VoIP Overview 032105

8/3/2019 VoIP Overview 032105

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QUALCOMM PROPRIETARY DO NOT DISTRIBUTEMarch 2005 VoIP Overview

VoIP on 1xEVVoIP on 1xEV--DO rev. ADO rev. AQQUALCOMM,UALCOMM, March 20March 20


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AgendaAgenda

Executive Summary

Telco-Quality VoIP Requirements

Network Architecture for VoIP over EV-DO rev A

Voice over Wireless IP: performance enhancing features

Capacity Simulation Results

QUALCOMM PROPRIETARY DO NOT DISTRIBUTEM h 2005 V IP O i


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Executive SummaryExecutive Summary

VoIP capacity of 48 users per sector can be achieved on EV-DO rev Awith dual-diversity handsets

Capacity is comparable to 1x circuit-switched voice capacity if similar

assumptions were made (e.g. 3GPP2 evaluation framework, smartblanking, MSO model, etc)

Additional capacity gains can be achieved by introducing interferencecancellation techniques. In this case, DO rev A can support up to 58users per sector

The VoIP over DO rev A system is reverse link limited, leaving asignificant amount of unused forward link capacity. This free capacitycan be used for forward link centric applications such as Gold orPlatinum multicast.

In order to successfully implement VoIP over DO rev A, some standardsneed to be revised and the corresponding features implemented. E.g. IS-

835-D, IS-878-A and TIA-1054



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Wireless VoIP RequirementsWireless VoIP Requirements

VoIP calls must have at least similar quality as circuit-switched voice calls

End-to-end delays for mobile-to-land and mobile-to-mobile calls should be similar toexisting networks (e.g. 280 msec)

Frame error rate should be controlled to no more than 2%

Voice quality degradation should be minimal during handoffs

Need to support existing vocoders and possibly new enhanced ones

Capacity of the wireless VoIP system must be comparable or better than

existing wireless circuit-switched systems

Backend solutions need to be in place to support equivalent functions asin todays systems

Authorization, Accounting, Roaming, etc

VoIP implementation should allow for the development of new integratedmedia services in addition to existing voice services



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Network Architecture and ProtocolsNetwork Architecture and Protocols



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Wireless VoIP Network ElementsWireless VoIP Network Elements

1. 1xEV-DO rev A RAN Provides delay-sensitive QoS support and admission control

Provides QoS reservation and activation based on enhanced

multi-flow packet application (EMFPA) protocol

Supports IS-835-D, IS-878-A and RoHC

2. SIP-based VoIP core elements

Registration, call setup and control

Gateway for PSTN calls Codec transcoding

3. EV-DO rev A devices

VoIP application

SIP support

EV-DO Rev ABTS

Internet

BSC PDSN

R-P Interface

AAA

SIPRegistrar

EdgeRouterVoIP App A12

QoSDomain

MGW

PSTNSS7

SIPProxy

LocationServer



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Required RAN Features for VoIPRequired RAN Features for VoIP

1. PDSN sends IP packets to BSC without PPP encapsulation Eliminates PPP overhead and allows for efficient air interface header compression

Improves packet dropping efficiency and allows for seamless make-before-breakhandoffs between RANs

One A10 between PDSN and RAN carrying PPP traffic for control functions

One or more A10s between the PDSN and RAN carrying IP traffic for differentQoS flows

New service option introduced allowing the PDSN to send IP packets to the RAN

IS-835-D in ballot resolution, Expected publication: 1Q 2005

IPFlow

CDMAFlow

CDMAFlow

IPFlow

IPFlow

IPFlow

CDMAFlow

QoS 1 QoS 2 QoS 3

ROHCChannel

ROHCChannel

IPFlow

ROHCChannelCDMA

Flow

IPFlow

CDMA

Flow

IPFlow

PPP in HDLC - like Framing

Header / PayloadCompression

IPFlow

PDSN

RAN

Existing architecture Proposed architecture



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QU CO O O O S Ua c 005 o O e e

Why Air Interface Header Compression?Why Air Interface Header Compression?

First, header compression needed to reduce RTP/UDP/IP header overheadof voice frames RTP Header: 12 bytes

UDP Header: 8 bytes

IP Header: 20 bytes (IPv4) or 40 bytes (IPv6) Full rate frame size is only 22 bytes. Overhead even more dominant for 1/2 and 1/4

rates

Alternatively, frame bundling could lower overhead but leads to increased

delays and burst error rates

A robust and efficient header compression scheme requires state fulloperation with error recovery and acknowledgement mechanisms

RAN based RoHC allows for efficient implementation Compression algorithm can take advantage of air interface error and timing recovery

Packet dropping based on air interface information done prior to header compression

Simplified connected-state BSC-BSC handoff with two independents ROHC contexts



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Header Compression System View (mobile to mobile)Header Compression System View (mobile to mobile)

Downlink HeaderCompressor

Uplink HeaderDecompressor CoreCore

VoIPVoIP

NetworkNetwork

Uplink Header

Compressor

RL

Downlink Header

Decompressor

FL

RANRAN RANRAN

AT_2AT_2AT_1AT_1

Typical RoHC header sizes with timer-basedcompression mode

2 or 4 bytes on IPv4 (UDP checksum enabled or disabled)

4 bytes on IPv6 (UDP checksum must be enabled)



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Required RAN Features for VoIP (contd)Required RAN Features for VoIP (contd)

2. QoS configuration and activation using the Enhanced Multi-Flow PacketApplication (EMFPA) Protocol

Streamlined process to minimize setup delays while consuming no resources untilactivation

TIA-1054 published in January 2005. Requires appropriate QoS support in IS-835-D,IS-878-A and IS-1878-A.

Configuration After session and PPP establishment upon power-up, VoIP application (AT) sends

reservation request to setup the appropriate QoS

After authorization, AN initiates QoS configuration

Upon successful configuration, AT instantiates the required IP filters at the PDSN Filters are classifiers based on source/destination IP addresses, Port numbers, etc.



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Registration / QoS ConfigurationRegistration / QoS Configuration

AT RAN PDSN AAA

SIP

Reg

Powerup

Session establishmentPPP carrying A10 setup

and Sub QoS ProfileAuthentication andSub QoS Profile

Appstarts

EMFPA: AttributeUpdateRequest

(ReservationKKQoSRequest for audio, control)

Authorization wrt Sub QoS Profile

EMFPA: AttributeUpdateAccept

EMFPA: AttributeUpdateRequest(Link Flow Configuration)


Set up A10s for carryingcontrol, media traffic

Resv (TFT)

ResvConfirm

SIP Registration

RTCMAC SubType 3: AttributeUpdateRequest(MAC Configuration)

RTCMAC SubType 3: AttributeUpdateAccept

Could bePart of

SessionEstablishment, repeated

for eachcodec

EMFPA: AttributeUpdateRequest(ReservationKKQoSResponse)


RAN performs authorization bychecking requested ProfileIDs

wrt subscriber QoS profiledownloaded from AAA

FL IP filters sent to PDSN.ReservationLabel binds to

A10 and RLP flow

VoIP app ready to place orreceive calls

Access to QoS for VoIP calls can be restricted to authorized subscribers and valid applications



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VoIP Call SetupVoIP Call Setup

QoS activation using EMFPA Protocol (TIA-1054)

AT requests the audio flow to be activated by sending a ReservationOnRequest

Control flows (SIP signaling) may be activated by default since they consume

minimal resources AN performs admission control by checking BS available resources

If successful, AN configures FL scheduler resources and RTCMAC resources if notalready done

AN sends ReservationAccept to AT. SIP call setup starts. SIP signaling used to setup and tear down VoIP calls

SIP compression may be used to reduce amount of signaling overhead and reducelatency

RAN sendsGrantedQoSindication to PDSN upon allocation and de-allocation of resources to a VoIP call

Can be used to generate AAA records allowing for time-based billing

If one A10 is dedicated to a VoIP flow, byte accounting may also be generated as perIS-835-D



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Call Setup(same QoSProfileID at the end points)Call Setup(same QoSProfileID at the end points)

AT1 RAN PDSNSIP

Proxy

AT2

Call

InitatedRUP: ConnectionRequest

EMFPA: ReservationOnRequest

(Default Codec)

Admission Control

EMFPA: ReservationAccept

SIP: Invite (Codec List)SIP: Invite (Codec List)

SIP: 100 Trying

Connection Setup

PDSN RAN

Connection Setup

SIP: Invite (Codec List)

Granted QoS

EMFPA:ReservationOnRequest

(Default Codec)

EMFPA:

ReservationAccept

Admission Control

Granted QoS

Ring User

SIP: 180 RingingSIP: 180 Ringing

User Picks

SIP: 200 OKSIP: 200 OK

SIP: ACK

Media Traffic

SIP: 100 Trying

SIP: PRACK

SIP: PRACK

Call SetupDelay

Call Setup Time = 2 * Conn Setup + RTD + QoS Activation



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Voice over Wireless IP:Speech quality enhancing techniques

Voice over Wireless IP:Speech quality enhancing techniques



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Speech Basics: Talk spurts and silence intervalsSpeech Basics: Talk spurts and silence intervals

Talk spurtsTalk spurts are produced when the user isare produced when the user is

speaking and the vocoder uses framesspeaking and the vocoder uses frames

different than 1/8 rate frame fordifferent than 1/8 rate frame for

communication of the speechcommunication of the speech

Silence intervalsSilence intervals are produced when theare produced when the

user goes silence and the speech vocoderuser goes silence and the speech vocoder

uses only 1/8 rate frames for communicationuses only 1/8 rate frames for communication

of the background noiseof the background noise

In a packet-switched network capacity can be improved by reducing theamount of frames used for background noise information



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Mouth-to-Ear Delay(ITU-T G.114)

285ms285ms

Maximum acceptable end-to-end delay of 285ms for user satisfaction

Commercial 3G 1x networks have 260-270ms delay

Packet-switched networks allow for capacity-delay tradeoff optimization



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Voice Frame Delay JitterVoice Frame Delay Jitter

The decoder ideally receives and plays one voice frame every 20ms However, the voice frame inter-arrival time is not constant. The variance in

the inter-arrival time is know asdelay jitter.

A de-jitter buffer is needed to allow evenly delivery of voice frames todecoder

0

50

100

150

200

250

300

350

400

1 101 201 301 401 501 601 701 801 901 1001 1101

Frame Count

Voice

FrameDelay(ms)

End2end Packet Delay

Running Average (20)

MAX Delay (20)

MIN Delay (20)

Same jitter, DifferentSame jitter, Different

End2End DelayEnd2End Delay

Different jitter,Different jitter,

DifferentDifferent

End2End DelayEnd2End Delay Sources of jitter in a 1xEV-DO

network Reverse link MAC

Forward link scheduler

RF channel conditions

Handoff

Core network



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Voice Quality Enhancing Features of VoIP over EV-DOVoice Quality Enhancing Features of VoIP over EV-DO

De-jitterBuffer

SmartNoise

DecoderTime

Warping

Encoder

LowerLayers

De-jitterBuffer

SmartNoise

DecoderTime

Warping

SmartBlanking

LowerLayers

1xEV-DO Rev A

Encoder

SmartBlanking

Smart blanking

Smart background noise

Adaptive de-jitter buffer

Speech time warping

Enhanced error concealment



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Smart Blanking / Background NoiseSmart Blanking / Background Noise

Current vocoders introduce unnecessary overhead by using a sequence of1/8 rate frames for background noise information during silence periods

Even with RoHC, IP Packet overhead is twice the size of the payload

A single prototype 1/8 rate frame can be repeatedly played back if it hassimilar statistical properties as the background noise

Transmitter (Smart Blanking)

Transmits the first 1/8 rate frame after the end of a talk spurt the prototype frame If background noise changes significantly during silence, sends a new 1/8 rate

prototype frame

Receiver (Smart Background Noise)

Transitions to Silence state when a 1/8 rate frame is received (or time out). When insilence state, playback prototype 1/8 rate frame

If a 1/8 rate frame is received during silence state, update prototype 1/8 rate frame

To avoid flatness of reconstructed background noise, erasures are puncture withsome probability



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Prototype 1/8 Rate Frame UpdatePrototype 1/8 Rate Frame Update

Sometimes, first 1/8 frames during silence interval are not representativeof background noise

Updates triggered by noise energy change

Updates triggered by noise frequency content change

Adaptive update algorithms optimize tradeoff between 1/8 rate overheadand noise quality reconstruction

Example: Noise from a rack of computers

Beginning of several silence periods shown

-20

-10

0

10

20

30

40

50

1 2 3 4 5 6 7 8 9 10

Frame number

Frameene

rgydeltawith

respecttoaverage,

dB

Few first frames do notFew first frames do not

represent the averagerepresent the average

background noisebackground noise



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Speech SamplesSpeech Samples

Harvard sentences with background noise generated by anelectric motor

Original recording

EVRC (Silence periods only)

Sending one 1/8 rate frame (Silence periods only)

Smart Blanking (Silence periods only)

Counting with background noise generated by a Rack ofcomputers

Original recording

EVRC

Sending one 1/8 rate frame

Smart Blanking



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Adaptive De-jitter BufferAdaptive De-jitter Buffer

The de-jitter buffer is an adaptive buffer that restores constant inter-packettiming prior to decoding

The required amount of buffer changes dynamically for each talkspurt. Itrepresents a tradeoff between additional end-to-end delay and frameerasure rate

20ms DeDe--jitter bufferjitter buffer DecoderDecoder

Traditional deTraditional de--jitter/decoder implementationjitter/decoder implementation

Even deliveryEven delivery

of voice framesof voice frames20ms of voice20ms of voice

per frameper frame

20ms

InIn--order delivery not guaranteedorder delivery not guaranteed

Variable Inter-arrival times

If there are too many frames in the buffer, end-to-

end delay is increased

If there are no frames in the buffer, an erasure hasto be fed to the decoder



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Adaptive De-jitter Buffer with Time Warping OperationAdaptive De-jitter Buffer with Time Warping Operation

Time Warping adjusts the playback duration of a speech segmentwithoutchanging its pitch

Time expansion or compression allows the de-jitter buffer size changeduring a talk spurt

Use of Speech Time Warping greatly enhances the system ability to trackvariable delay and jitter, reducing the overall latency of the de-jitteroperation for same target PER

20ms DeDe--jitterjitterbufferbuffer

Decoder & TWDecoder & TW

DeDe--jitter/decoder/time warping implementationjitter/decoder/time warping implementation

UnUn--even delivery ofeven delivery of

voice framesvoice frames

1010--35ms of voice35ms of voice

per frameper frame

(when required)(when required)

20ms

InIn--order delivery not guaranteedorder delivery not guaranteed

Variable Inter-arrival times

If there are too many frames in the buffer, playback

time reduces

If there are few frames in the buffer, playback timeincreases



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Time Warping OperationTime Warping Operation

In order to preserve pitch during time-warping, long-term similarities ofthe human speech must be found among samples

Maximum auto-correlation algorithm determines a set of suitablesamples to execute time warping

Small speech segments are merged to achieve compression or

repeated to achieve expansion

1 pitch1 pitch


Addi i l F f D ji B ff Ti W i


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Additional Features of De-jitter Buffer/Time WarpingAdditional Features of De-jitter Buffer/Time Warping

Playback time of buffered frames can be expanded prior to certainknown periods of air-link unavailability

Forward link handoffs

Reverse link silence periods

One wayOne way

delaydelaySpeechSpeechsegmentsegment

User1User1 User2User2

ConversationalConversational

RTDRTD

ConversatioConversatio

RTDRTD

Reduction of conversational roundtripdelay (RTD)

The perceived RTD is defined by the delayof the first and last packet of speechsegments

First packets in a talkspurt can be time-

expanded when played back No need to wait until de-jitter buffer fills up

with the target number of frames

Last packets in a talkspurt can be time-

compressed when played back


E h d E C lE h d E C l


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Enhanced Erasure ConcealmentEnhanced Erasure Concealment

Vocoders designed for circuit-switched networks perform extrapolation oferased packets purely based on the last packet

For packet-switched networks, where a de-jitter buffer is present, future

packets may be available for improved voice quality

Simulations show that future frame is available for enhanced concealmentapproximately 40% of the time

A good portion of the lost information can be linearly interpolated with

increased accuracy

Pitch estimation

Minimal dampening or fading of the speech during erasure concealment can be achieved

GoodGoodErasureErasureGoodGood

Playback Time linePlayback Time line

CircuitCircuit

VoIPVoIP



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Capacity Simulation ResultsCapacity Simulation Results


Si l ti F kSi l ti F k


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Simulation FrameworkSimulation Framework

QUALCOMM developed a complete network simulation model representing 19

cells / 57 sectors

The model follows the 3GPP2 Simulation Strawman with minor modifications to

closely match practical implementations

All EV-DO rev A terminals are assumed to have dual receive diversity

Voice frames are simulated from source to destination including all possible

impairments

Capacity is determined based on collected statistics: mouth-to-ear delay, frame

error rate (FER), rise-over-thermal (ROT), etc.


M d l A tiModel Assumptions


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Model AssumptionsModel Assumptions

Source traffic is modeled based on MSO model IS-871. Full rate: 29%, rate: 4%, rate: 7%, 1/8 rate: 6%, Blanked: 54%

There is exactly one vocoder frame per IP packet (no bundling)

Packet overheads:

4 bytes of RTP/UDP/IP overhead is assumed with RoHC

2 bytes of RLP and Stream layers

4 bytes of MAC trail, FCS and tail bits Constant delay components:

Circuit-to-DO or

DO-to-Circuit**DO to DO

35ms 35ms

5ms 10ms

10ms 20ms

15ms 15ms

3ms 3ms

68ms 83ms

* Additional propogation delay can be added as 0.005ms/km for long distance calls [TIA/EIA TSB116]

** Additional delay is assumed for circuit network depending on characteristics (e.g., 1xRTT to DO)

Core IP Network or PSTN *

Voice frame decoding (not including de-jitter/time warper)

TOTAL

Delay component

Vocoder (alg., proc.)

Packet Processing (turbo encod, demod, decod, MAC)

RAN (BTS-PDSN)


VoIP Capacity of 1xEV-DO rev AVoIP Capacity of 1xEV-DO rev A


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VoIP Capacity of 1xEV-DO rev AVoIP Capacity of 1xEV-DO rev A

System capacity of 48 VoIP users per

sector can be achieved given maximum

reverse link load criterion

Reverse link ROT should not exceed 7dB by

more than 1% of time

Approximately 89 MAC channels are

allocated per sector

Similar capacity figure could be achieved by 1xRTT circuit-switched voice if dual

diversity handsets and smart blanking were employed


Mouth-to-Ear Delay Statistics (mobile to mobile)Mouth-to-Ear Delay Statistics (mobile to mobile)


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Mouth-to-Ear Delay Statistics (mobile to mobile)Mouth-to-Ear Delay Statistics (mobile to mobile)

* 3G1x delay values are based on field measurements from commerc* 3G1x delay values are based on field measurements from commercial 1xRTT networksial 1xRTT networks

Average end to end delay is approximately 100ms lower than in existing 1xRTTdeployments

Even at capacity, 99% of packet delays are better than typical circuit-switched

delay


Mouth-to-Ear Delay Statistics (2)Mouth-to-Ear Delay Statistics (2)


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Land to DOLand to DO

1xRTT to DO wt. tandem (+133ms)1xRTT to DO wt. tandem (+133ms)

1xRTT to DO w/o1xRTT to DO w/o

Tandem (+95ms)Tandem (+95ms)

Mouth-to-Ear Delay Statistics (2)Mouth-to-Ear Delay Statistics (2)

DO to LandDO to Land

DO to 1xRTT wt. tandem (+133ms)DO to 1xRTT wt. tandem (+133ms)

DO to 1xRTT w/oDO to 1xRTT w/o

Tandem (+95ms)Tandem (+95ms)

End-to-end delays of land to DO calls arewithin ITUsVery Satisfiedrank (


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Capacity Enhancements: Pilot Interference CancellationCapacity Enhancements: Pilot Interference Cancellation

A significant portion of reverse link interference is due to Pilot signal transmissionswhen a large number of VoIP users are present

Hence VoIP capacity can be improved by removing this pilot interference at the base

station

QUALCOMM is offering Pilot InterferenceCancellation (PIC) as an optional FPGA

feature supported by the CSM6800 solution

System capacity of 58 VoIP users per

sector can be achieved for 1%@7dB RoT

Approximately 100 MAC channels areallocated per sector

PIC increases VoIP capacity by 20%


Mouth to Ear Delay Statistics with PICMouth to Ear Delay Statistics with PIC


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Mouth-to-Ear Delay Statistics with PICMouth-to-Ear Delay Statistics with PIC


Summary of VoIP Capacity over EV-DO Rev ASummary of VoIP Capacity over EV-DO Rev A


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Summary of VoIP Capacity over EV-DO Rev. ASummary of VoIP Capacity over EV-DO Rev. A

1xEV-DO rev A with dual-antenna handsets provides a competitivealternative for wireless VoIP services


Mixed VoIP and Platinum MulticastMixed VoIP and Platinum Multicast


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Mixed VoIP and Platinum MulticastMixed VoIP and Platinum Multicast

Platinum Channel at 187kbps 1/8 interlace

Platinum Channel at 328kbps 1/4 interlace (minus control channe

Excess Forward Link Capacity Allows Operators to Introduce

Multicast with Minimal End-to-End Delay Impact


Mixed VoIP and Platinum Multicast (2)Mixed VoIP and Platinum Multicast (2)


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Mixed VoIP and Platinum Multicast (2)Mixed VoIP and Platinum Multicast (2)

Even at Voice Capacity, 95 Percentile End-to-End Delays is Below 300ms

Additional Multicast Channel Capacity Can Be Available During Non Busy Hours

Every DO rev A Carrier Used for VoIP Can Provide Free Multicast Capacity


Future Simulation WorkFuture Simulation Work


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Future Simulation WorkFuture Simulation Work

Evaluate capacity vs. delay tradeoffs for the following scenarios

Mixed VoIP and Best Effort users

Mixed VoIP, BE and Multicast

New update in 4-6 weeks



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Thank You!Thank You!



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Backup SlidesBackup Slides


Reverse Link ModelingReverse Link Modeling


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gg

Reverse Link is operated in a manner optimized for voice service

12-slot termination mode (i.e., 1% PER target after 3 sub-packets)

For VoIP flow, each IP packet is sent in one MAC packet when power headroom allows (max AT TX power

is 23dBm). MAC layer does not limit data rate

No retransmission of failed packets (RLP disabled)

DRC length of 8 slots, independent of handoff state

The AT does not drop any packets due to delay

Overhead and Feedback Channels

Pilot, DRC, DSC, RRI, ACK, overheads are modeled

RAB, RPC, H-ARQ channel performance also modeled

Parameters:

-6dB DRC-to-pilot

-9dB / -15dB DSC-to-pilot (sho / no sho)

-6dB RRI-to-pilot ACK channel load on RL is modeled as constant load


Forward Link ModelingForward Link Modeling


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gg

Each IP packet gets a timestamp at the access network (AN) and scheduled on FL

Multi-user (MU) scheduler minimizes packet delay while increasing link utilization

Uses both single-user and multi-user packets as appropriate

Multi-user packets can carry data for up to 8 different users

Uses the packet arrival timestamp (no knowledge of prior traffic delay)

MU scheduler may drop some packets that have experienced excessive delay (e.g., 200ms)

at the FL transmission queue

No retransmission of failed packets (RLP disabled)

H-ARQ and D-ARQ* are modeled

The DRC and ACK channels are modeled

DRC and ACK errors and erasures are taken into account

DRC erasure mapping algorithm is implemented

76.8kbps Control Channel and other packet overheads are modeled


Simulation Configurations (mobile to mobile)Simulation Configurations (mobile to mobile)


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DO to DODO to DO

DODO

RANRANCore IPCore IP

Ntwk.Ntwk.DODO

RANRANDO ATDO AT

DeDe--jitterDO ATDO AT

Encoder jitter

Circuit to DO (e.g., 1xRTT to DO)Circuit to DO (e.g., 1xRTT to DO)

CircuitCircuit

PhonePhone

PSTN*PSTN* DO ATDO AT

DeDe--jitterjitter

DO ATDO AT

EncoderEncoder

DO RANDO RAN

DeDe--jitterjitter

DO to Circuit (e.g., DO to 1xRTT)DO to Circuit (e.g., DO to 1xRTT)

CircuitCircuit

PhonePhone

CircuitCircuit

RANRAN

CircuitCircuit

RANRAN

Encoder

DODO

RANRAN

PSTN*PSTN*

* PSTN and tandem encoding may not be required if both circuit and VoIP networks usesame vocoder (e.g., EVRC).


Simulation Configurations (land to/from mobile)Simulation Configurations (land to/from mobile)


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Circuit to DO (e.g., POTS to DO)

Core IPCore IP

Ntwk.Ntwk.DODO

RANRANDO ATDO AT

DeDe--jitterjitter

Land VoIP to DO (e.g., PC client to DO)Land VoIP to DO (e.g., PC client to DO)

CircuitCircuit

PhonePhone

Circuit to DO (e.g., POTS to DO)

PSTNPSTN DO ATDO AT

DeDe--jitterjitter

DO ATDO AT

EncoderEncoder

DO RANDO RAN

DeDe--jitterjitter

DO to Circuit (e.g., DO to POTS)DO to Circuit (e.g., DO to POTS)

CircuitCircuit

PhonePhone

Land VoIPLand VoIP

PhonePhone

DODO

RANRAN

PSTNPSTN

DO to Land VoIP (e.g., DO to PC client)

DO ATDO AT

EncoderEncoder

DO RANDO RAN

DO to Land VoIP (e.g., DO to PC client)

Land VoIPLand VoIP

PhonePhone

DeDe--JitterJitter

Core IPCore IP

Ntwk.Ntwk.

VoIP Overview 032105

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