voice over ip 與 ip telephony 簡介 資策會 網路及通訊實驗室 conference over ip team...
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Voice over IP Voice over IP 與 與 IP Telephony IP Telephony 簡介簡介
資策會 網路及通訊實驗室資策會 網路及通訊實驗室Conference over IP TeamConference over IP Team
楊政遠 博士楊政遠 博士[email protected]
2003/07/26
Review - PSTNReview - PSTN• PSTN(Public Switch Telephone Network)
– Signaling: System Signal No: 7 (SS7)
– Carrier: T1 主幹 and successors …...
STP
Local loop
DTMF
Signaling plane
Bearer plane
客戶端 (CPE)
局端 (CO)
Review - Voice ConferenceReview - Voice Conference
• Basic issues of voice conference setup
1. ?
3. Digital voice packets
2. Conference setup
AD/DA compress/decompress AD/DA
compress/decompress
phonebookserver
4. Conference terminate
Review - basic issuesReview - basic issues
• Telephony Issues (PSTN v.s. VoIP)– Signaling
• Addressing / Control– PSTN - SS7 (ITU E.164)– VoIP - H.323 、 SIP 、 MGCP 、 Megaco/H.248
• Capability exchange– PSTN - Analog voice / -law 、 A-law PCM– VoIP - Digital voice / G.711 、 G.723.1 、 G.729
Review - basic issuesReview - basic issues
• Telephony Issues (PSTN v.s. VoIP)– Bearer
• Transport– PSTN - TDM (Time Division Modulation) Trunk – VoIP - RTP over UDP/IP
• Delay and Jitter– PSTN - circuit switching / propagation delay– VoIP - packet switching / unbounded delay and jitter
• Internetworking between the existent PSTN, GSM/GPRS and future 3G all IP network.
Review - short conclusionReview - short conclusion
• Signaling– Addressing: find call party– Call control: control the call progress – Capabilities exchange: negotiate the media
types of this call
• Media transport (bearer)– media processing– media transmission
Media TransportMedia TransportMedia ProcessingMedia ProcessingMedia Transmission
Process of digital voice transmissionProcess of digital voice transmission
Low-pass filterSampling &A/D convert
Silentdetection Compression
RTP packetencapsulation
RTP packetdecapsulation
DecompressionTiming reconstruct
D/A convert
Internet
VoIP Endpoint FunctionalityVoIP Endpoint Functionality
AD/DA converter
DSP coding
Buffering andpacketization
Jitter buffer
TCP/IPprotocol stack
Network interface
PCM
copper wire
IP
framesPhone
interface
Digital SignalProcessing
Packet handling
Digitalization SpeechDigitalization Speech
• Digitalization Speech– Low Pass Filter (LPF) 300 Hz ~ 3000 Hz– Sampling and Quantization
VoIP VoIP Endpoint FunctionalityEndpoint Functionality
AD/DA converter
DSP coding
Buffering andpacketization
Jitter buffer
TCP/IPprotocol stack
Network interface
PCM
copper wire
IP
framesPhone
interface
Digital SignalProcessing
Packet handling
Digitalization SpeechDigitalization Speech
• PCM (Pulse Code Modulation)– digital quantization introduces distortions
Digitalization SpeechDigitalization Speech
• main speech coding techniques – waveform codec, source codec and hybrid codec
VoIP VoIP Endpoint FunctionalityEndpoint Functionality
AD/DA converter
DSP coding
Buffering andpacketization
Jitter buffer
TCP/IPprotocol stack
Network interface
PCM
copper wire
IP
framesPhone
interface
Digital SignalProcessing
Packet handling
R,G,B bitmap
Y,Cb,Crmatrix
ColorTransform
frequencymatrix Quantizer
DiscreteCosine
Transform
Huffmanencoder
RTP packetencapsulation
Quantizationtable
Huffmantable
Media TransportMedia TransportMedia ProcessingMedia Transmission Media Transmission
Voice Quality of ServiceVoice Quality of Service
• Interactive Voice QoS factors– Packet lost– Delay– Jitter
Voice QoS - Packet LostVoice QoS - Packet Lost
0%5%
10%15%20%25%30%35%40%45%
Per
fect
Exc
elle
nt
Go
od
Acc
epta
ble
An
no
yin
g
Bad
Un
usa
ble
G.711
G.723.1InternetIntranet
• Minimize one-way delay, keep it below 150ms ITU G.114 states one-way delay <= 150 msec ~200 msec is acceptable
GPRS BackboneIP Network
IP based network
variable delay 20~300 or more ms
Fixed delay 1. Framing: 20~30 ms 2. Processing: 15 ms 3. Transmission: 10ms 4. Decompress/buffer: 25 ms
Variable delay 1. Buffer: 5~20 ms 2. Network: 20 ~ ? ms
Framing (algorithm): 20 ~ 30 msCompress (H/W DSP): 5 msProcessing (packetize): 10 ms
Receiving buffer: 20 msDecompress delay: 5 ms
Voice QoS - DelayVoice QoS - Delay
• Codec algorithm delay ( Ex. G.729 )– serialize the frame ( 10 ms)– look ahead (5 ms)
total algorithm delay = 15 ms
Frame
next sample Sampling & A/D converter
8000 Hz
Voice QoS - DelayVoice QoS - Delay
0100200300400500600700800900
Per
fect
Exc
elle
nt
Go
od
Acc
epta
ble
An
noy
ing
Bad
mill
isec
ond
s
G.711Intranet
Internet
Voice QoS - DelayVoice QoS - Delay
Voice QoS - JitterVoice QoS - Jitter
Jitter (Delay Variation)Jitter (Delay Variation)
0
100
200
300
400
500
600
Per
fect
Exc
elle
nt
Go
od
Acc
epta
ble
An
no
yin
g
Bad
Un
usa
ble
mill
isec
on
ds
G.711Intranet
Internet
Packet Handling LatencyPacket Handling Latency
• Jitter– variability in the arrival rate of data is called jitter
H i H o w are yo u
H i H o ...w a re yo u
J itte r
sender
rece ive
• Jitter bufferH i H o w are yo u
H i H o ...w a re yo u
J itte r
sender
rece ive
H i H o w are yo u< 1 5 0 ~2 0 0 m s
J itte r b u ffe r/ S m o o th e r
p layback
Voice QoS - JitterVoice QoS - Jitter
DefinitionsDefinitions
• Voice over IP (VoIP)– Voice over Internet Protocol
• voice packet over well controlled IP network !
– does not imply Voice over Internet
• IP Telephony– Telephony system based on Internet Protocol– Inter-operabilities
• standards• compatibility
Voice packets transmissionVoice packets transmission
• TCP(reliable) or UDP(unreliable) ?– The characteristics of interactive voice/video
• on-the fly (realtime)– retransmission is none-sense
• human physiology– tolerate few information lost independently
• isochronal– timing information snapshot and re-construct
– media frame encapsulated in RTP/UDP/IP
IP header(20 bytes)
UDP header(8 bytes)
RTP header(12 bytes)
media payload
RTP: A Transport Protocol for Real-Time RTP: A Transport Protocol for Real-Time Applications (RFC 1889) Applications (RFC 1889)
http://www.ietf.org/html.charters/avt-charter.htmlhttp://www.ietf.org/html.charters/avt-charter.html
• The simplest RTP fixed header
RTP (RFC1889)RTP (RFC1889)
IP header UDP header RTP header RTP payload
Fields of RTP HeaderFields of RTP Header
• V (version):– RFC 1889 RTP version 2, V=2
• P (padding): – padding bytes ?
• X (extension):– RTP header extension ?
• CC (count of contributor):– number of media contributors (for mixer)
• M (marker):– media specified
• audio: the begin of talk spurt• video: begin of end of video frame
• PT (payload type):– Defined by RFC 1990
Fields of RTP HeaderFields of RTP Header• Sequence number:
– increment by one– initial value is random
• Timestamp:– reflect the sampling instant of the 1st data bytes– format depends on application– initial value is random, increments monotonically
• Sync SRC:– synchronization source ID– random choice– RTP session global uniquely
RTP Header profile (RFC1900)RTP Header profile (RFC1900)
PT encoding nameaudio/video
(A/V)clock rate
(Hz)Channels(audio)
0 PCMU A 8000 11 1016 A 8000 12 G721 A 8000 13 GSM A 8000 15 DVI4 A 8000 16 DVI4 A 16000 17 LPC A 8000 18 PCMA A 8000 19 G722 A 8000 1
10 L16 A 44100 211 L16 A 44100 115 G728 A 8000 125 CelB V 9000026 JPEG V 9000028 nv V 9000031 H261 V 9000032 MPV V 90000
SignalingSignaling AddressingAddressing Call control Call control Capabilities Capabilities exchangeexchange
ReviewReview
• The milestone of Voice over IP– the 1st experiment of voice packet over IP
• 1974 Network Voice Protocol (RFC741)
– the 1st commercial Internet telephony AP, Windows 3.1• Vocaltec, 1995
– the 1st version of H.323• ITU, 1996
– the 1st widely deployed H.323 AP • Microsoft NetMeeting, May, 1996
– the 1st commerical Internet Telephony Service• Delta Three, 1996
VoIP signaling protocol standardVoIP signaling protocol standard
• ITU-T H.323– http://www.itu.int/rec/recommendation.asp?
type=folders&lang=e&parent=T-REC-H.323
• IETF MGCP – RFC2705
• IETF SIP– RFC3261– http://www.ietf.org/html.charters/sip-charter.html
• IETF/ITU-T Megaco/H.248– RFC3015
Session Initiation ProtocolSession Initiation Protocol
• SIP Architecture– RFC3261
SIP UserAgent
SIP UserAgent
SIP UserAgent
RegistrarProxyServer
RedirectServer
SIP Server
INVITE SIP:[email protected] SIP/2.0…….
180, Ringing
200, OK
ACKRTP (voice)
BYE
ACK
Caller CalleePickup & dial
ringing
pick up
on-hook
SIP BASIC Call flow
ringback
VoIP protocol standard - SIPVoIP protocol standard - SIP
Request MethodsRequest Methods
INVITE The user is begin invited to participate in a session.
ACK The client has received a final response to an INVITE.
OPTIONS The server is begin queried as to its capabilities.
BYE The user wishes to release the call.
CANCEL It cancels a pending request (not completed request).
REGISTER It conveys the user’s location information to a SIP server.
Response Status LineResponse Status Line
• SIP-Version SP Status-Code SP Reason-Phrase CRLF– Status-Code =
– SIP/2.0 SP 180 SP Ringing CRLF
1xx Informational
2xx Success
3xx Redirection
4xx Client-Error
5xx Server-Error
6xx Global-Failure
SIP Request ExampleSIP Request Example
INVITE sip:[email protected] SIP/2.0 Method type, request URL and SIP version
Call-ID:[email protected] Globally unique ID for this call
Content-type:application/sdp The body type, an SDP message
Cseq:1 INVITE Command Sequence number and type
From:sip:[email protected] User originating the request
To:sip:[email protected] User being invited into the call
Via:SIP/2.0/UDP 140.92.61.55:5060 IP Address and port of previous hop
Blank line separates header from body
v=0 SDP version
o=smayer 280932498 280932498 IN IP4 140.92.62.105
Owner/creator and session identifier
s=sip session The name of session
p=+886-2-25643588 Phone number of caller
c=IN IP4 140.92.61.105 Connection information
t=0 0 Time the session is active
m=audio 49170/1 RTP/AVP 1 media name and transport
SIP RegistrationSIP Registration
SIP Registrar(domain: iptel.org)
Location Server
jiri@
195.3
7.7
8.1
73
REGISTER sip:iptel.org SIP/2.0
From:sip:[email protected]
To:sip:[email protected]
Contact:<sip:195.37.78.173>
Expires:3600
SIP/2.0 200 OK
SIP Operation in Proxy ModeSIP Operation in Proxy Mode
SIP Proxy Server
Location Server
7.7
8.1
73
INVITE
From:sip:[email protected]
To:sip:[email protected]
Call-ID:[email protected]
SIP/2.0 200 OK
jiri
?
SIP/2.0 200 OK
INVITE sip:[email protected]
From:sip:[email protected]
To:sip:[email protected]
Call-ID:[email protected]
ACK sip :[email protected]
[email protected]@195.37.78.1
73
SIP Operation in Redirect ModeSIP Operation in Redirect Mode
SIP RedirectServer
Location Server
Calle
e@
hom
e.co
m
Calle
e
?
INVITE
302 Moved TemporarilyContact: [email protected]
ACK sip:[email protected] sip:[email protected]
SIP/2.0 200 OK
ACK sip:[email protected]
RTCPRTP
SIP, H.323 and MGCPSIP, H.323 and MGCP
IP
MGCP
Call Control and Signaling Signaling and Gateway Control
Media
H.225
Q.931
H.323
H.323 Version 1 and 2 supports H.245 over TCP, Q.931 over TCP and RAS over UDP.
H.323 Version 3 and 4 supports H.245 over UDP/TCP and Q.931 over UDP/TCP and RAS over UDP.
SIP supports TCP and UDP.
TCP
RAS
UDP
SIPH.245
Audio/Video
RTSP
Protocol wars - Protocol wars - Viewpoint from CISCOViewpoint from CISCO
Projected Port (DS0) Protocol Transition Rates
Q1CY99 Q1CY00 Q1CY01 Q1CY02 Q1CY03 Q1CY04
Calendar Quarters
% P
ort
Un
it S
ales
MGCP / H.248 - DS0s
SIP - DS0s
H.323 - DS0s
MixedH.323 & SIP
20%
40%
60%
80%
100%
Next Generation Converged Network Next Generation Converged Network andand
IP Telephony systemIP Telephony system
0
1
2
3
4
5
6
7
8
9
10
1997 1998 1999 2000 2001 2002 2003 Year
Rel
ativ
e tr
affi
c
Total
Data
Telephony
Siemens
• Telecommunication deregulation– Investment reward : Data network > voice network– Cost - single network architecture– Cost - open standards / short time-to-market
• Open VoIP and supplemental standards– H.323 、 MGCP 、 Megaco/H.248 、 SIP
• Bandwidth is no more a critical issue– DWDM , xDSL / cable , Fast/Giga Ethernet
• Quality of Service guarantee
Next Generation Converged NetworkNext Generation Converged Network
Next Generation Converged NetworkNext Generation Converged Network
T ru n kg a te w a y
PSTN
M e d iag a te w a y
c o n tro llo r
PSTNInternet
T ru n kg a te w a y
S ig n a lin gg a te w a y
M G C PM E G A C O /H .2 4 8
STP
SCP
SSPSSP
S ig n a lin gg a te w a y
M e d iag a te w a y
c o n tro llo r
STP
SCP
PO TS
PO TS
M G C P /S IPp h o n e
S o ftsw itc h
softsw itchsoftsw itch
ana logyphone se t
R e s id e n tia lg a te w a y
S IP ¡B M G C P ¡BM E G A C O /H .2 4 8
M G C P ¡BM E G A C O /H .2 4 8
S IP -T
S IP -T
S IP -T
Next Generation Converged NetworkNext Generation Converged Network
• Residential Gateway / Integrated Access Device
Call Control and Switching
Operation System Support
Feature and Application Creation
• IP Telephony System must support
IP Telephony SystemIP Telephony System
SIP based IP Telephony SystemSIP based IP Telephony System
CDR Server(s)
Feature Server(s)
Provisioning Server(s)
3rd Party Billing System
RADIUS
SNMP NetworkManager
ClearingHouse
Internet
SIP proxy ServerSIP proxy Server
PSTN
Gateway
SIP proxy Server
SIP IP Phone MGCP Device
MGCP/SIPTranslator
SIP proxy Server
H.323/SIP Translator
SIP proxy Server
H.323 Terminal
SIP based
VOCAL System [http://www.vovida.org/]
H.323 Translator: Acts as a Gatekeeper to control H.323 endpoints.Talks SIP to the rest of the network for routing and features.
SIP based IP Telephony SystemSIP based IP Telephony System
MGCP Translator: Acts as a call agent to control MGCP end points. Talks SIP to the rest of the network for routing and features.
SIP based IP Telephony SystemSIP based IP Telephony System
SIP based IP Telephony SystemSIP based IP Telephony SystemSIP proxy Server: Acts as a trusted boundary for calls entering or leaving a network. Provides authentication and collects billing information for the CDR server.
CDR Server: Collects billing information from Marshal Servers and interfaces with billing systems using the RADIUS accounting protocol.
SIP based IP Telephony SystemSIP based IP Telephony System
Provisioning Server: Used to provision, configure and manage subscribers and servers from a GUI.
SIP based IP Telephony SystemSIP based IP Telephony System
Feature Server: Provide CPL based or XML scripts that run basic telephony features.
SIP based IP Telephony SystemSIP based IP Telephony System
VoIP Feature ServicesVoIP Feature Services• Feature services are the value-added functions of the
phone system– Core features
• Calling Information– Calling Number Delivery (CND) or Calling Line Identification (CLID) /
Calling Party Identity Blocking (CIDB)• Calling Forwarding
– Forward All Calls (CFA) / Forward - No Answer Mode (CFNA) / Forward - Busy Mode ( CFB )
• Call Blocking / Call Screening– Set features
• Call transfer / Call Return / Call waiting / Cancel Call Waiting ( CCW )– Scriptable features
• Call Processing Language (CPL)
IP Telephony - SoftswitchIP Telephony - Softswitch
Softswitch
Cellular Station
Media GatewaysIAD with DSL/Cable Modem
Digital Cross Connect
SS7 Gateway
SS7
Application Servers
Q.931/Q.2931
CPL
SIP
MGCP
MEGACO
SIPH.323
MGCP
33GPP Network ModelGPP Network Model
Endpoints with voice driving Endpoints with voice driving converged IP infrastructureconverged IP infrastructure
VideoTelephony
VoicePortals
PC toPhone
IP PhonesPDA
UnifiedMessaging Voice-enabled
Websites
InstantMessenger
Voice Service FocusVoice Service Focus
PSTN
IPSec orMPLS
SOHO
Internet
SS7
BranchOffice
HQ
Ent/SMB B
Messaging,ACD, IVR
HQBranch Office
Soft Switches
Enterprise A
Enterprise B
Ent/SMB AIOS Telephony
Services
CallManager
1. 1. Managed IP Managed IP TelephonyTelephony
3. 3. IP Centrex and IP Centrex and Hosted AppsHosted Apps
2. 2. Voice-Enabled Voice-Enabled Data VPNData VPN
4. 4. Integrated Integrated AccessAccess
V
V V
All IP NetworkAll IP Network
3G/4G Wireless Coverage
Home WLAN
Restaurant WLAN
Office LAN Hotel WLAN/LAN
Airport WLAN
LAN, WLAN hot spots LAN, WLAN hot spots
and 3G/4G wireless mobilityand 3G/4G wireless mobility
Wireless LAN Voice MobilityWireless LAN Voice Mobility
The Big Technical Challenge: The Big Technical Challenge: 802.11 VoIP Mobility802.11 VoIP Mobility
• Two Types of mobility:– Macro Mobility is the change of domain/administration
• Between “hotspots”
• Between Cellular (wide area) and WLAN (local area)
– Micro Mobility is the change of sub-net attachment (Campus, Enterprise)
Hotspot A(AP AP
Hotspot B
AP AP
Internet
Micro-Mobility Macro-Mobility Micro-Mobility
Call Control an Mobility ProtocolsCall Control an Mobility Protocols
• Two protocol approaches to support mobility– Support mobility at Network Layer: Mobile IP– Support Mobility at the Application Layer: SIP– H.323 is not expected to play a significant role in VoIP mobility
• SIP is widely supported in PC market and applications– Microsoft has included SIP as part of Windows XP release– Sip Handles Proxy server, NAT and Firewall issues– Ideal For HOME/SOHO/Consumer Market
• Mobil IP is desired but requires significant infrastructure investment
An Example: Loosely Coupled An Example: Loosely Coupled Cellular GPRS-WLAN IntegrationCellular GPRS-WLAN Integration
Operators IP
Network
Internet
HLR - AuC
CG
Dual ModeMN
SGSN GGSN
Billing Mediator
GPRS CORE
APBSS-1
WLAN Network
APBSS-2
APBSS-N
GPRS/UTRANNetwork
AP: WLAN Access PointBSS: Basic Service SetCG: Charging GatewayHLR: Home location registerAuC: Authorization centerSGSN: Serving GPRS support nodeGGSN: Gateway GPRS support nodeCAG: Cellular access gatewayFA: Foreign AgentHA: Home Agent
Billing System
FA/AAA
HA
CAG
SIP Roaming SupportSIP Roaming Support
• Logging into different IP networks away from home
• Basic Steps:1.Get an IP address
• Use DHCP
2.Register with local proxy• For firewall transversal for UDP
3.Register with home Registrar• For calls routing
SIP Roaming SupportSIP Roaming Support
Home.comVisit.com
From:[email protected]:166.1.2.3 From:[email protected]
Contact:[email protected]
INVITE
INVITEINVITE
• Remote registration
From:[email protected]:166.4.5.6
Move
SIP Roaming SupportSIP Roaming Support
• Precall mobilityHome.comDHCP
IP Address INVITE
INVITE
OK
302 moved temporarily
ACK
ACK
MEDIA
SIP Roaming SupportSIP Roaming Support
• Midcall mobility
INVITE
OK
ACK
MEDIA
MEDIA
中斷 ?