Overcoming VoIP Quality Challenges

Dr. Jan Linden, VP of Engineering

Global IP Solutions

Outline

VoIP Quality Challenges

Latency

Codec Choice

Conferencing

How to Measure Speech Quality

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QualityCost

VoIP Design Considerations

Speech Quality

Time to Market

Flexibility

Ease of Use

Network Impairments

Power Consumption

Cost

Signaling

Features

Infrastructure

Device Considerations

VoIP DesignChallenges

Coping with Network Degredation

Power Consumption

Hardware Issues (Processor, OS, Acoustics, etc.)

Echo Cancellation

Additional Voice Processing Components

Environment – Background Noise,

Room Acoustics, etc.

Speech Codec

Both Sides of the Call Need to be Considered

Network

Codec

Hardware

EchoPower

Voice Environment

Major Challenges for VoIP End-point Design

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DelayMajor effect is “stepping on each other’s talk”Usage scenario affects annoyance factor – higher delay can be tolerated for mobile devicesLong delays make echo more annoying

Packet LossSmooth concealment

necessary

Network JitterJitter buffer necessary to ensure continuous playoutTrade-off between delay and quality

Impact of IP Networks

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Sources of Latency

Codec Capture Playout Network delay Jitter buffer OS interaction Transcoding

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A/DPre-

processing

Speech encoding

IP interface

D/APost-

processing

Speech decoding

Jitter buffer

IP Network

A/DPre-

ProcessingSpeech Encoding

IP Interface

D/APost-

ProcessingSpeech

DecodingJitterButter

IP Network

Impact of Delay on Voice Quality

ITU-T (G.114) recommends:– Less than 150 ms one-way delay for most applications (up to 400 ms

acceptable in special cases)

Users have got used to longer delays– Still, low delay very important for high quality

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2

3

4

0 250 500 750One-w ay transmission time [ms]

Mea

n O

pini

on S

core

Data from ITU-T G.114

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Speech Codec

Many conflicting parameters affect choice of codec

Determines upper limit of quality

Support of several codecs necessary

– Interoperability

– Usage scenario

IPR issues a significant concern

Speech Codec

Packet-loss Robustness

Memory

Input Signal Robutness

Sampling Rate

Complexity

Delay

Bit-rate

Quality

Complexity

Bit-rate Input Signal Robutness

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Audio Spectrum

Better than PSTN quality is achievable in VoIP

– Utilizing full 0 – 4 kHz band in narrowband

– Wideband coding offers more natural and crispier voice

Telephony band

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NarrowbandSpeech (PSTN)

Audio Spectrum vs. Speech Quality

Frequency

WidebandSpeech

Super WidebandSpeech

4 kHz 8 kHz 22.1 kHz

Speech Quality

16 kHz

CDSpeech

10 kHz

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Speech Codec Design for VoIP

Many standard codecs designed for bit errors, not packet loss

– Error propagation issue for CELP codecs

Variable bit rate attractive for IP networks Packet overhead significant (5 – 32 kb/s)

– Makes low bit rate codecs less attractive

Packet loss concealment a must Jitter buffer design has significant impact on quality Alternatives to standards

– De-facto standards like iSAC– Open source like Speex

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Echo Cancellation

High delay in VoIP makes echo problem more prominent Network/Line echo cancellation for gateways Acoustic echo cancellation

– Hands-free/speakerphone– Small devices

Biggest challenge is AEC for PC – Acoustic setup unknown and changing– Wideband speech– Very few solutions on the market

Limited quality degradation since G.711 used on the PSTN side

VoIP to PSTN

Severe quality degradation common since low bit-rate codecs typically used on both sides

VoIP to Cellular

Usually occurs in Session Border Controllers

Can normally be avoided

VoIP to VoIP

Transcoding occurs when the endpoints are using different codecs– Every transcoding introduces distortion– Low bit-rate codecs very sensitive to transcoding

Transcoding between networks

Transcoding in conferencing– Mixing done in decoded domain results in transcoding

Effects of Transcoding

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Spot Jitter Patterns - Increase Delay to Keep

Good Quality when Unavoidable

Packet Loss Concealment - Capable of Handling Several Lost

Packets in a Row

Very Quick Jitter Buffer Adaptation – Conditions Change Very Rapidly (on a milisecond basis)

Minimize Delay Everywhere – every milisecond counts

How to Make the VoIP Software Robust?

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Subjective Methods

Test the “right thing”, i.e. subjective quality

Takes all types of degradation into account

Time consuming and costly

Lack of repeatability

Objective Methods

Simple and affordable Inaccurate but repeatable results Sensitive to any processing (non-

linear filtering, echo cancellation, time warping etc.)

– Time synchronization major challenge not yet solved

Sensitive to background and equipment impairments

One step behind development of codecs and error concealment

Next generation algorithm in standardization process (P.OLQA)

Measuring Voice Quality

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Audio Conferencing Design includes a trade-off between quality and

scalability Client based or server based

– Server based offers better scalability than client based

– Can be combined

Transcoding often unavoidable Two strategies:

– Mix incoming signals to form one output signal

– Only relay packets and mix at client side

Multi-codec support– In relay mode all endpoints need to support all codecs

Narrowband and wideband– Both can be present in a conference

– Narrowband participant will hear everything in narrowband

– Wideband participant hears others in narrowband or wideband

B

E

C

D

A

B+C+D+E

A+C+D+EA+B+D+E

A+B+C+E

A+B+C+D

Conclusions

Latency has a significant impact on the perceived quality in VoIP

– Low latency, high quality (e.g. NetEQ) jitter buffer necessary

Choose the right codec for the usage scenario– Or a codec that can adapt like iSAC

Transcoding should be avoided, if possible

Significantly better quality than PSTN possible– Wideband coding

No good objective measure for speech quality exists– Always combine with subjective evaluation

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