1. 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
2003/07/26

2. 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

3. Review - Voice Conference
Basic issues of voice conference setup
phonebook
server
1. ?
2. Conference setup
3. Digital voice packets
AD/DA
compress/decompress
4. Conference terminate
AD/DA
compress/decompress

4. 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 、G.729

5. 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.

6. 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

7. Media Transport Media Processing Media Transmission

8. Process of digital voice transmission
Low-pass filter
Sampling &
A/D convert
Silent
detection
RTP packet
encapsulation
Compression
Internet
Timing
reconstruct
RTP packet
decapsulation
D/A convert
Decompression

9. VoIP Endpoint Functionality
PCM
DSP coding
Phone
interface
frames
Buffering and
packetization
Jitter buffer
Digital Signal
Processing
Packet
handling
AD/DA
converter
TCP/IP
protocol stack
Network
interface
copper wire IP

10. Digitalization Speech
Low Pass Filter (LPF) 300 Hz ~ 3000 Hz
Sampling and Quantization

11. Digitalization Speech
PCM (Pulse Code Modulation)
digital quantization introduces distortions

12. Digitalization Speech
main speech coding techniques
waveform codec, source codec and hybrid codec

13. R,G,B bitmap
Color
Transform
RTP packet
encapsulation
Y,Cb,Cr
matrix
Huffman
encoder
Huffman
table
Discrete
Cosine
Transform
frequency
matrix
Quantizer
Quantization
table

14. Media Transport Media Processing Media Transmission

16. Voice Quality of Service
Interactive Voice QoS factors
Packet lost
Delay
Jitter

17. Voice QoS - Packet Lost
Intranet
Internet

18. Voice QoS - Delay Minimize one-way delay, keep it below 150ms
ITU G.114 states one-way delay <= 150 msec ~200 msec is acceptable
Fixed delay
1. Framing: 20~30 ms
2. Processing: 15 ms
3. Transmission: 10ms
4. Decompress/buffer: 25 ms
Framing (algorithm): 20 ~ 30 ms
Compress (H/W DSP): 5 ms
Processing (packetize): 10 ms
Variable delay
1. Buffer: 5~20 ms
2. Network: 20 ~ ? ms
GPRS Backbone
IP Network
IP based network
variable delay
20~300 or more ms
Receiving buffer: 20 ms
Decompress delay: 5 ms

19. 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

20. Voice QoS - Delay
Internet
Intranet

21. Voice QoS - Jitter

22. Jitter (Delay Variation)
Internet
Intranet

23. Packet Handling Latency
Jitter
variability in the arrival rate of data is called jitter

24. Voice QoS - Jitter
Jitter buffer

25. Definitions Voice over IP (VoIP) IP Telephony
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

26. 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
(8 bytes)
RTP header
(12 bytes)
media payload

27. RTP: A Transport Protocol for Real-Time Applications (RFC 1889)

28. RTP (RFC1889) The simplest RTP fixed header IP header UDP header
RTP header
RTP payload

29. Fields of RTP Header V (version): P (padding): X (extension):
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

30. Fields of RTP Header Sequence number: Timestamp: Sync SRC:
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

31. RTP Header profile (RFC1900)

32. Signaling Addressing Call control Capabilities exchange

33. 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

34. VoIP signaling protocol standard
ITU-T H.323
IETF MGCP
RFC2705
IETF SIP
RFC3261
IETF/ITU-T Megaco/H.248
RFC3015

35. Session Initiation Protocol
SIP Architecture
RFC3261
SIP User
Agent
SIP User
Agent
SIP Server
SIP User
Agent
Registrar
Proxy
Server
Redirect
Server

36. VoIP protocol standard - SIP
SIP BASIC Call flow
Caller
Callee
Pickup & dial
INVITE SIP/2.0
…….
ringing
ringback
180, Ringing
pick up
200, OK
ACK
RTP (voice)
on-hook
BYE
ACK

37. Request 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.

38. Response Status Line SIP-Version SP Status-Code SP Reason-Phrase CRLF
SIP/2.0 SP 180 SP Ringing CRLF
1xx
Informational
2xx
Success
3xx
Redirection
4xx
Client-Error
5xx
Server-Error
6xx
Global-Failure

39. SIP Request Example INVITE sip:cdweng@netrd.iii.org.tw SIP/2.0
Method type, request URL and SIP version
Globally unique ID for this call
Content-type:application/sdp
The body type, an SDP message
Cseq:1 INVITE
Command Sequence number and type
User originating the request
User being invited into the call
Via:SIP/2.0/UDP :5060
IP Address and port of previous hop
Blank line separates header from body
v=0
SDP version
o=smayer IN IP
Owner/creator and session identifier
s=sip session
The name of session p=
Phone number of caller
c=IN IP
Connection information
t=0 0
Time the session is active
m=audio 49170/1 RTP/AVP 1
media name and transport

40. SIP Registration REGISTER sip:iptel.org SIP/2.0
Location Server
REGISTER sip:iptel.org SIP/2.0
Contact:<sip: >
Expires:3600
SIP/ OK
SIP Registrar
(domain: iptel.org)

41. SIP Operation in Proxy Mode
Location Server
jiri ?
INVITE
INVITE
ACK sip
SIP/ OK
SIP/ OK
SIP Proxy Server

42. SIP Operation in Redirect Mode
Location Server
Callee ?
INVITE
SIP Redirect
Server
302 Moved Temporarily
Contact:
ACK
SIP/ OK
INVITE
ACK

43. Call Control and Signaling
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
RTP
RTCP
RTSP
TCP
UDP
H.323 – packet based multimedia communication system
H.225 – call signaling protocol
H.245 – call control protocol
RAS – Registration Admission Signaling
SIP – Session Initiation Protocol (RFC 2543)
MGCP - Media Gateway Control Protocol
H.248/Megaco – Media Gateway Control Protocol
RTP – Real Time Transport Protocol (RFC 1889)
RTCP – Real Time Transport Control Protocol (RFC 1889)
RTSP – Real Time Streaming Protocol (RFC2324)
UDP – User Datagram Protocol
TCP - Transmission Control Protocol
IP – Internet Protocol 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.

44. Protocol wars - Viewpoint from CISCO
Projected Port (DS0) Protocol Transition Rates
100%
80%
60%
Mixed
H.323
& SIP
% Port Unit Sales
MGCP / H DS0s
40%
SIP DS0s
20%
H DS0s
Q1CY99
Q1CY00
Q1CY01
Q1CY02
Q1CY03
Q1CY04
Calendar Quarters

45. Next Generation Converged Network and IP Telephony system

46. Siemens Total Data Telephony 1 2 3 4 5 6 7 8 9 10 1997 1998 1999 20001 2 3 4 5 6 7 8 9 10
1997
1998
1999
2000
2001
2002
2003
Year
Relative traffic
Total
Data
Telephony
Siemens

47. 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

48. Next Generation Converged Network

49. Next Generation Converged Network
Residential Gateway / Integrated Access Device

50. Call Control and Switching
IP Telephony System
IP Telephony System must support
Feature and
Application Creation
Operation
System Support
Call Control and Switching

51. SIP based IP Telephony System
Clearing
House
Internet
SIP proxy Server
PSTN
Gateway
SIP IP Phone
MGCP Device
MGCP/SIP
Translator
H.323/SIP Translator
H.323 Terminal
Provisioning Server(s)
Feature Server(s)
CDR Server(s)
3rd Party Billing System
RADIUS
SNMP
Network
Manager
SIP based
VOCAL System [

52. 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.

53. 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.

54. 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.

55. SIP based IP Telephony System
CDR Server: Collects billing information from
Marshal Servers and interfaces with billing systems using
the RADIUS accounting protocol.

56. SIP based IP Telephony System
Provisioning Server: Used to provision, configure and manage subscribers and servers from a GUI.

57. SIP based IP Telephony System
Feature Server: Provide CPL based or XML scripts that run basic telephony features.

58. 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)

59. IP Telephony - Softswitch
Application Servers
SS7 Gateway
SIP
SS7
CPL
Digital Cross Connect
Q.931/Q.2931
Softswitch
MGCP
Cellular Station
MEGACO
MGCP
H.323
SIP
IAD with DSL/Cable Modem
Media Gateways

60. 3GPP Network Model

61. Endpoints with voice driving converged IP infrastructure
PDA
IP Phones
PC to
Phone
Instant
Messenger
Voice
Portals
Video
Telephony
Unified
Messaging
Voice-enabled
Websites

62. Voice Service Focus 1. Managed IP Telephony 2. Voice-Enabled Data VPN
Soft
Switches
SS7
SOHO HQ
PSTN
CallManager
3. IP Centrex and Hosted Apps
IPSec or
MPLS V V
Messaging, ACD, IVR
Internet V
Branch
Office
Ent/SMB A
Ent/SMB B
Enterprise A
IOS Telephony Services
4. Integrated Access HQ
Branch Office
Enterprise B

63. and 3G/4G wireless mobility
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

64. Wireless LAN Voice Mobility

65. 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 (
Hotspot B AP AP AP AP
Micro-Mobility
Macro-Mobility
Micro-Mobility

66. 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

67. An Example: Loosely Coupled Cellular GPRS-WLAN IntegrationHA
GPRS/
UTRAN
Network
CAG
HLR - AuC
Operators IP
Network
GPRS CORE
SGSN
GGSN
Internet CG
Billing Mediator
Billing System
WLAN Network
FA/AAA
Dual
Mode MN
AP: WLAN Access Point
BSS: Basic Service Set
CG: Charging Gateway
HLR: Home location register
AuC: Authorization center
SGSN: Serving GPRS support node
GGSN: Gateway GPRS support node
CAG: Cellular access gateway
FA: Foreign Agent
HA: Home Agent AP
BSS-1 AP
BSS-2 AP
BSS-N

68. SIP Roaming Support Logging into different IP networks away from home
Basic Steps:
Get an IP address
Use DHCP
Register with local proxy
For firewall transversal for UDP
Register with home Registrar
For calls routing

69. SIP Roaming Support Remote registration Visit.com Home.com
Contact:
Move
Contact:
INVITE
INVITE
INVITE

70. SIP Roaming Support Precall mobility MEDIA DHCP Home.com INVITE
IP Address
302 moved temporarily
ACK
INVITE OK
ACK
MEDIA

71. SIP Roaming Support
Midcall mobility
MEDIA
中斷?
INVITE OK
ACK
MEDIA