) Setup H.225 Alert & Connect H.245 Gatekeeper B Setup ACF H.245 IP Network H.245 ARQ Setup ARQ Alert & Connect V Gateway A RTP V Gateway B Phone B Phone A 2002-7-24 Krnet 2002 78
H.225 호제어시그널링 (Call Control Signaling) H.225표준, Q931 signal Message Setup Call Proceeding Alerting Release Complete Faclity 호제어처리방식종류 DECS, GKRCS 2002-7-24 Krnet 2002 79
Q.931 Signaling 절차 UNI ATM 망 UNI SET UP CALL PROCESSING CONNECT CONNECT ACK VCC 설정 SET UP CALL PROCESSING CONNECT CONNECT ACK Signaling Cell(ATM Forum) VPI=0, VCI=5 ATM Cell Header Signaling Message 2002-7-24 Krnet 2002 80
H.245 H.245 미디어제어및전송 (Media Control and Transport) H.245 제어동작과정 capabilities 교환 단말기의음성, 비디오및데이터를위한송수신능력 Master-Slave 결정 Round Trip : 송신단말기에서수신단말기까지의지연을결정. 로지컬채널시그널링 2002-7-24 Krnet 2002 81
H.245 System Control Capabilities Exchange Codec negotiation - Transmitter/Receiver capabilities Mode Request - Request specific transmission format Master/Slave Determination Open Logical Channel/Ack Create new audio/video/t.120 stream Commands and Indications out of band DTMF 2002-7-24 Krnet 2002 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) 323 2. H.225 Q.931 signaling- -setup, proceed, alert, connect (TCP) 4. H.245 media negotiation- -TCS, MSD, OLC, OLCA (TCP) 5. RTP media (UDP) 323 2002-7-24 Krnet 2002 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 3. H.225 Q.931 signaling with faststart media negotiation-- setup, proceed, alert, connect (TCP) 1. H.225 RAS ARQ/ACF location/admission (UDP) 4. H.225 RAS ARQ/ACF admission (UDP) 323 5. RTP media (UDP) 323 2002-7-24 Krnet 2002 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 4bytes V E R CC M Payload Type Sequence Number 4bytes 4bytes RTP Timestamp Synchronization Source (SSRC) ID 2002-7-24 Krnet 2002 85
Session Initiation Protocol 2002-7-24 Krnet 2002 86
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) 2002-7-24 Krnet 2002 87
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 2002-7-24 Krnet 2002 88
SIP Distributed Architecture SIP Components Location Server Redirect Server Registrar Server PSTN User Agent Proxy Server Proxy Server Gateway 2002-7-24 Krnet 2002 89
User Agents SIP Components 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). 2002-7-24 Krnet 2002 90
Redirect Server SIP Components 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. 2002-7-24 Krnet 2002 91
SIP Methods: SIP Messages 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. 2002-7-24 Krnet 2002 92
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. 2002-7-24 Krnet 2002 93
통신설정과정 SIP 을이용한 6 가지통신설정단계 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. 2002-7-24 Krnet 2002 94
SIP - Session Initiation Protocol IETF RFC2543, March 1999 Function Invite user to interactive sessions Terminate and modify sessions Independent of session Benefits Conveyed as message body Based on IP Interoperability Scalability Platform for applications 1 SIP Client 2 12 Request Response 4 3 11 SIP Proxy SIP Redirect Server 5 1 0 8 9 SIP Client (User Agent Server) Location Service 6 7 SIP Proxy 2002-7-24 Krnet 2002 95
SIP/ SAP/ SDP SIP Session Initiation 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 SAP Session Announcement Protocol Multicast multimedia session directory Announcers and listeners SDP Session Description Protocol Not really a protocol Also used by MGCP 2002-7-24 Krnet 2002 96
H.323 and SIP H.323 Widely deployed (in gateways and PC s) Full backwards compatibility Multimedia ready Complex protocol (bulky code size) SIP Firewall friendly User friendly Shorter call setup time Easy implementation (smaller code size) 새로운표준으로우세한상황으로전개되고있음. Hot Topic: SIP and H.323 Interworking 2002-7-24 Krnet 2002 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 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 2002-7-24 Krnet 2002 98
Why SIP and H.323 Must Co-exist Established base of large-scale H.323 networks ITSPs have been around for several years Equipment vendors only supported H.323 until recently SIP is the new buzz word 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 Krnet 2002 99
System Decomposition 2002-7-24 Krnet 2002 100
ITU H.323 Decomposition Service Agent Gatekeeper H.323 User GK Network GK gateway 분산시스템개방형인터페이스독립적발전가능 Media Gateway Controller MGCP Media Gateway 2002-7-24 Krnet 2002 101
스위치의구현기술진화모델 800 AIN Provider Control NGN API 교환기교환기교환기교환기 기본기능만내장한교환기 다기능을내장한교환기 SCP 의지원을받는교환기 차세대지능망구조로다기능을외부로부터지원받는교환기 Vendor Control 교환기 교환과제어가완전분리된교환기 2002-7-24 Krnet 2002 102
IP Gateway 의기능요소 제어정보 ( Control ) Telephony Protocols IP Call Control Protocols H.323 (ITU) ISUP Signaling Gateway SIP(IETF) 이용자정보 (Media) PCM streams GSTN Side Media Gateway Controller IP Side 64kbps IP Packets Jitter Voice Stream PCM Media Gateway Voice Stream IP Packet Delay Sequence 변경 2002-7-24 Krnet 2002 103
Gateway Decomposition IETF SG: Signaling Gateway MGC: Media Gateway Controller MG: Media Gateway Gateway Function SS7, Q.931 SG Sigtran SSP MGC IP Signaling (e.g. H.323, SIP) IP Network PSTN Megaco Bearer Channels MG Packetized Media (e.g., RTP) 2002-7-24 Krnet 2002 104
VoIP Gateway 의구성 (IETF) SG: Signaling Gateway MGC: Media Gateway Controller MG: Media Gateway IP Device PSTN PSTN SSP SS7, Q.931 SG SCTP MGC SIP+ MGC SCTP SG SS7, Q.931 SSP Gateway MGCP, MEGACO, H.GCP/H.248 Gateway Bearer Channels MG RTP/RTCP IP Network MG Bearer Channels 2002-7-24 Krnet 2002 105
VoIP Standards ( ) Call Control & Service Control Applications H.225 Annex G SIP-T or BICC Call Control & Service Control ( ) H.225 H.225 Bearer Control H.245 SDP Bearer Control Media MGCP/Megaco MGCP/Megaco /AAL1/2/5 RTP Media 2002-7-24 Krnet 2002 106
Gateways Media Gateway (MG) 미디어변환기능수행 e.g., TDM-to-RTP No call states Gateway의종류 Trunking gateways digital circuits 으로 PSTN 접속 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 network 에접속 2002-7-24 Krnet 2002 107
Media Gateways Controller Media Gateway Controller (MGC) MGC 는망내의지능이분산가능하도록함. Call processing, routing, billing, protocol conversion etc. Media gateway 의제어프로토콜을사용 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 2002-7-24 Krnet 2002 108
ITU H.323 Gatekeeper VoIP Controllers Name Address translation : Alias IP @ Admission control : CAC, authorisation of access to a LAN Bandwidth control Zone management : registration of terminals, gateways and MCUs IETF SIP server 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 2002-7-24 Krnet 2002 109
What is MGCP? IETF RFC 2705 Media Gateway Control Protocol Media Gateway Control Protocol - A protocol for controlling telephony gateways from external call control elements called media gateway controllers or call agents. Components Call agent or media gateway controller 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. 2002-7-24 Krnet 2002 110
Components Call Agent or Media Gateway Controller (MGC) SIP H.323 Call Agent or Media Gateway Controller (MGC) MGCP MGCP Media Gateway (MG) Media Gateway (MG) 2002-7-24 Krnet 2002 111
단순화한호흐름 (Call Flow) Phone A 가오프후크하면, Gateway A 는 Call agent 에서신호를전송 Gateway A 는발신음을송출하고, 이용자가입력한번호를수신함. 이번호는 call agent 로전달됨. call agent 는호라우팅을결정함. call agent 가 Gateway B 에명령을내림. Gateway B 는 phone B 에호출음송출 call agent 는양측게이트웨이에 RTP/RTCP sessions 을개설하도록명령을내림. 2002-7-24 Krnet 2002 112
MGCP Call Flow 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 Gateway A RTP/ RTCP Gateway B Analog Phone 2002-7-24 Krnet 2002 113
Call Agent Commands: EndpointConfiguration NotificationRequest CreateConnection ModifyConnection DeleteConnection AuditEndpoint AuditConnection MGCP Commands Gateway Commands: Notify DeleteConnection RestartInProgress 2002-7-24 Krnet 2002 114
MGCP: Characteristics of 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. 2002-7-24 Krnet 2002 115
Decomposed Gateway Components Signaling Gateway (SG) 대역외신호방식 (SS7, Q.931) 의신호를종단하여, MGC 로전달 (Sigtran) 신호변환기능은수행하지않음. (i.e., no ISUP-H.323 interworking) SS7 SSP App ISUP Signaling Gateway IWF MGC/ Softswitch App ISUP MTP-3 MTP-3 M3UA M3UA MTP-2 MTP-2 SCTP SCTP IWF = Inter-working Function M3UA = MTP-3 Adaptation Layer IP IP 2002-7-24 Krnet 2002 116
Connection oriented SCTP Protocol Features Multiple control streams per port In-sequence delivery within a stream, if desired Reliable packet delivery Selective retransmission of lost/corrupted packets Detection of duplicate packets Packet integrity protected by checksum and verification tags Explicit support for multi-homed hosts Fail-over to a redundant link(s), but no load sharing Supports IPv4 and IPv6 addressing TCP-like flow control and congestion avoidance Internet friendly Fragmentation of long packets on network MTU boundaries Bundling of multiple small chunks into single transmission unit 2002-7-24 Krnet 2002 117
VoIP 와 Next Generation Network 2002-7-24 Krnet 2002 118
네트워크발전추세 음성과데이타망의통합 XoIP 모든미디어가 IP로전송됨. QoS 만족을위한전송망의진화 광대역전송기술의도입 유선분야 : xdsl, Cable, Optical network 무선분야 : W-CDMA, W-LAN, Bluetooth Content 및 application을망에서제공 Personal rich media Storage networks Application Services 2002-7-24 Krnet 2002 119
차세대통신망의진화경로 PSTN 및지능망 Internet 및데이타망 Internet Telephony 지능망서비스를인터넷에서제어 Hybrid Networks VoP Networks ALL IP 핵심망 Next Generation Networks 무선통신망 2002-7-24 Krnet 2002 120
현재의 voice 서비스구조 서비스제공구조 PSTN 에기반, Services Nodes 와 SCP 사용 제한된 customer data features Back end provisioning no customer interaction Limited customer control of new features or services 현재의 IP network architecture 데이터서비스중심이며, 제한된음성서비스제공 Easy customer or end user provisioning - Web Flexible environment for new features and services Powerful platform for integrated data, voice and video services 통신사업자의목표는음성과데이터서비스를통합하고, 공통의 IP 기반망플랫폼에서서비스를제공하고자함. 2002-7-24 Krnet 2002 121
Hybrid Network 개념 PSTN 의지능망과 IP 데이타망 /VoIP 망과연동하는망서비스 Web 기반서비스관리 PSTN 의지능망서비스의내용을웹에서변경 적용가능분야 ; 평생전화번호서비스의번호변경 예 : 대만의청화통신 데이터저장및접근 LDAP(Lightweight Directory Access Protocol) 이용 Service Invocation and Call Control PINT(PSTN and Internet Internetworking) SPIRITS(Services In the PSTN/IN Requesting Internet service) New Applications for specialized resources NTT 의 L-mode 2002-7-24 Krnet 2002 122
Network Architecture Call Agent Call Agent SS7 Gateway TCAP/SS7 ISUP/SS7 STP SCP MGCP MGCP RGW Internet Trunking Gateway PSTN RGW Voice Residential Gateway(RGW) Trunking Gateway(TGW) Call Agent - Media gateway controller SS7 Gateway 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 2002-7-24 Krnet 2002 123
Application Layer Call intelligence Service creation/ execution Mgmt of provisioning Call-Control Layer (Softswitch) Resource mgmt Bearer control Call routing/translation IN/AIN Protocol Support AS or SCP MGC Application Specific Inter-App Protocols IP: SIP, SIP-T Control Protocols M3UA/SCTP VoATM: BICC (Sigtran) TDM: ISUP, Q.931, CAS Signaling Layer Signal processing Signal control SG Signaling Protocols AS or SCP SG MGC SIP TCAP MEGACO/H.248 Media Layer Media processing Media control Tone & announcement QoS MG TDM, IP/RTP, ATM Media Transport MG 2002-7-24 Krnet 2002 124
Next Gen Network One Network for all Service 특성 모든형태의서비스를제공하기위해서새로운제어, 관리및신호기술을사용하는 packet 기반네트워크 음성과데이타망이분리되지않음. 다른서비스에대해서 Overlay 망의형태로제공되는것이아님. 음성, 데이터, 멀티미디어가 IP 중심으로전달되며, 각미디어특성에맞는 QoS 가제공됨. 개방형구조를가진플랫폼상에서구현됨. 이때, 전송과교환이분리된형태를가짐. 2002-7-24 Krnet 2002 125
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 Mobile Enterprise Customers Remote Office/SOHO Residential Users 2002-7-24 Krnet 2002 126
개방형네트워크기술표준화동향 MSF(Multiservice Switching Forum) Telcodia, Cisco, MCI WorldCom 등 14 개통신관련회사를주축으로 98 년 8 월에구성 ATM, Frame Relay, IP, 영상서비스등을모두하나의전송및교환인프라로수용할수있는방안에대해서 reference architecture 를제시하고구성기능및기능간의개방형인터페이스에대한정의 ISC(International Soft Switch Consortium) Voice over IP 를중심으로 Cisco, Lucent 등과같은대형통신업체와 Venture 기업까지포함한조직으로서 MGC(Call Agent) 의 architecture, 상호연동성등에대한활동 주로 Soft Switch 를개발하거나제품을가진업체들이중심이되어활동 2002-7-24 Krnet 2002 127
MSF VoP Architecture Applications Plane Service Service Agent Control Plane SS7 Signaling Gateway Call Agent Signalling Controller Proxy Signaling Call Agent Switching Plane Switching Fabric Adaptation Plane TGW TGW Voice TDM Voice TDM 2002-7-24 Krnet 2002 128
An NGN VoP Architecture Service Agent Call Connection Agent SCP IP Phones, PCs, H.323 Terminals STP SS7 Gateway Customer Gateway Access Network Access Gateway Core Packet Network Voice Streams Trunk Gateway NB Trunks LE 2002-7-24 Krnet 2002 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 15 여기관참여중 Cisco, HP, Lucent, Nortel, Telcordia 등 WG : Application, Architecture, Device Control, SIP, Session Mn. 활동사항 : IP 망을기반으로기존망과의연동기능에대한구조제시 VoIP 2002-7-24 Krnet 2002 130
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 2002-7-24 Krnet 2002 131
SoftSwitch Model Circuit Switch Model Services SoftSwitch Model Services Controller *GCP Soft Switch *GCP T D M Time Slot Interchange T D M T D M Packet Data Backbone T D M Line Card Trunk Card Media Gateway Media Gateway 2002-7-24 Krnet 2002 132
Classic Softswitch SS7 Network Local Carrier IMT SG SCP SG MGC Service Routing/Call Control/RM IMT Local Carrier MGCP/ MEGACO MG Managed Packet Network MG ISDN PRI MRS FS App Server ISDN PRI ISP B PBX Third Party Apps 2002-7-24 Krnet 2002 133
Softswitch 의연동 Softswitch1 Softswitch2 STP SS7 SoftSwitch BICC SIP-T SoftSwitch ISUP ANSI/ITU Trunk MG MEGACO/H.248 IP/ATM SIP Trunk MG Fax LE C DLC Access MG SIP LE C Fax 2002-7-24 Krnet 2002 134
sub mgmt messaging security billing network admin sub mgmt messaging security billing network admin sub mgmt messaging security billing network admin IP 기반통합서비스망구조 Today s Business Environment Next Generation Communications Networks intelligent sub mgmt unified message central security central billing network admin Internet / packet network (ATM / IP) wireless network wireline network Internet wireless wired data wireless access wireline access data access 서비스내용의중복 특정망에적용된컨텐츠 Content-driven environments Seamless, integrated service Access distinctions minimized wireless and wireline voice and data 2002-7-24 Krnet 2002 135
Service Intelligent IP Network Vision Application Layer Service Control Layer Network Layer 제 3 자응용 IP Service and Application Control QoS Authentication Security Multicast Metro Optical RAS DSL Cable Frame/ATM Wireless 응용 Hosting Aggregation Network 메시징응용 Web 기반서비스선택 Application Mediation Layer Policy Server LDAP based IP Service Switch Enterprise lp/mpls Core Soft Switch ASP Programmability 제 3 자응용 Web access Open APl's Multimedia Call Control VoIP and Video H.323/SIP MGCl (H.248) Media GW Internet PSTN 2002-7-24 Krnet 2002 136
NexGen Service Architecture: Service provider / ASP application servers Utility Services Communications Services Messaging Services Commerce Services Customer Profiles Directory Servers Internet Secure Intranets IP-based Transport and Signaling Content Services Entertainment Services Wireless voice/data IN/AIN/WIN PSTN Packet Core Wireless Data Wireline voice Wireless data Web appliances Dialup/broadband Packet voice 2002-7-24 Krnet 2002 137
VoIP 응용분야 2002-7-24 Krnet 2002 138
VoIP 응용분야의주요이슈 One Pipe access for all comm. Services Unified IP infrastructure VoDSL case xdsl VoDSL Wireless Internet Case GSM+GPRS IMT-2000 ALL IP Network 2002-7-24 Krnet 2002 139
고속인터넷접속서비스 IAS: Internet Access Server DSLAM: DSL Access Multiplexer Modem IAS PSTN/ISDN Data Network POP POP ISP-1 ISP-2 DSLAM POP ISP-3 ADSL Modem 음성과데이터의결합 - 접속망은 Hybrid - 핵심망은분리 2002-7-24 Krnet 2002 140
VoDSL (Voice over DSL) 음성과데이터를다중화하여, ATM over xdsl의형식으로전송 복수의음성채널을하나의 AAL2 VC로다중화함. 데이터채널은 AAL5 VC로전송 Power Spectrum POTS Voice Data AAL2 AAL5 상향 하향 Freq ATM xdsl 100 khz ADSL frequency usage 1 MHz Frequency Non-voiceband frequency 2002-7-24 Krnet 2002 141
VoDSL (Voice over DSL) 연결 Transmitted by AAL2(I.366.2) V - IF Subscriber Switch VoDSL CLAD LAN Router IAD VC for Voice DSLAM ATM-SW ISP VC for IP VC for Voice and VC for IP are separated by ATM Switch 2002-7-24 Krnet 2002 142
제 2 단계 VoDSL ATM - SW Media GW Controller LAN Router IAD DSLAM Media Gateway PSTN Switch ISP Access Gateway 가 Media GW 와 MGC 로대체됨. MGC 에서교환기능을수행함. 2002-7-24 Krnet 2002 143
VoDSL Solution Characteristics VoDSL Solution Characteristics 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) Based on ATM, not on IP Voice packetization based on native ATM using AAL2 Why ATM? ATM is there: most DSL is ATM-based 대역폭이효율성이우수함. QoS 가제공되는것이입증되었음. No security concerns (using PVCs) Almost no local service providers looking for a VoIP solution 2002-7-24 Krnet 2002 144
Current VoDSL Network IAD xdsl Voice Gateway GR 303 Class 5 Switch PSTN DSLAM Packet Backbone IAD xdsl Broadband Access Server ISP1 ISP2 IAD = Integrated Access Device DSLAM = DSL Access Multiplexer 2002-7-24 Krnet 2002 145
GSM Architecture GMSC PSTN BTS A-bis BSC A MSC MAP GPRS Register H/VLR HLR/AUC GPRS Register 2002-7-24 Krnet 2002 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 2002-7-24 Krnet 2002 147
무선망에서의인터넷접속구조 (GPRS) BTS A-bis BSC PCU MSC : Mobile Switching Center GMSC : Gateway MSC PCU : Packet Control Unit SGSN: serving GPRS support Node GGSN: gateway GPRS support Node Circuit Switched Traffic Gb Packet Traffic A GMSC MSC Peer Elements SGSN Gn GGSN MAP MAP Gi GPRS Register H/VLR PSTN HLR/AUC GPRS Register Public Switched Packet Network 2002-7-24 Krnet 2002 148
GSM/GPRS Network Configuration BTS BTS BSC MSC/ VLR MAP HLR PSTN BTS BSC Intranet SGSN GGSN BTS BSC Internet Radio Link TCP/IP connection GTP Tunnel.GPRS is a IP packet network for P-to-P packet data transportation 2002-7-24 Krnet 2002 149
VoIP over GPRS codec RTP UDP IP End-to-end IP BSS SGSN IP GGSN codec RTP UDP IP GW 회선교환 network codec RTP UDP IP Protocol stack in the MS Public IP codec RTP UDP IP 2002-7-24 Krnet 2002 150
VoIPoGPRS : 이용자평면 CODEC RTP UDP IP SNDCP Relay SNDCP GTP IP GTP LLC RLC MAC GSM RF Relay RLC BSSGP MAC GSM RF Network Service L1bis LLC BSSGP Network Service L1bis UDP / TCP Um Gb Gn Gi MS BSS SGSN GGSN IP L2 L1 UDP / TCP IP L2 L1 2002-7-24 Krnet 2002 151
VoIPoGPRS 의개념정리 VoIPoGPRS 은 GSM voice services 을대체하는것은아님. No bandwidth gain on the radio interface Lower speech quality (lower Codec quality, delay 와 loss 증가 ) 단말기가변경되어야하고, 게이트웨이및 Call Agent 가요구됨. VoIPoGPRS 는다음의기능을제공함. click-and-talk, browse and phone 등과같은멀티미디어서비스와결합기능제공 Particular long-distance tariff distortions Integrated end-to-end multi-media 통신으로의진화경로 실시간패킷음성전달에표준의개발이요구됨. QoS renegotiation or multiple QoS per PDP-Context Mobile constrained Coding-Scheme New SNDCP features such as RTP/UDP/IP compression 2002-7-24 Krnet 2002 152
조각 1 조각 1 2 분기 2 분기 3 분기 3 분기 4 분기 4 분기 IMT-2000 의표준환경 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) Flexible connection between Radio modules & Core Networks based on operator needs Core Networks Evolved GSM (MAP) Evolved ANSI-41 IP-based Networks Inter-Network Roaming Network-to-Network Interfaces 2002-7-24 Krnet 2002 153
IMT-2000 의 All IP Network ALL IP 망 IMT-2000 이동통신망을 IP기술로구성하여 패킷망을기반으로하여 실시간, 비실시간데이터를동시에서비스할수있도록함. 패킷음성서비스, 멀티미디어서비스제공 All IP network 의특성 이동통신망과인터넷의결합 IP protocols 을사용하여패킷전송 IP 기반의호처리프로토콜사용 IP 수용하는단말사용 실시간 IP 기능을가진 radio access network 및 Cable 과 HIPERLAN 을포함한다른접속방식을지원함. 2002-7-24 Krnet 2002 154
ALL IP Network 의목표 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 2002-7-24 Krnet 2002 155
IP Transport ALL IP 망에서의주요이슈 GPRS(3GPP) 및 PPP-MIP(3GPP2) 에기반한망에서진화함에따른망의구조를도출하고, 망기능요소를정의하며이들요소들사이에인터페이스를설계 무선접속망에 IP 의도입 (IP BTS) 전송기능과제어기능을분리하여독립적으로구현 이동성관리 음성및실시간서비스에패킷기술을도입함으로써, Mobile IP 와같은이동성관리기능의성능개선 핸드오버의성능개선 QoS end-to-end QoS 를위한백본망의개선 QoS 구조로서 RSVP 나 DiffServ 등의도입방안 2002-7-24 Krnet 2002 156
VoIP ALL IP 망에서의주요이슈 VoIP 의기능을무선통신망에도입 (multimedia domain) 새로운호제어, 미디어제어기능을핵심망의요소로포함 신속한서비스의도입 개방형구조를취함으로서신속한서비스의생성및진화가가능토록함. 2002-7-24 Krnet 2002 157
3GPP All IP 의개념 Release 99 Architecture - Hybrid 회선모드단말 Iu-cs Circuit domain PSTN/ISDN 패킷모드단말 UTRAN Iu-ps packet domain Internet hybrid circuit/packet architecture 2 separate domains - circuit and packet Pure All IP Architecture 회선모드단말 packet mode terminal UTRAN / GERAN Iu-ps packet domain long term vision call control H.323 or SIP no support for circuit mode terminals GW PSTN/ISDN Internet 2002-7-24 Krnet 2002 158
3GPP All IP 의개념 (2) Pure All IP Architecture 회선모드단말 UTRAN / GERAN Iu-ps packet domain GW PSTN/ISDN Internet packet mode terminal Pure All IP CS option Architecture 회선모드단말 UTRAN / GERAN GW Iu-ps packet domain GW PSTN/ISDN Internet packet mode terminal 2002-7-24 Krnet 2002 159
3GPP All IP Network Architecture R Alternative Access Network Uu CAP TE MT ERAN R Um Iu-ps' Iu TE MT UTRAN Applications & Services SCP Gr Gp HSS SGSN Mh Gn Gc Cx Gf Gi EIR Legacy mobile signaling Network R-SGW Ms CSCF Mr MRF Gi GGSN CSCF Mw Mm Mg Gi MGCF Mc Gi MGW T-SGW Multimedia IP Networks PSTN/ Legacy/External Gn SGSN GGSN Other PLMN GPRS Based Signalling Interface Signalling and Data Transfer Interface 2002-7-24 Krnet 2002 160
New Functional Elements Call State Control Function (CSCF) Call Agent Home Subscriber Server (HSS) HLR Transport Signaling Gateway Function (T-SGW) 유선망과신호연동 Roaming Signalling Gateway Function (R-SGW) Media Gateway Control Function (MGCF) Media Gateway Function (MGW) Multimedia Resource Function (MRF) Intelligent Peripheral MSC Server(for Option 2) Gateway MSC Server(for Option 2 2002-7-24 Krnet 2002 161