CS644Advanced Topics in

Networking

VoIP

Wed, 2004/9/8

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VoIP

• Basics• Codecs• Performance issues

– Delay, loss, quality– Other subjective/objective measures

• Available tools

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History of VoIP

• 1980s– Packet voice over satellite links in Lincoln Lab

• Early 1990s– MBone popularized audio and video conferenci

ng tools– Nevot, vat, vic, rat

• Late 1990s– Commercialization of VoIP

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Digitizing Voice

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Time

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Talkspurts

Talk and Silence in Voice

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Time

Silence period

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Voice to Packets

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Time

Look-ahead

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Popular Waveform Codecs

• PCM (Pulse Code Modulation)– G.711– Audible frequency (300-3KHz)– By Nyquist theorem, sample 8bits at 8KHz = 6

4 Kbps

• ADPCM (Adaptive Differential PCM)– G.726– Sample differences: 4bit at 8KHz

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Alternative for compression

• companding– non-linear quantization: μ-law (G.711)

• waveform– exploit statistical correlation

• model– model voice, extract parameters

• subband– split signal into bands and cod e individually (MPEG au

dio)• exploit masking properties of human ear

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Other Codecs

• G.729 for mobile telephony– Conjugate structure, algebraic-code-excited linear prediction (CS-ACELP)– 80bits at 100Hz = 8 kbps– requires 40% of 100MHz Pentium– G.729A: reduced-complexity version

• G.723.1 for videotelephony– MP-MLQ (Multipulse, multilevel quantization)– 30ms blocks of 240 16-bit samples into 24B = 6.3kbps– requires 25% of 100MHz Pentium

• GSM full rate– GSM 06.10, 13Kbps– Regular Pulse Excited (RPE) codec– GSM 06.20 = half-rate

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Basic Attributes of Codecs• Bit rate

– economy of bandwidth– in multimedia applications, more b/w to video

• Silence compression– voice activity detector (VAD)– discontinuous transmission (DTX)– comfort noise generator (CNG)

• Complexity– limiting factor in terms of memory and CPU (e.g. in portable devicies)

• Delay– frame-size, look-ahead: algorithmic delay– processing delay: depends on complexity and H/W

• Resilience to loss• Quality• Others

– layered coding– decoding telephone digits?

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RTP/RTCP

• Use of Real Time Protocol – adds sequence number, timestamp, payload ty

pe

• Use of Real Time Control Protocol– transmits control packets to participants– offers feedback on performance– correlate and synchronize different media stre

ams

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RTP Header

Version

Padding

Extension bit

A/V encoding method Sequence Number

Timestamp

Synchronization Source Identifier (SSRC)

Payload

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Session Management

• H.323• SIP (Session Initiation Protocol)

– e2e, client-server session signalling protocol– not a transport/QoS reservation/gateway contr

ol protocol

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H.323

• Suite of ITU-T recomm. protocols for multimedia collaboration among 2+ entities

• H.225.0-Q.931– pt2pt call signalling– initiation/proceeding/alerting/connection/termination

• H.245– call-control– master-slave determination/terminal capability set/log

ical channel management/

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H.323 Call Flow

Setup

Alerting/Connect

Capabilities Exchange / MSD

Open Logical Channel

Open Logical Channel Acknowledge

RTP Stream

RTP Stream

RTCP Stream

H.225(TCP port 1720)

H.245(TCP dynamic port)

UDP

End System A

End System B

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Key Components of SIP

• SIP end devices– UA client (originates calls)– UA server (listens for incoming calls)

• SIP workhorses– proxy server

• relays call signaling (i.e. acts as both client/server)– redirect server

• redirects callers to other servers– registar

• accept registration requests from users• maintains user’s whereabouts at a Location Server

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Basic SIP Capabilities

• user location– correct device with which to comm to reach a particula

r user• user availability

– willing/able to take part in a session• user capabilities

– determines choice of media, codecs• session setup

– establishes session parameters, s.a. port nubmers• session management

– call forwarding, modifying session parameters

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SIP RFC 2543 Methods

• INVITE• ACK• BYTE• CANCEL• OPTIONS• REGISTER

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Session/Call Control - SIP

cisco.com proxy

princeton.eduproxy

mypc.cisco.com yourpc.princeton.edu

Sue@cisco.com Meeyoung@princeton.edu

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Performance Issues

• Delay– transmission, router-processing, propagation, and que

ueing delays– packetization

• Loss– network drop– late arrival

• FEC (Forward Error Correction)– Adds redundant data– Issues?

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Playout Delay Adjustment

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Playout Delay

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Voice Quality Rating Measures

• MOS (Mean Opinion Score)– range of 1.0 to 4.5 (very poor to best)– mean of response scores of human beings

• PSQM– Perceptual Speech Quality Measurement– Automated human listener– Designed to measure the perceived quality of codec-quality voice based

on– Evaluates original signal against “perceptual model” and compares code

d signal againt it based on• distortion, effects of noise, and overall perceptual fidelity

• E-model– ITU-T Recommendation G.107– range of 0 to 100– R = R0 -Is- Id- Ie+A– effects of noise, simultaneous impairments at quantization, mouth-to-de

lay impairment, signal distortion, advantage factor (willingness to tolerate deterioration)

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Readily Available Tools

• Audio on Linux– /dev/audio for μ-law– /dev/dsp for general samples

• Java sound I/F– java.sun.com/proudcts/java-media/sound/– higher layer

• Java Media Framework (JMF) includes RTP

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Other Public VoIP Tools

• rat (Robust Audio Tool)– http://www-mice.cs.ucl.ac.uk/multimedia/soft

ware/rat/

• Public VoIP S/W repository– http://www.vovida.org/

• Many VoIP freeware for Windows

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References

• Voice over IP Fundamentals, J. Davidson, J. Peters, 2000, Cisco Press

• ITU-T Recomm. P.861 [PSQM]• IP Telephony with H.323, V. Kumar, M. Korpi, S. Sengodan, 2000,

Wiley [H.323]• C. Boutremans, J-Y Le Boudec, “Adaptive Joint Playout Buffer and

FEC Adjustment for Internet Telephony”, INFOCOM 2003