Title: Telephony
1Telephony Internet Telephony
- Shivkumar Kalyanaraman
- Based upon slides of Henning Schulzrinne
(Columbia)
2Telephony
3Telephone Network What is It?
- Specialized to carry voice traffic
- Aggregates like T1, SONET OC-N can also carry
data - Also carries
- Telemetry, video, fax, modem calls
- Internally, uses digital samples
- Switches and switch controllers are special
purpose computers - Pieces
- 1. End systems
- 2. Transmission
- 3. Switching
- 4. Signaling
4Telephone Network What is It?
- Single basic service two-way voice
- low end-to-end delay
- guarantee that an accepted call will run to
completion
- Endpoints connected by a circuit, like an
electrical circuit - Signals flow both ways (full duplex)
- Associated with reserved bandwidth and buffer
resources
5Public 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)
6Telephone System Overview
- Analog narrowband circuits home- central office
- 64 kb/s continuous transmission, with compression
across oceans - ?-law 12-bit linear range - 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)
7Telephony 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 call 64 Kbps
8Telephone Network Design
- Fully connected core
- simple routing
- telephone number is a hint about how to route a
call - But not for 800/888/700/900 numbers these are
pointers to a directory that translates them into
regular numbers - hierarchically allocated telephone number space
9Telephone Network Design
10Telephone Pieces End Systems
11Telephone Pieces End Systems
- Transducers key to carrying voice on wires
- Dialer
- Ringer
- Switch-hook
12Last-Mile Transmission Environment
- Wire gauges19, 22, 24, 26 gauge(smaller better)
- Diameters 0.8, 0.6, 0.5, 0.4 mm (larger better)
- Various forms of noise (twisting reduces noise)
- Bridged-tap noise bit-energy diverted to
extension phone sockets - Crosstalk
- Ham radio
- AM broadcast
- Insertion loss -140 dBm noise floor
- 100 million times more sensitive than normal
modems - Bandwidth range 600 kHz
- Notch effects in insertion loss due to
bridged-taps - Transmission PSD -40dBm 90 dBm budget
132-wire vs 4-wire Sidetones and Echoes
- Both trans reception circuits need two wires
- 4 wires from every central office to home
- Alternative Use same pair of wires for both
transmission and reception - Signal from transmission flows to receiver
sidetone
- Reverse Effect received signal at end-system
bounces back to CO (esp if delay 20 ms) echo - Solutions balance circuit (attenuate side-tone)
echo-cancellation circuit (cancel echoes).
14Dialing
- Pulse
- sends a pulse per digit
- collected by central office (CO)
- Interpreted by CO switching system to place call
or activate special features (eg call
forwarding, prepaid-calls etc) - Tone
- key press (feep) sends a pair of tones digit
- also called Dual Tone Multifrequency (DTMF)
- CO supplies the power for ringing the bell.
- Standardized interface between CO and end-system
digital handsets, cordless/cellular phones
15Telephone Pieces Transmission Muxing
- Trunks between central offices carry hundreds of
conversations - Cant run thick bundles! Instead, send many calls
on the same wire - Multiplexing (a.ka. Sharing)
- Analog multiplexing
- Band-limit call to 3.4 KHz and frequency shift
onto higher bandwidth trunk - obsolete
- Digital multiplexing
- first convert voice to samples
- 1 sample 8 bits of voice
- 8000 samples/sec call 64 Kbps
16Transmission Multiplexing (contd)
- How to choose a sample?
- 256 quantization levels, logarithmically spaced
(why?) - sample value amplitude of nearest quantization
level - Two choices of levels (? law and A law)
- Time division multiplexing
- Trunk carries bits at a faster bit rate than
inputs - n input streams, each with a 1-byte buffer
- Output interleaves samples
- Need to serve all inputs in the time it takes one
sample to arrive - output runs n times faster than input
- Overhead bits mark end of frame (why?)
17Transmission Multiplexing
- Multiplexed trunks can be multiplexed further
- Need a standard! (why?)
- US/Japan standard is called Digital Signaling
hierarchy (DS)
18Telephone Pieces Switching
19Telephone Pieces Switching
- Problem
- each user can potentially call any other user
- cant have (a billion) direct lines!
- Switches establish temporary circuits
- Switching systems come in two parts switch and
switch controller
20Switching System Components
21Switch What does it do?
- Transfers data from an input to an output
- many ports (up to 200,000 simultaneous calls)
- need high speeds
- Some ways to switch
- 1. space division switching eg crossbar
- if inputs (or crosspoints) are multiplexed, need
a schedule (why?)
22Crossbar Switching Elements
23Switching (Contd)
- Another way to switch
- time division (time slot interchange or TSI)
- also needs a service schedule (why?)
- To build larger switches we combine space and
time division switching elements
24Telephone pieces Signaling
- A switching system has a switch and a switch
controller - Switch controller is in the control plane
- does not touch voice samples
- Manages the network
- call routing (collect dialstring and forward
call) - alarms (ring bell at receiver)
- billing
- directory lookup (for 800/888 calls)
25Signaling
- Switch controllers are special purpose computers
- Linked by their own internal computer network
- Common Channel Interoffice Signaling (CCIS)
network - Earlier design used in-band tones, but was hacked
- Also was very rigid (why?)
- Messages on CCIS conform to Signaling System 7
(SS7)
26Signaling (contd)
- One of the main jobs of switch controller keep
track of state of every endpoint - Key is state transition diagram
27Telephony Routing of Signaled Calls
- Circuit-setup (I.e. the signaling call) is what
is routed. - Voice then follows route, and claims reserved
resources. - 3-level hierarchy, with a fully-connected core
- ATT 135 core switches with nearly 5 million
circuits - LECs may connect to multiple cores
28Telephony Routing algorithm
- If endpoints are within same CO, directly connect
- If call is between COs in same LEC, use one-hop
path between COs - Otherwise send call to one of the cores
- Only major decision is at toll switch
- one-hop or two-hop path to the destination toll
switch. - Essence of telephony routing problem
- which two-hop path to use if one-hop path is
full - (almost a static routing problem )
29Features of telephone routing
- Resource reservation aspects
- Resource reservation is coupled with path
reservation - Connections need resources (same 64kbps)
- Signaling to reserve resources and the path
- Stable load
- Network built for voice only.
- Can predict pairwise load throughout the day
- Can choose optimal routes in advance
- Technology and economic aspects
- Extremely reliable switches
- Why? End-systems (phones) dumb because
computation was non-existent in early 1900s. - Downtime is less than a few minutes per year
topology does not change dynamically
30Features of telephone routing
- Source can learn topology and compute route
- Can assume that a chosen route is available as
the signaling proceeds through the network - Component reliability drove system reliability
and hence acceptance of service by customers - Simplified topology
- Very highly connected network
- Hierarchy full mesh at each level simple
routing - High cost to achieve this degree of connectivity
- Organizational aspects
- Single organization controls entire core
- Afford the scale economics to build expensive
network - Collect global statistics and implement global
changes - Source-based, signaled, simple alternate-path
routing
31Telecommunications Regulation History
- FCC regulations cover telephony, cable, broadcast
TV, wireless etc - Common Carrier provider offers conduit for a
fee and does not control the content - Customer controls content/destination of
transmission assumes criminal/civil
responsibility for content - Local monopolies formed by ATTs acquisition of
independent telephone companies in early 20th
century - Regulation forced because they were deemed
natural monopolies (only one player possible in
market due to enormous sunk cost) - FCC regulates interstate calls and state
commissions regulate intra-state and local calls - Bells 1000 independents interconnected
expanded - FCC rulemaking process
- Intent to act, solicitation of public comment etc
32Deregulation of telephony
- 1960s-70s gradual de-regulation of ATT due to
technological advances - Terminal equipment could be owned by customers
(CPE) explosion in PBXs, fax machines,
handsets - Modified final judgement (MFJ) breakup of ATT
into ILECs (incumbent local exchange carrier) and
IXC (inter-exchange carrier) part - Long-distance opened to competition, only the
local part regulated - Equal access for IXCs to the ILEC network
- 1 long-distance number introduced then
- 800-number portability switching IXCs retain
800 number - 1995 removed price controls on ATT
33Telecom Act of 1996
- Required ILECs to open their markets through
unbundling of network elements (UNE-P),
facilities ownership of CLECs. - Today UNE-P is one of the most profitable for
ATT and other long-distance players in the local
market due to apparently below-cost regulated
prices - ILECs could compete in long-distance after
demonstrating opening of markets - Only now some ILECs are aggressively entering
long distance markets - CLECs failed due to a variety of reasons
- But long-distance prices have dropped
precipitously (ATTs customer unit revenue in
2002 was 11.3 B compared to 1999 rev of 23B) - ILECs still retain over 90 of local market
- Wireless substitution has caused ILECs to develop
wireless business units
34US Telephone Network Structure (after 1984)
35Exchange Area Network
36IP Telephony, VoIP etc
37IP Telephony Overview
- 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
38Telephone Service Penetration in the US
ATT Divestiture
39Trends Price of Phone Calls NY - London
ATT Divestiture
40Trends Data vs Voice Traffic
Since we are building future networks for data,
can we slowly junk the voice infrastructure and
move over to IP?
41Trends Phone vs Data Revenues
42Private 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
43IP Telephony PBX Replacement
Another campus
Corporate/Campus
7040
8151
External line
8152
7041
PBX
PBX
8153
7042
8154
7043
Internet
LAN
LAN
44Voice over Packet Market Forecast North America
45Invisible 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
46IPtel for appliances Presence
47Taxonomy 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)
48Speech Quality of Various Coders
49Applications 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)
50Pulse Amplitude Modulation (PAM)
51Pulse Code Modulation (PCM)
PCM PAM quantization
52Quantization
53Companded 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
54Companding
55How it works for T1 Lines
- Companding blocks are shared by all 16 channels
56Adaptive Gain Encoding
Automatic Gain control (AGC), but accounting for
silence periods
57Time Waveform of Voiced/Unvoiced Sound
High correlation (0.85) between samples, cycles,
pitch intervals etc
58Differential PCM
Exploits sample-to-sample correlation (0.85)
differences require fewer bits feedback avoids
cascading quantization errors
59Delta Modulation
- Used in first-generation PBXs (switching was more
sensitive to - Digital conversion cost and less sensitive to
quality or data rate)
60Adaptive Predictive Coding
Adapt both the prediction coefficients (alphas)
and the estimates Based upon past or present
samples 20 dB prediction gain
61Subband 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
62G.722 (7 kHz) audio Codec
63Recall 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
64Vocoders
Encode only perceptually important aspects of
speech w/ fewer bits than waveform coders eg
power spectrum vs time-domain accuracy
65LPC Analysis/Synthesis
66Speech Generation in LPC
67Multi-pulse LPC
68CELP Encoder
69Example 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
13 kbps (Full-Rate coding).
70Speech Quality of Various Coders
71Speech Quality (Contd)
72VoIP 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
73Internet Multimedia Protocol Stack
74IP 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
75Internet Telephony Protocols H.323
76H.323 (contd)
- Terminals, Gateways, Gatekeepers, and Multipoint
Control Units (MCUs)
77H.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
78SIP 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
79Light-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)
80SDP 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
81SIP 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
82SIP Addresses Food Chain
83SIP 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)
84IP 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
85SIP-based Architecture
86Example 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
87PSTN to IP Call
88IP to PSTN Call
89Traditional voice mail system
Bob can listen to his voice mails by dialing some
number.
90SIP-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.
91Voicemail 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
92SIP-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
93MGCP 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
94MGCP Architecture
Goal large-scale phone-to-phone VoIP deployments
RGW Residential Gateway TGW Trunk Gateway
95Summary
- 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