Internet Telephony

Shivkumar KalyanaramanRensselaer Polytechnic Institute

1

Internet Telephony

Shivkumar KalyanaramanBased upon slides of Henning Schulzrinne (Columbia)


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Telephony: history and evolution IP Telephony: Why ?

Adding interactive multimedia to the webBeing 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

Overview


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Public Telephony (PSTN) History 1876 invention of telephone 1915 first transcontinental telephone (NY–SF) 1920’s 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)


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Telephone Service in the USAT&T Divestiture


5

Telephone 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 sClock synchronization framing errors

AT&T: 136 “toll”switches in U.S. Interconnected by T1, T3 lines & SONET rings

Call establishment “out-of-band” using packet-switched signaling system (SS7)


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Telephony: Multiplexing

Telephone Trunks between central offices carry hundreds of conversations: Can’t 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


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Trends: Price of Phone Calls: NY - LondonAT&T Divestiture


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Trends: Data vs Voice Traffic

Since we are building future networks for data, canwe slowly junk the voice infrastructure and move over to IP?


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Trends: Phone vs Data Revenues


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Private Branch Exchange (PBX)

7043

7040

7041

7042

External line

Telephoneswitch

Private BranchExchange

212-8538080

Anotherswitch

Corporate/Campus

InternetCorporate/Campus LAN

Post-divestiture phenomenon...


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External line

7043

7040

7041

7042

PBX

Corporate/Campus

InternetLAN

8154

8151

8152

8153

PBX

Another campus

LAN

IP Telephony: PBX Replacement


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Voice over Packet Market Forecast – North America


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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


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IPtel for appliances: “Presence”


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Taxonomy of Speech CodersSpeech Coders

Waveform Coders Source Coders

Time Domain: PCM, ADPCM

Frequency Domain: e.g. Sub-band coder,Adaptive transform coder

Linear Predictive Coder

Vocoder

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 kbpsVocoders:

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)


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Speech Quality of Various Coders


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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)


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Pulse Amplitude Modulation (PAM)


19

Pulse Code Modulation (PCM)

* PCM = PAM + quantization


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Quantization


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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


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Companding


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How it works for T1 Lines

• Companding blocks are shared by all 16 channels


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Adaptive Gain Encoding

Automatic Gain control (AGC), but accounting for silence periods


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Time Waveform of Voiced/Unvoiced Sound

High correlation (0.85) between samples, cycles, pitch intervals etc


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Differential PCM

Exploits sample-to-sample correlation (0.85) => differences require fewer bits; feedback avoids cascading quantization errors


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Delta Modulation

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


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Adaptive Predictive Coding

Adapt both the prediction coefficients (alphas) and the estimatesBased upon past or present samples => 20 dB prediction gain


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Subband Coding

Frequency domain analysis of input instead of time-domain Analysis: adjust quantization based upon energy level of each bandEg: G.722 coder: 7kHz voice w/ 64 kbps


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G.722 (7 kHz) audio Codec


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Recall: Taxonomy of Speech CodersSpeech Coders

Waveform Coders Source Coders

Time Domain: PCM, ADPCM

Frequency Domain: e.g. Sub-band coder,Adaptive transform coder

Linear Predictive Coder

Vocoder

Waveform coders: attempts to preserve the signal waveform not speech specific.

PCM 64 kbps, ADPCM 32 kpbs, CVSDM 32 kbpsVocoders:

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


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Vocoders

Encode only perceptually important aspects of speech w/ fewer bits than waveform coders: eg: power spectrum vs time-domain accuracy


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LPC Analysis/Synthesis


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Speech Generation in LPC


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Multi-pulse LPC


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CELP Encoder


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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 =>13 kbps (Full-Rate coding).


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Speech Quality of Various Coders


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Speech Quality (Contd)


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VoIP Camps

ISDN LAN conferencin

gIP

H.323I-multimedia

WWW

IP

SIPCall Agent

SIP & H.323

IP

“Softswitch” BISDN, AIN

H.xxx, SIP

“any packet”

BICC

Conferencing Industry

Netheads“IP over

Everything”

Circuit switch

engineers “We over

IP”

“Convergence” ITU

standards

Our focus


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Internet Multimedia Protocol Stack


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IP Telephony Protocols: SIP, RTP

Session Initiation Protocol - SIP Contact “office.com” asking for “bob” Locate Bob’s current phone and ring Bob picks up the ringing phone

Real time Transport Protocol - RTP Send and receive audio packets


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Internet Telephony Protocols: H.323


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H.323 (contd) Terminals, Gateways, Gatekeepers, and

Multipoint Control Units (MCUs)


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H.323 vs SIP

IP and lower layersTCP UDP

TPKT

Q.931 H.245 RAS RTCPRTP

Codecs

Terminal Control/Devices

Transport Layer

SIP SDPRTP

CodecsRTCP

Terminal Control/Devices

Typical UserAgent Protocol stack for Internet


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SIP vs H.323

Text based request response

SDP (media types and media transport address)

Server roles: registrar, proxy, redirect

Binary ASN.1 PER encoding

Sub-protocols: H.245, H.225 (Q.931, RAS, RTP/RTCP), H.450.x...

H.323 Gatekeeper

- Both use RTP/RTCP over UDP/IP- H.323 perceived as “heavyweight”


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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)


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SDP: Session Description Protocol Not really a protocol – describes data carried by other

protocols Used by SAP, SIP, RTSP, H.332, PINT. Eg:

v=0o=g.bell 877283459 877283519 IN IP4 132.151.1.19s=Come here, Watson!u=http://[email protected]=IN IP4 132.151.1.19b=CT:64t=3086272736 0k=clear:manhole coverm=audio 3456 RTP/AVP 96a=rtpmap:96 VDVI/8000/1m=video 3458 RTP/AVP 31m=application 32416 udp wb


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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


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SIP Addresses Food Chain


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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)


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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)

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IP SIP Phones and Adaptors1

3

Analog phone adaptor

Palmcontrol

2

54


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SIP-based Architecture

SIP proxy,redirectserver

SQLdatabase

sipd

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

SIPH.323

convertor

NetMeetingsip323

H.323

rtspd

SIP/RTSPUnified

messaging

RTSP media server

sipum

Quicktime

RTSP clients

RTSP

SIP conference

server

sipconf

T1/E1 RTP/SIP

Telephone

Cisco 2600 gateway

Telephoneswitch Web based

configurationWeb

server


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SIP proxy,redirectserver

SQLdatabase

sipd

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

Web based configuration

Web server

Call Bob

Example Call• Bob signs up for the service from the web as “[email protected]”• He registers from multiple phones

• Alice tries to reach Bob INVITE ip:[email protected]

• sipd canonicalizes the destination to sip:[email protected]

• sipd rings both e*phone and sipc

• Bob accepts the call from sipc and starts talking

ecse.rpi.edu


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PSTN to IP Call

PBXPSTN

External T1/CAS

Regular phone(internal)

Call 93971341

SIP server

sipd

Ethernet

3

SQLdatabase

4 7134 => bob

sipc

5

Bob’s phone

GatewayInternal T1/CAS(Ext:7130-7139)

Call 71342


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IP to PSTN Call

Gateway(10.0.2.3)

3

SQLdatabase

2Use sip:[email protected]

Ethernet

SIP server

sipdsipc

1Bob calls 5551212

PSTN

External T1/CASCall 55512125

5551212

PBX

Internal T1/CASCall 85551212 4

Regular phone(internal, 7054)


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Traditional voice mail system

Alice939-7063

Bob853-8119

Dial 853-8119

Phone is ringing

.. The person is not available nowplease leave a message ...

... Your voice message ...

Disconnect

Bob can listen to his voice mails by dialing some number.


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SIP-based Voicemail Architecture

INVITE [email protected]

Alice

phone1.office.com

Bob

Alice calls [email protected] through SIP proxy.SIP proxy forks the request to Bob’s phone as well as to a voicemail server.

vm.office.com

The voice mail server registers with the SIP proxy, sipd



REGISTER [email protected]


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Voicemail Architecture

v-mail

rtspd

Alice

vm.office.com;

200 OK

200 OK

CANCEL

SETUP

RTP/RTCP

phone1.office.com;

Bob

After 10 seconds vm contacts the RTSP server for recording.

vm accepts the call.Sipd cancels the other branch and ......accepts the call from Alice.Now user message gets recorded


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SIP-H.323: Interworking ProblemsEg: Call setup translation

Q.931 SETUP

Q.931 CONNECTINVITE

200 OK

ACK

Terminal Capabilities

Terminal CapabilitiesOpen Logical Channel

Open Logical Channel

H.323 SIP

Destination address ([email protected])

Media capabilities (audio/video)

Media transport address (RTP/RTCP receive)

• H.323: Multi-stage dialing


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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


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MGCP Architecture

RGW: Residential GatewayTGW: Trunk Gateway

Goal: large-scale phone-to-phone VoIP deployments


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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

Date post:	22-Feb-2016
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Internet Telephony

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