End to End QoS for

Voice Services in NGNs

Ian Jenkins
Chief Voice Technologist
BT Group CTO
 Why voice? What about other services.
• Voice has high QoS demands
– Existing service benchmark
– Exemplifies the characteristics of other interactive service
• Similar problems to other ‘conversational’ services. e.g. video telephony

• Special issues with inter-working with legacy networks.

• Voice Services are dominant
– Voice services volumes are increasing.
– Revenues declining but still enormous ($17T worldwide).

• Data services generally anticipate best efforts service

• One way streaming services
– Less onerous delay requirements.

Ian Jenkins – Chief Voice Technologist

 Session Overview

• What do we mean by QoS for VoIP services?

• What classes of voice service uses VoIP technology?
• What affects voice quality?
• Mechanisms for improving voice quality
• The problems of achieving end to end voice QoS
• IP interconnect for different voice services.

Ian Jenkins – Chief Voice Technologist

 What do we mean by QoS for VoIP
services?
• IP Engineers:-
– Diff Serv packet marking
– Traffic Engineered MPLS LSPs.

• Network Architects:-
– Bandwidth allocation and admission control

• VoIP engineers:-
– Mean Opinion Scores

• Customer Service Centre:-

– Service Availability and Consistency

• Product Managers:-
– Service Wrap

Ian Jenkins – Chief Voice Technologist

 So what is QoS?

• Simple answer
– The ‘S’ of QoS is “Service” – “Service” is all of these.

• However, session focus is on those aspects that have

particular implications for VoIP
– What affects voice quality?
– What gives service consistency?

• But first we need to define what Service we want.

– Voice and VoIP do not define a service.

Ian Jenkins – Chief Voice Technologist

 What classes of voice service uses VoIP
technology? ….. Internet Telephony

• Best efforts transport

• No Service Provider control over network design /
resources
• Low bit rate codecs
– G.723 6.3 kbps
– G.726 16,24,32 kbps

• Audio range 3.4Khz

• Inter-working with other voice services via tradition
PSTN TDM interconnect

Ian Jenkins – Chief Voice Technologist

 Enterprise Telephony

• Control of own network -> engineered for service mix,

– Bandwidth provision
– Packet Marking
– Network Testing

• Low bit rate codec (Typically G.729a)

• Audio range 3.4Khz
• Inter-working with other services via PSTN TDM IX

Ian Jenkins – Chief Voice Technologist

 Mobile Telephony
• 3G ATM packet access now, IP Core 1-2 years
• Control of own network -> engineered for service mix
• Bandwidth management of radio and backhaul links
• Bandwidth management / Overprovision of IP core
– Economics
– Ensured quality during loss of network capacity

• Low bit rate codec

– GSM HR, FR, EFR (5.1->13 Kbps)
– 3G AMR (8->12Kbps)

• Audio range <3.4Khz,

• Inter-working with other services via
– PSTN & other Mobile (TDM / IP)
– 3G GSMA GPRS Roaming Network

Ian Jenkins – Chief Voice Technologist

 PSTN Telephony
• Control of own network -> engineered for wide range
of traffic patterns (dial up internet to televotes)
• Bandwidth management on access, interconnect and
backhaul links to IP Core to ensure IP links are not
over committed.
• Bandwidth managed core / Overprovision IP Core
– Economics
– Maintain quality on loss of network capacity

• 64k bit rate G.711 codec

• Audio range 3.4Khz
• Inter-working with other services via PSTN IX (TDM
and IP).
Ian Jenkins – Chief Voice Technologist
 ISDN Telephony
• Control of own network -> engineered for service mix
• Require very low bit error rate (packet loss) ~ 1 in 106
• Bandwidth management on access, interconnect and
backhaul links to IP Core to ensure IP links are not
over committed.
• Bandwidth managed core / Overprovision IP Core
– Economics
– Maintain quality on loss of network capacity

• 64k bit rate G.711 or G.722 codec

• Audio range 3.4Khz or 7Khz
• Inter-working with other services via PSTN/ISDN IX
(TDM and IP).
Ian Jenkins – Chief Voice Technologist
 Wideband VoIP
• High bandwidth IP access to the terminal
– Broadband / WiFI / WiMAx / Campus LAN Access
• End to end IP connection
– If not E2E IP or lower capability called terminal, then negotiate a lower rate
and inter-work with different codec.
• 24-32kps wideband codecs
– AMR-WB (G.722.2)
– iPCM
– BV-32K
– Others
• Audio range 7Khz
• Inter-working with other voice services via PSTN but at G.711
not wideband.

Ian Jenkins – Chief Voice Technologist

 What classes of voice service uses
VoIP technology?

All with different service characteristics,

user needs and user expectations.
Ian Jenkins – Chief Voice Technologist
 What affects voice quality?

• Basic signal-to-noise ratio

• Impairments which occur simultaneously with
voice signal
• Impairments caused by delay ***
• Distortion Impairment ***

– *** Particular interest to VoIP solutions.

Ian Jenkins – Chief Voice Technologist

 What affects voice quality?
- The E-Model
Objective network parameters
Impairments which Impairments
Basic signal-to-noise occur caused by Distortion Expectation
ratio simultaneously with delay Impairment Factor
voice signal

R = R0 - I s - I d - I e + A

Rating Factor [0,100]

Prediction of User Perception:-
Mean Opinion Score
Relative Quality (better / worse)
Premature Termination
Ian Jenkins – Chief Voice Technologist
 What affects voice quality?
Distortion Impairment - Ie
• Speech Coding Algorithm
– Compression of transmitted bit rate

• Transcoding
– Conversion from one codec to another

• Silence suppression
– Clipping of voice content

• Note:- All these can also add to delay

Ian Jenkins – Chief Voice Technologist

 Distortion Impairment - Ie
Speech Coding / Codec Type
codec bit intrinsic
origin standard type Ie
rate (kb/s) quality R
G.711 PCM 64 0 94.3 PSTN
G.726, G.727 ADPCM 16 50 44.3
24 25 69.3
32 7 87.3
40 2 92.3
ITU-T
G.728 LD-CELP 12.8 20 74.3
16 7 87.3
G.729(A) CS-ACELP 8 10 84.3 Corporate
G.723.1 ACELP 5.3 19 75.3
MP-MLQ 6.3 15 79.3 Internet
GSM-FR RPE-LTP 13 20 74.3
ETSI GSM-HR VSELP 5.6 23 71.3
GSM-EFR ACELP 12.2 5 89.3
Mobile

Ian Jenkins – Chief Voice Technologist

 What affects voice quality?
Mouth to Ear Delay - Id
• Echo
– Acoustic echo
– 2/4 Hybrid in PSTN wire access and analogue telephones

• Talker echo
Far End (acoustic and/or 2/4w)

• Listener echo
Far End

• Impaired interaction
• Good IP terminals and echo cancellation techniques
make impaired interaction the more important issue.

Ian Jenkins – Chief Voice Technologist

 Effect of Delay on Voice Quality
• ITU-T Recommendations G.114 and G.131 specify the following
tolerable mouth-to-ear delay bounds for undistorted (analog or
G.711@64 kb/s) voice with (hybrid or acoustic) echo
echo control
needed

acceptable conditionally acceptable unacceptable

mouth-to-ear
delay
25 ms 150 ms 400 ms

Conversation Interactive Impairment Increases

Ian Jenkins – Chief Voice Technologist

 So what is good anyway?
ETSI TIPSPAN definitions
Class 4 (Best) 3 (High) 2 (Medium) 1 (Best Effort)

Listener Speech
Equivalent or Equivalent or
Quality Better than Undefined
better than better than
G.711 G.726 at 32 GSM-FR
(One-way Non-
conversational) kbit/s

End-to-end < 100ms < 100ms < 150ms < 400ms

Delay (G.114)

Overall
Transmission
> 90 > 80 > 70 > 60
Quality Rating (R)

Corp-
PSTN GSM Internet
orate
Ian Jenkins – Chief Voice Technologist
 Mechanisms for improving voice quality
• Lossy networks
– Packet loss concealment
– Forward correcting codecs
• Limited bandwidth
– Low bit rate codecs
– If breaking out to PSTN then compressing codecs adds to delay and
introduces some impairment if transcoding to other than G.711
• Jitter
– Packet prioritisation
– Data Packet fragmentation on low bandwidth links
– Bandwidth fill of links.
– MPLS
• Bandwidth management
– E-2-E bandwidth allocation for VoIP packets
– Admission control prevents over commitment of bandwidth resulting
in increased packet loss, jitter and queuing delay

Ian Jenkins – Chief Voice Technologist

 The problems of achieving end to end
voice QoS
• VoIP services have grown up either as:-
– Voice applications independent of the underlying network e.g. Skype /
Vonage
or
– VoIP islands that are engineered for quality but rely on the low delay
characteristics of TDM networks outside. E.g. corporate VoIP, VoIP trunking,
international VoIP carriers.
with
– PSTN as the low delay, clearing house for VoIP connectivity.
– No overall transmission planning.

• But when PSTNs start using VoIP there will no longer be a low
delay public network
– Mixing services will have different quality effects.

Ian Jenkins – Chief Voice Technologist

 Global Delay Plan for PSTN Quality

Global
Xmissn /
Switching
Country 1 Country 2

<400ms
<150ms <100ms <150ms

Ian Jenkins – Chief Voice Technologist

 Country Transmission plan for PSTN
Quality Telephony rk
t w o
Country 1 n e I P
p r
e sing
Carrier 2 ms u
3
(IP) 5 at o r
= > er
Carrier 1 rd
a o p Carrier 6
a n d
(TDM) Carrier 3
IP
St (TDM)

UK Carrier 4
Carrier 5
(IP)

<150ms
Ian Jenkins – Chief Voice Technologist
 Main Contributors to Delay in VoIP
Networks
• TDM to IP conversions -> 25-80ms
– Packetisation delays
– Processing delays
• Low bit-rate codecs -> G.729 ~15ms
• Error correcting codecs
– for ‘lossy’ networks
– GSM EFR -> 20ms
– G.723 -> 37.5ms
• Silence Suppression -> 50ms
• Poor jitter and associated large jitter buffers
– 2-3ms -> 100ms
• Packet Loss Concealment
– 10+ms

Ian Jenkins – Chief Voice Technologist

 Example:- Meeting the UK 35ms Budget

• Codec G.711 -> 64Kbps (125µsec)

• Delay in media gateways
– Packetisation of 10ms
• 1:2 Packet header to payload ratio i.e. increases the bandwidth.
– (20ms packetisation give 1:4 Packet header to payload ratio)
– Processing and queuing <15ms
• Low loss networks
– No need for packet loss concealment
– Good for support of ISDN services
• Low jitter design
– Minimise jitter buffers -> 2-3ms

Ian Jenkins – Chief Voice Technologist

 IP Interconnect for different voice
services
• PSTN is often used as the fall-back method for
service connectivity.
– Exploits the very low delay characteristics of TDM PSTN.
– What happens when we have PSTNoIP?

• PSTNoIP must be low delay to work with legacy

PSTN network and preserve PSTN quality.
• Can we mix all VoIP services up indiscriminately,
high delay with low delay? Does this destroy Service
differentiation?
• How is inter-service connectivity provided without
eroding QoS differentiation?

Ian Jenkins – Chief Voice Technologist

 IP Interconnect:- What helps?
• Optimise routing through networks to minimise
IP/TDM conversions
– No. Portability ‘Call Forwarding’ Solutions
– Number translation services (Freephone / Personal Numbering)
– Enum technology may help, e.g. UK PSTN to USA internet voice use
early breakout based on dialled number look-up

• If IP end-to-end, support codec negotiation.

– Avoid transcoding
– E.g. Transit networks between mobile operators
(remove EFR->G.711->EFR conversions)

• VoIP end to end……

….. but when will that be across the world?

Ian Jenkins – Chief Voice Technologist

 Thank you

Ian Jenkins – Chief Voice Technologist