Transcript 新世代網際電話標準與架構剖析

Voice over IP 與 IP Telephony 簡介
資策會 網路及通訊實驗室
Conference over IP Team
楊政遠 博士
[email protected]
2003/07/26
Review - PSTN
• PSTN(Public Switch Telephone Network)
– Signaling: System Signal No: 7 (SS7)
– Carrier: T1 主幹 and successors …...
Signaling plane
STP
局端 (CO)
Bearer plane
客戶端(CPE)
Local loop
DTMF
Review - Voice Conference
• Basic issues of voice conference setup
phonebook
server
1. ?
2. Conference setup
AD/DA
compress/decompress
3. Digital voice packets
4. Conference terminate
AD/DA
compress/decompress
Review - 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 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 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 Transport
Media Processing
Media Transmission
Process of digital voice transmission
Low-pass filter
Sampling &
A/D convert
Silent
detection
Compression
RTP packet
encapsulation
Internet
D/A convert
Timing
reconstruct
Decompression
RTP packet
decapsulation
VoIP Endpoint Functionality
Phone
interface
PCM
DSP coding
Digital Signal
Processing
AD/DA
converter
copper wire
frames
Buffering and
packetization
Jitter buffer
Packet
TCP/IP
protocol stack handling
Network
interface
IP
Digitalization Speech
• Digitalization Speech
– Low Pass Filter (LPF) 300 Hz ~ 3000 Hz
– Sampling and Quantization
Digitalization Speech
• PCM (Pulse Code Modulation)
– digital quantization introduces distortions
Digitalization Speech
• main speech coding techniques
– waveform codec, source codec and hybrid codec
R,G,B
bitmap
Color
Transform
Y,Cb,Cr
matrix
Discrete
Cosine
Transform
RTP packet
encapsulation
Huffman
table
Huffman
encoder
frequency
matrix
Quantizer
Quantization
table
Media Transport
Media Processing
Media Transmission
Voice Quality of Service
• Interactive Voice QoS factors
– Packet lost
– Delay
– Jitter
Voice QoS - Packet Lost
G.711
G.723.1
Unusable
Bad
Annoying
Internet
Good
Excellent
Perfect
Intranet
Acceptable
45%
40%
35%
30%
25%
20%
15%
10%
5%
0%
Voice QoS - Delay
• Minimize one-way delay, keep it below 150ms
ITU G.114 states one-way delay <= 150 msec ~200 msec is acceptable
Framing (algorithm): 20 ~ 30 ms
Compress (H/W DSP): 5 ms
Processing (packetize): 10 ms
IP based network
GPRS Backbone
IP 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
Receiving buffer: 20 ms
Decompress delay: 5 ms
Voice QoS - Delay
• Codec algorithm delay ( Ex. G.729 )
– serialize the frame ( 10 ms)
– look ahead (5 ms)
total algorithm delay = 15 ms
next sample
Sampling &
A/D converter
8000 Hz
Frame
900
800
700
600
500
400
300
200
100
0
Internet
Intranet
Bad
Annoying
Acceptable
Good
Excellent
G.711
Perfect
milliseconds
Voice QoS - Delay
Voice QoS - Jitter
Jitter (Delay Variation)
600
Internet
400
Intranet
G.711
300
200
100
Unusable
Bad
Annoying
Acceptable
Good
Excellent
0
Perfect
milliseconds
500
Packet Handling Latency
• Jitter
– variability in the arrival rate of data is called jitter
Hi
How
are
you
sender
Jitter
receive
Hi
Ho ...w are you
Voice QoS - Jitter
• Jitter buffer
Hi
How
are
you
sender
Jitter
receive
Jitter buffer
/ Smoother
Hi
Ho ...w are you
playback
< 150 ~
200 ms
Hi
How
are
you
Definitions
• 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 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 RTP header
(8 bytes)
(12 bytes)
media payload
RTP: A Transport Protocol for Real-Time
Applications (RFC 1889)
http://www.ietf.org/html.charters/avt-charter.html
RTP (RFC1889)
• The simplest RTP fixed header
IP header UDP header RTP header
RTP payload
Fields 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 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)
PT
encoding name
0
1
2
3
5
6
7
8
9
10
11
15
25
26
28
31
32
PCMU
1016
G721
GSM
DVI4
DVI4
LPC
PCMA
G722
L16
L16
G728
CelB
JPEG
nv
H261
MPV
audio/video
(A/V)
A
A
A
A
A
A
A
A
A
A
A
A
V
V
V
V
V
clock rate
(Hz)
8000
8000
8000
8000
8000
16000
8000
8000
8000
44100
44100
8000
90000
90000
90000
90000
90000
Channels
(audio)
1
1
1
1
1
1
1
1
1
2
1
1
Signaling
Addressing
Call control
Capabilities exchange
Review
• 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 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 Protocol
• SIP Architecture
– RFC3261
SIP User
Agent
SIP User
Agent
SIP Server
SIP User
Agent
Registrar
Proxy
Server
Redirect
Server
VoIP protocol standard - SIP
SIP BASIC Call flow
Caller
Pickup & dial
Callee
INVITE SIP:[email protected] SIP/2.0
…….
ringback
ringing
180, Ringing
pick up
200, OK
ACK
RTP (voice)
on-hook
BYE
ACK
Request Methods
INVITE
ACK
The user is begin invited to participate in a session.
The client has received a final response to an INVITE.
OPTIONS The server is begin queried as to its capabilities.
BYE
CANCEL
The user wishes to release the call.
It cancels a pending request (not completed request).
REGISTER It conveys the user’s location information to a SIP server.
Response Status Line
• SIP-Version SP Status-Code SP ReasonPhrase CRLF
– Status-Code =
1xx
Informational
2xx
Success
3xx
Redirection
4xx
Client-Error
5xx
Server-Error
6xx
Global-Failure
– SIP/2.0 SP 180 SP Ringing CRLF
SIP 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 Registration
[email protected]
Location Server
SIP Registrar
(domain: iptel.org)
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 Mode
Location Server
[email protected]
INVITE sip:[email protected]
From:sip:[email protected]
To:sip:[email protected]
From:sip:[email protected]
To:sip:[email protected]
Call-ID:[email protected]
Call-ID:[email protected]
SIP/2.0 200 OK
INVITE sip:[email protected]
SIP Proxy Server
SIP/2.0 200 OK
ACK sip :[email protected]
[email protected]
[email protected]
SIP Operation in Redirect Mode
Location Server
[email protected]
302 Moved Temporarily
Contact: [email protected]
[email protected]
INVITE sip:[email protected]
SIP Redirect
Server
ACK sip:[email protected]
INVITE sip:[email protected]
SIP/2.0 200 OK
ACK sip:[email protected]
[email protected]
SIP, H.323 and MGCP
Call Control and Signaling
Signaling and
Gateway Control
Media
Audio/
Video
H.323
H.225
H.245
Q.931
RAS
SIP
MGCP
TCP
RTP
RTCP
RTSP
UDP
IP
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.
Protocol wars - Viewpoint from CISCO
Projected Port (DS0) Protocol Transition Rates
% Port Unit Sales
100%
Q1CY99
80%
60%
40%
MGCP / H.248 - DS0s
SIP
- DS0s
H.323
- DS0s
Q1CY00
Q1CY01
Mixed
H.323
& SIP
20%
Q1CY02
Calendar Quarters
Q1CY03
Q1CY04
Next Generation Converged Network
and
IP Telephony system
10
Total
9
Data
8
Telephony
Relative traffic
7
6
5
4
3
2
1
0
1997
1998
1999
2000
2001
2002
2003
Year
Siemens
Next Generation Converged Network
• 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 Network
softswitch
Signaling
gateway
Media
gateway
controllor
SIP-T
SIP-T
MGCP
MEGACO/H.248
SCP
Softswitch
SIPЎBMGCPЎB
MEGACO/H.248
Signaling
gateway
MGCPЎB
MEGACO/H.248
MGCP/SIP
phone
Trunk
gateway
PSTN
SSP
softswitch
SIP-T
Trunk
gateway
STP
Media
gateway
controllor
Internet
Residential
gateway
SCP
STP
PSTN
SSP
POTS
POTS
analogy
phone set
Next Generation Converged Network
• Residential Gateway / Integrated Access
Device
IP Telephony System
• IP Telephony System must support
Feature and
Application Creation
Operation
System Support
Call Control and Switching
SIP based IP Telephony System
PSTN
Internet
Gateway
SIP proxy Server
SIP proxy Server
Provisioning
Server(s)
SNMP
Network
Manager
Feature
Server(s)
SIP based
CDR
Server(s)
RADIUS
H.323/SIP
Translator
SIP proxy Server
H.323
Terminal
Clearing
House
MGCP/SIP
Translator
SIP proxy Server
SIP IP Phone
SIP proxy Server
3rd Party Billing
System
MGCP Device
VOCAL System [http://www.vovida.org/]
SIP based IP Telephony System
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 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 System
SIP proxy Server: Acts as a trusted boundary for calls entering or
leaving a network. Provides authentication and collects
billing information for the CDR server.
SIP based IP Telephony System
CDR Server: Collects billing information from
Marshal Servers and interfaces with billing systems using
the RADIUS accounting protocol.
SIP based IP Telephony System
Provisioning Server: Used to provision, configure and
manage subscribers and servers from a GUI.
SIP based IP Telephony System
Feature Server: Provide CPL based or XML scripts that
run basic telephony features.
VoIP 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 - Softswitch
Application
Servers
SS7
Gateway
SIP
SS7
Digital
Cross
Connect
CPL
Q.931/Q.2931
Softswitch
MGCP
Cellular
Station
H.323
IAD with
DSL/Cable Modem
MEGACO
MGCP
SIP
Media Gateways
3GPP Network Model
Endpoints with voice driving
converged IP infrastructure
PDA
IP Phones
PC to
Phone
Voice
Portals
Unified
Messaging
Instant
Messenger
Video
Telephony
Voice-enabled
Websites
Voice Service Focus
2. Voice-Enabled
Data VPN
1. Managed IP
Telephony
Soft
Switches
SS7
SOHO
PSTN
HQ
CallManager
IPSec or
MPLS
Messaging,
ACD, IVR
V
V
3. IP Centrex and
Hosted Apps
Internet
V
Branch
Office
Ent/SMB A Ent/SMB B
Enterprise A
HQ
Enterprise B
IOS Telephony
Services
Branch Office
4. Integrated
Access
All IP Network
3G/4G Wireless Coverage
Home WLAN
Office LAN
Hotel WLAN/LAN
LAN, WLAN hot spots
and 3G/4G wireless mobility
Restaurant WLAN
Airport WLAN
Wireless LAN Voice Mobility
The Big Technical Challenge:
802.11 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)
Internet
Hotspot A
(
AP
AP
Micro-Mobility
Hotspot B
AP
Macro-Mobility
AP
Micro-Mobility
Call 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
Cellular GPRS-WLAN Integration
HA
GPRS/
UTRAN
Network
CAG
HLR - AuC
GPRS CORE
SGSN
GGSN
Operators
IP
Network
CG
Billing Mediator
Internet
Billing System
WLAN Network
FA/AAA
Dual
Mode
MN
AP
BSS-1
AP
BSS-2
AP
BSS-N
AP:
BSS:
CG:
HLR:
AuC:
SGSN:
GGSN:
CAG:
FA:
HA:
WLAN Access Point
Basic Service Set
Charging Gateway
Home location register
Authorization center
Serving GPRS support node
Gateway GPRS support node
Cellular access gateway
Foreign Agent
Home Agent
SIP 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 Support
• Remote registration
Visit.com
Home.com
Move
From:[email protected]
Contact:166.1.2.3
From:[email protected]
Contact:[email protected]
From:[email protected]
Contact:166.4.5.6
INVITE
INVITE
INVITE
SIP Roaming Support
• Precall mobility
DHCP
Home.com
INVITE
IP Address
302 moved temporarily
ACK
INVITE
OK
ACK
MEDIA
SIP Roaming Support
• Midcall mobility
MEDIA
INVITE
中斷?
OK
ACK
MEDIA