Internet Telephony

1
Internet Telephony

• Shivkumar Kalyanaraman
• Based upon slides of Henning Schulzrinne
  (Columbia)

2
Overview

• Telephony history and evolution
• IP Telephony Why ?
• Adding interactive multimedia to the web
• Being able to do basic telephony on IP with a
  variety of devices
• Basic IP telephony model
• Protocols SIP, H.323, RTP, Coding schemes, MGCP,
  RTSP
• Future Invisible IP telephony and control of
  appliances

3
Public Telephony (PSTN) History

• 1876 invention of telephone
• 1915 first transcontinental telephone (NYSF)
• 1920s first automatic switches
• 1956 TAT-1 transatlantic cable (35 lines)
• 1962 digital transmission (T1)
• 1965 1ESS analog switch
• 1974 Internet packet voice
• 1977 4ESS digital switch
• 1980s Signaling System 7 (out-of-band)
• 1990s Advanced Intelligent Network (AIN)

4
Telephone Service in the US
ATT Divestiture
5
Telephone System Overview

• Analog narrowband circuits home-gt central office
• 64 kb/s continuous transmission, with compression
  across oceans
• ?-law 12-bit linear range -gt 8-bit bytes
• Everything clocked a multiple of 125 s
• Clock synchronization ? framing errors
• ATT 136 tollswitches in U.S.
• Interconnected by T1, T3 lines SONET rings
• Call establishment out-of-band using
  packet-switched signaling system (SS7)

6
Telephony Multiplexing

• Telephone Trunks between central offices carry
  hundreds of conversations Cant run thick
  bundles!
• Send many calls on the same wire multiplexing
• Analog multiplexing
• bandlimit call to 3.4 KHz and frequency shift
  onto higher bandwidth trunk
• Digital multiplexing convert voice to samples
• 8000 samples/sec gt call 64 Kbps

7
Trends Price of Phone Calls NY - London
ATT Divestiture
8
Trends Data vs Voice Traffic
Since we are building future networks for data,
can we slowly junk the voice infrastructure and
move over to IP?
9
Trends Phone vs Data Revenues
10
Private Branch Exchange (PBX)
Post-divestiture phenomenon...
7040
212-8538080
External line
7041
Telephone switch
Corporate/Campus
Private Branch Exchange
Another switch
7042
7043
Internet
Corporate/Campus LAN
11
IP Telephony PBX Replacement
Another campus
Corporate/Campus
7040
8151
External line
8152
7041
PBX
PBX
8153
7042
8154
7043
Internet
LAN
LAN
12
Voice over Packet Market Forecast North America
13
Invisible Internet Telephony

• VoIP technology will appear in . . .
• Internet appliances
• home security cameras, web cams
• 3G mobile terminals
• fire alarms
• chat/IM tools
• interactive multiplayer games

14
IPtel for appliances Presence
15
Taxonomy of Speech Coders

• Waveform coders attempts to preserve the
  signal waveform not speech specific (I.e. general
  A-to-D conv)
• PCM 64 kbps, ADPCM 32 kpbs, CVSDM 32 kbps
• Vocoders
• Analyse speech, extract and transmit model
  parameters
• Use model parameters to synthesize speech
• LPC-10 2.4 kbps
• Hybrids Combine best of both Eg CELP (used in
  GSM)

16
Speech Quality of Various Coders
17
Applications of Speech Coding

• Telephony, PBX
• Wireless/Cellular Telephony
• Internet Telephony
• Speech Storage (Automated call-centers)
• High-Fidelity recordings/voice
• Speech Analysis/Synthesis
• Text-to-speech (machine generated speech)

18
Pulse Amplitude Modulation (PAM)
19
Pulse Code Modulation (PCM)
PCM PAM quantization
20
Quantization
21
Companded PCM

• Small quantization intervals to small samples and
  large intervals for large samples
• Excellent quality for BOTH voice and data
• Moderate data rate (64 kbps)
• Moderate cost used in T1 lines etc

22
Companding
23
How it works for T1 Lines

• Companding blocks are shared by all 16 channels

24
Adaptive Gain Encoding
Automatic Gain control (AGC), but accounting for
silence periods
25
Time Waveform of Voiced/Unvoiced Sound
High correlation (0.85) between samples, cycles,
pitch intervals etc
26
Differential PCM
Exploits sample-to-sample correlation (0.85) gt
differences require fewer bits feedback avoids
cascading quantization errors
27
Delta Modulation

• Used in first-generation PBXs (switching was more
  sensitive to
• Digital conversion cost and less sensitive to
  quality or data rate)

28
Adaptive Predictive Coding
Adapt both the prediction coefficients (alphas)
and the estimates Based upon past or present
samples gt 20 dB prediction gain
29
Subband Coding
Frequency domain analysis of input instead of
time-domain Analysis adjust quantization based
upon energy level of each band Eg G.722 coder
7kHz voice w/ 64 kbps
30
G.722 (7 kHz) audio Codec
31
Recall Taxonomy of Speech Coders

• Waveform coders attempts to preserve the
  signal waveform not speech specific.
• PCM 64 kbps, ADPCM 32 kpbs, CVSDM 32 kbps
• Vocoders
• Analyse speech, extract and transmit model
  parameters
• Use model parameters to synthesize speech
• LPC-10 2.4 kbps
• Hybrids Combine best of both Eg CELP

32
Vocoders
Encode only perceptually important aspects of
speech w/ fewer bits than waveform coders eg
power spectrum vs time-domain accuracy
33
LPC Analysis/Synthesis
34
Speech Generation in LPC
35
Multi-pulse LPC
36
CELP Encoder
37
Example GSM Digital Speech Coding

• PCM 64kbps too wasteful for wireless
• Regular Pulse Excited -- Linear Predictive Coder
  (RPE--LPC) with a Long Term Predictor loop.
• Subjective speech quality and complexity (related
  to cost, processing delay, and power)
• Information from previous samples used to predict
  the current sample linear function.
• The coefficients, plus an encoded form of the
  residual (predicted - actual sample), represent
  the signal.
• 20 millisecond samples each encoded as 260 bits
  gt13 kbps (Full-Rate coding).

38
Speech Quality of Various Coders
39
Speech Quality (Contd)
40
VoIP Camps
Circuit switch engineers We over IP
Convergence ITU standards
Conferencing Industry
Netheads IP over Everything
H.323
SIP
Softswitch
BICC
ISDN LAN conferencing
I-multimedia WWW
Call Agent SIP H.323
BISDN, AIN H.xxx, SIP
IP
IP
IP
any packet
41
Internet Multimedia Protocol Stack
42
IP Telephony Protocols SIP, RTP

• Session Initiation Protocol - SIP
• Contact office.com asking for bob
• Locate Bobs current phone and ring
• Bob picks up the ringing phone
• Real time Transport Protocol - RTP
• Send and receive audio packets

43
Internet Telephony Protocols H.323
44
H.323 (contd)

• Terminals, Gateways, Gatekeepers, and Multipoint
  Control Units (MCUs)

45
H.323 vs SIP
Typical UserAgent Protocol stack for Internet
Terminal Control/Devices
Terminal Control/Devices
Q.931
H.245
RTCP
RAS
RTCP
SIP
SDP
Codecs
Codecs
RTP
RTP
TPKT
TCP
UDP
Transport Layer
IP and lower layers
46
SIP vs H.323

• Binary ASN.1 PER encoding
• Sub-protocols H.245, H.225 (Q.931, RAS,
  RTP/RTCP), H.450.x...
• H.323 Gatekeeper

• Text based request response
• SDP (media types and media transport address)
• Server roles registrar, proxy, redirect

- Both use RTP/RTCP over UDP/IP - H.323 perceived
as heavyweight
47
Light-weight signaling Session
InitiationProtocol (SIP)

• IETF MMUSIC working group
• Light-weight generic signaling protocol
• Part of IETF conference control architecture
• SAP for Internet TV Guide announcements
• RTSP for media-on-demand
• SDP for describing media
• others malloc, multicast, conference bus, . . .
• Post-dial delay 1.5 round-trip time (with UDP)
• Network-protocol independent UDP or TCP (or AAL5
  or X.25)

48
SDP Session Description Protocol

• Not really a protocol describes data carried by
  other protocols
• Used by SAP, SIP, RTSP, H.332, PINT. Eg
• v0
• og.bell 877283459 877283519 IN IP4 132.151.1.19
• sCome here, Watson!
• uhttp//www.ietf.org
• eg.bell_at_bell-telephone.com
• cIN IP4 132.151.1.19
• bCT64
• t3086272736 0
• kclearmanhole cover
• maudio 3456 RTP/AVP 96
• artpmap96 VDVI/8000/1
• mvideo 3458 RTP/AVP 31
• mapplication 32416 udp wb

49
SIP functionality

• IETF-standardized peer-to-peer signaling protocol
  (RFC 2543)
• Locate user given email-style address
• Setup session (call)
• (Re)-negotiate call parameters
• Manual and automatic forwarding
• Personal mobility different terminal, same
  identifier
• Call center reach first (load distribution) or
  reach all (department conference)
• Terminate and transfer calls

50
SIP Addresses Food Chain
51
SIP components

• UAC user-agent client (caller application)
• UAS user-agent server à accept, redirect, refuse
  call
• redirect server redirect requests
• proxy server server client
• registrar track user locations
• user agent UAC UAS
• often combine registrar (proxy or redirect
  server)

52
IP SIP Phones and Adaptors
1

• Are true Internet hosts
• Choice of application
• Choice of server
• IP appliances
• Implementations
• 3Com (3)
• Columbia University
• MIC WorldCom (1)
• Mediatrix (1)
• Nortel (4)
• Siemens (5)

Analog phone adaptor
2
3
Palm control
4
5
4
53
SIP-based Architecture
54
Example Call

• Bob signs up for the service from the web as
  bob_at_ecse.rpi.edu

• sipd canonicalizes the destination to
  sipbob_at_ecse.rpi.edu

• He registers from multiple phones

• sipd rings both ephone and sipc

• Bob accepts the call from sipc and starts talking

• Alice tries to reach Bob
• INVITE ipBob.Wilson_at_ecse.rpi.edu

ecse.rpi.edu
55
PSTN to IP Call
56
IP to PSTN Call
57
Traditional voice mail system
Bob can listen to his voice mails by dialing some
number.
58
SIP-based Voicemail Architecture
vm.office.com
The voice mail server registers with the SIP
proxy, sipd
Alice calls bob_at_office.com through SIP proxy.
SIP proxy forks the request to Bobs phone as
well as to a voicemail server.
59
Voicemail Architecture
v-mail
vm.office.com
After 10 seconds vm contacts the RTSP server for
recording.
vm accepts the call.
Sipd cancels the other branch and ...
rtspd
...accepts the call from Alice.
Now user message gets recorded
60
SIP-H.323 Interworking ProblemsEg Call setup
translation
H.323
SIP
Q.931 SETUP
INVITE
Destination address (Bob_at_office.com)
Q.931 CONNECT
200 OK
Terminal Capabilities
Media capabilities (audio/video)
Terminal Capabilities
ACK
Open Logical Channel
Media transport address (RTP/RTCP receive)
Open Logical Channel

• H.323 Multi-stage dialing

61
MGCP and Megaco

• Media Gateway Controller Protocol (RFC 2705)
• Controlling Telephony Gateways from external call
  control elements called media gateway controllers
  (MGC) or call agents
• Gateways Eg RGW physical interfaces between
  VoIP network and residences
• Call control "intelligence" is outside the
  gateways and handled by external call control
  elements
• Goal scalable gateways between IP telephony and
  PSTN
• Successor to MGCP H.248/Megaco

62
MGCP Architecture
Goal large-scale phone-to-phone VoIP deployments
RGW Residential Gateway TGW Trunk Gateway
63
Summary

• Telephony and IP Telephony
• Protocols SIP, SDP, H.323, MCGP
• Example operation and services
• Calls, voice mail etc
• Future Integration with Web and long-term
  replacement for current telephone systems