Date post: | 28-Mar-2015 |
Category: |
Documents |
Upload: | seth-hitson |
View: | 212 times |
Download: | 0 times |
Carleton University
1February 25th, 2014
Voice over IP
Presenter: Tony Hutchinson
System Engineering Manager
February 25th, 2014 Slide 2
Voice over IPCarleton University
Biographical Information
Tony Hutchison
Expertise: VoIP and network design PBX Design, TDM, ISDN, Ethernet, PSTN (PRI, BRI and Analogue), EMC, Safety Telephony and data, TDM and PDH design
Current Position - 1998 to Present System Engineer Manager – Mitel Networks (Canada) VoIP design, PBX, Hosted (Cloud) Services, Network Design Technical interface with RnD and customer facing Sales/System Engineers
Previous Positions (UK) Telecom Sciences – SME PBX System Engineer Philips – SME ISDN PBX System Designer (for the global market) GEC – Transmission and Multiplex system (analogue and digital design)
Education Birmingham University (UK): Electronic and Computer Engineering (Hons.)
February 25th, 2014 Slide 3
Voice over IPCarleton University
Agenda
Executive Summary
History
Business Case
Services
Convergence
Infrastructure
Challenges
February 25th, 2014 Slide 4
Voice over IPCarleton University
Executive Summary
Business Case
VoIP
Toll Quality
?
Security?
Not In
tern
et
Convergence
of cabling
Moves, Adds,
Changes
Business Expansion
Enterprise/Branch office
Distributed BusinessHot DeskingResilie
ncy
Toll Bypass
Cut acrossgeographies
WorldCom
TDM Backup
Two towers NY
IP-Trunks
Cost ofdedicated
links Cost ofinfrastructure
Traffic rates
Line cardsVs L2
switches
Structurdwiring
Convergence
Voice/Data
Video
Conference
CTI
InteractivePresentation/
Passive
Services
Dial Tone
Hook Switch
Soft Phone
Teleworker
Road warrior
VPN
Voice Mail
UnifiedMessaging
PDA integration
BusinessApplications
Hotel
Security
Video/Cable
Infrastructure
SIP
Megaco
Signalling
ProprietaryFeatures
ManagedNetwork
SLA
VPN
ISP Access
MPLS
MAN
WAN
Fixed link
FrameRelayHistory
Lots Learnt
Lots
Forgotten
Signalling
Protocols
2W/4W
Echo
Sidetone
Acoustics
Ergonomics TransmissionLevels
LocalInternational
Challenges
Internet
Local
ISP
ManagedNetwork
OSI 7 Laye
rVoice
Quality
Toll
MOS
PESQ,PSQM
Maintaining
Guarantee
Bandw
idth
CODEC
G.711
G.729
G.726
WidebandVAD
VLAN
PriorityTOS
Diffserv
E911
Hybrid
Echo
Delay
Security
DOS NAT
Firewall
Encryption
PacketSize
Overhead Traffic
Delay
Jitter BufferConnectionless
PowerNetwork
impairments
Packet Loss
Culture
DataWorld Telecom
World
Training
NewStandards
Demarcation
Clock Sync
FAXIn-BandData
MODEM
POS
H.323
24/7
24/7
Agents
Presence
Real Internet
CableTV
Telephone
Numbers
GeographicIndependence
IPTDM
February 25th, 2014 Slide 5
Voice over IPCarleton University
Agenda
Executive Summary
History
Business Case
Services
Convergence
Infrastructure
Challenges
February 25th, 2014 Slide 6
Voice over IPCarleton University
History
There has been much experience learnt in 100 years
Some is so common place, it has been forgotten
With IP some of these lessons need to be re-learnt
Echo was previously just louder side-tone Added delays now affect conversation quality Network Clocks were previously well defined Data path wasn’t lossy, with potential gaps in speech
February 25th, 2014 Slide 7
Voice over IPCarleton University
Agenda
Executive Summary
History
Business Case
Services
Convergence
Infrastructure
Challenges
February 25th, 2014 Slide 8
Voice over IPCarleton University
Business Case
So why all this interest in IP? Isn’t it just another transport medium?
Yes Connectionless
Not constrained to a physical location Path between two user points is not pre-defined, can change dynamically Bandwidth is only consumed when needed
Cost Alternative The long haul carriers (e.g. AT&T) are already carrying data traffic in their large
networks (at a lower cost) So, send voice as data and pay less!
Why Now? Moore’s Law Cheaper Processing More readily available
February 25th, 2014 Slide 9
Voice over IPCarleton University
Business Case
So why deploy IP rather than TDM?
Easier and cheaper maintenance: Integration of data and voice onto one network
Consolidation of trunk access to a central SIP gateway (IP) across the business
Lower operating costs: Integration of remote offices over a common corporate data network, rather than through PSTN. Single Dial Plan.
Access from anywhere: Power users such as Teleworker and sales ‘Road Warrior’. Global Access
Lower product costs: Integration of a voice application onto a central server, e.g. voice mail, means reduced number of devices. The remote sites no longer need their own local VM.
Security, Resiliency and Availability: In NY (September 11th) the IP infrastructure kept running; the PSTN didn’t
Future applications will be data centric, e.g. “Presence” Displacement of current TDM systems and businesses
February 25th, 2014 Slide 10
Voice over IPCarleton University
Business Case
There are still reasons for both IP and TDM to live together
Legacy devices are still going to be around (for some time) and people will still use these, e.g. FAX, remote MODEM
TDM is still likely to be the connection to the PSTN Most businesses have a directory number via the PSTN. Not all have a fixed IP
address. By 2017, expected that most new PSTN core installations will be IP only
Mobile and 4G will increase VoIP uptake in the next 7-10 years.
By 2014 expected ratio is 5 mobile to 1 land-line connection Existing landlines are being bundled with IP access for “last mile” access ~50% of mobile connections by 2020 expected via IP = $345Billion! ~30% of mobile IP connections expected through Google, Facebook, Yahoo Largest growth area (mobile/smart phones) expected to be Asia Pacific
February 25th, 2014 Slide 11
Voice over IPCarleton University
Business Case
In the Business PBX space three main tiers are emerging:
Managed Hosted Centrex
Globally the uptake is increasing, with predictions beyond $16billion
That’s a big market, and competition is fierce!
Hosting offers opportunity for VoIP without local “boxes”. High growth sector, but still early adopter cycle
Wireless connections and new data modes allow IP connections to be provisioned much easier in countries where it has traditionally been difficult to provide standard telephone cables and wires.
Global Service Revenues
0
2,000
4,000
6,000
8,000
10,000
12,000
14,000
16,000
18,000
2007 2008 2009 2010 2011 2012
Year
Rev
enue
s ($
mill
ion)
IP centrex service rental
Hosted IP PBX service rental
Managed IP PBX service rental
Total service revenues
“a change that will result in a growth in deployment from 8.5 million SIP trunks in 2009 to 24.3 million trunks in 2013” - Heavy Reading IP Services Insider (US)
February 25th, 2014 Slide 12
Voice over IPCarleton University
Business Case
Largest revenue split today (Business Phones)
Americas Europe
Largest growth sectors: Latin America Eastern Europe MEA Many smaller countries just
adding IP infrastructure
Slowest growth sectors: North America Western Europe But it’s still growth!
Revenue Split 2011
NA
LA
W.Euro
E.Euro
MEA
Asia-Pac
ROW
February 25th, 2014 Slide 13
Voice over IPCarleton University
Agenda
Executive Summary
History
Business Case
Services/Content
Convergence
Infrastructure
Challenges
February 25th, 2014 Slide 14
Voice over IPCarleton University
Services/Content
What services are people looking for?
Basic hook-switch and dial tone Call handling features Advance features such as call centres, agents Remote location, e.g. Teleworker, Remote Agent Networking between sites Virtual Private Networks New features such as voice recognition Integration with current applications such as customer accounts, hotel
registration, etc. Business Process Improvements Unified Communications (UC) and mobility, including Fixed Mobile
Convergence
February 25th, 2014 Slide 15
Voice over IPCarleton University
Services/Content
Today the industry is comfortable at the level of V1 applications
Biggest features are Toll Bypass and Networking
Early adopters are now taking V2 and V3 applications
Remote workers and Applications that don’t require access to the office Remote ACD, help desks, etc “Road Warriors” - Sales Service Personnel Mobility integration Common access number for all connectionsUnified Communications: Voice, Video, Applications
Aff
ect
on
b
usin
ess
February 25th, 2014 Slide 16
Voice over IPCarleton University
Services/Content
Unified Communications (UC)
Globally Accessible E-mail, V-Mail, video and mobile services
Presence and call routing
Redirection of calls based on time, availability and caller to different end points Integration with multiple call routing applications, Microsoft, e.g. Lync™ and Active Directory
Fixed Mobile Convergence
One number - able to pick up calls at desk and mobile, or alternative number Switchover between mobile carrier and in-house Wireless LAN
ACD and call routing
Service is handled by same agent to give more personalized service Agents located globally - full language support
Speech Recognition
Redirection of calls based on user spoken words
E-Business
Workforce is distributed, and mobile. Inventory tracking, e.g. RFID tagging On phone Advertising, e.g. hotel
Business Process ImprovementBusiness Process Improvement
February 25th, 2014 Slide 17
Voice over IPCarleton University
Agenda
Executive Summary
History
Business Case
Services
Convergence
Infrastructure
Challenges
February 25th, 2014 Slide 18
Voice over IPCarleton University
Convergence
What do we mean by convergence?
Combining of different worlds Different mindsets and cultures Different set of standards Use of personal devices (Smart phone) for both business and
personal use – “Bring Your Own Device (BYOD)”
And why now?
Processing power is cheaper - Moore’s law! Phones have more power today than early PCs PCs and phones are standard desktop tools
Voice and data networks can be combined to ONE Phones can now interact directly with data devices
February 25th, 2014 Slide 19
Voice over IPCarleton University
Convergence
Convergence in the network is unseen by the user.
What does the user see at the access point?
Two line jacks into ONE?
In reality, once installed, building wiring isn’t removed
On new installations, it’s cheaper to pull too many wires, than not enough
Integration of ServicesIntegration of Services
February 25th, 2014 Slide 20
Voice over IPCarleton University
PSTN, Mobile/CellCircuit Switched
$455B
Internet, IP DataConnectionless
$18B
FR, ATM, PrivateLine
ConnectionOriented
$18B
Cable TV$75B
VoIP VoDSL
Convergence
Four main business areas are converging
Voice, TV/Video, VPN and Data
Triple Play Broadcast TV - 100% users Telephony - 100% users Internet - 40% users and up
Voice is still the biggest revenue earner
Incumbents need to grow and expand
Many Cable TV providers now offer IP connectivity, many also voice.
New IP providers: Hosted VoIP, SIP Trunks, Video on Demand
Courtesy: ATM Forum
February 25th, 2014 Slide 21
Voice over IPCarleton University
Convergence
Business A Business B
Merging of business functions to common IP network
LANLAN
Long Distance PSTNe.g. AT&T
CO,E.g. Verizon CO,
E.g. Bell
IP Network 1
SIP Trunk Gateway
SIP Trunk Gateway
SIP Trunk Gateway
Existing TDM
IP Network 2
Hosted SoftSwitch
Peer2Peer BGP Router
Existing IP
Usage
Migration
February 25th, 2014 Slide 22
Voice over IPCarleton University
Agenda
Executive Summary
History
Business Case
Services
Convergence
Infrastructure
Challenges
February 25th, 2014 Slide 23
Voice over IPCarleton University
Infrastructure
What are the building blocks of the system and how are these connected?
Common Architectures and voice media paths Signalling Protocols Network Interconnections
February 25th, 2014 Slide 24
Voice over IPCarleton University
Infrastructure
The voice media paths and switching define the type of system. Three main types are defined:
IP Enabled PBX Here a line card is simply replaced by an Ethernet card. Voice switching
is done in TDM. This is not scalable and adds unnecessary delay.
Hybrid PBX TDM and IP are handled equally, only traversing a gateway when IP and
TDM devices need to connect.
Typical in an SME/Enterprise environment
IP-PBX (Hosted including Cloud Services) All switching is done in IP. TDM connections are generally only to the
PSTN via external gateway, which may be off-site. Model used for Hosted services, both Private (e.g. single business) and
Public (e.g. Skype)
G/W
IPPhone
IPPhone
TDMPhone
TDMPhone
Hybrid
G/W
IPPhone
IPPhone
TDMPhone
TDMPhone
IP-Enabled
G/W
IPPhone
IPPhone
IP-PBX
February 25th, 2014 Slide 25
Voice over IPCarleton University
Infrastructure
Basic VoIP system building blocks
Gateway between IP and TDM Media Gateway Controller Call Control Features and Services End users
Different protocols use different names, but functions are essentially the same
Peer to Peer or Central Control?
Central is good at resolving resource conflicts Peer to peer is resilient to network failure SIP can handle both aspects
MediaGateway
MediaGatewayController
FeatureServer, e.g.Voice Mail
CallControl/MediaServer
IPPhone
IP
PSTN
IPPhone
MediaStreaming
Signalling
February 25th, 2014 Slide 26
Voice over IPCarleton University
Infrastructure
Signalling Protocols are numerous and include:
H.323 MGCP/Megaco SIP Proprietary
Why so many Signalling protocols?
Different starting perspectives of the requirements They all offer some advantage for different users Most are evolving as new features start to roll out
February 25th, 2014 Slide 27
Voice over IPCarleton University
Infrastructure
H.323
Overview specification and includes: H.225 - Signalling H.245 - Media streaming TCP/IP and RTP/UDP/IP
One of the early protocols Standards based, uses current ISDN technology, works well for
interoperability between vendors Features are basic, but well proven Well proven ground rules about interoperability Centralised call control, based on known proven techniques, call state aware Slow to evolve Difficult to scale to millions of users Central call control = single point of failure Telephone routing biased rather than at application level
February 25th, 2014 Slide 28
Voice over IPCarleton University
Infrastructure
MGCP/MEGACO
MGCP was initially a proposal to IETF for a stateless gateway protocol, it has similarities to H.323, and has the ability to evolve
Combined forces with ITU to create MEdia GAteway COntrol Similar to H.323 in content, but reduced messaging New standard and evolving Allows central and distributed call control access to a gateway Was thought to be the front runner with Enterprise business but little is heard Difficulties again in scaling from a global view. Different gateways need
different controllers which need to intercommunicate.
February 25th, 2014 Slide 29
Voice over IPCarleton University
SIP (Session Initiation Protocol), RFC2543
More Client Server based and allowing Peer to Peer interaction. Call control can be distributed End devices need to be more intelligent than simple phones Has the ability to evolve quickly, and scale to large numbers Simple protocol, but lacks certain PBX capabilities Vendor specific options provide features Inter-vendor working is usually determined through “bake-off” but improving
as more vendors implement agreed solutions Networking features low, but improving Open Standards through IETF, agreed by many established industry leaders Continual proposal of new features and extensions SIP Extensions to include “proprietary” features to make them more
mainstream
SIP is the Internet Phone signalling protocol of choice
Infrastructure
February 25th, 2014 Slide 30
Voice over IPCarleton University
Infrastructure
Business 1Business 1
Service Provider 1
Internet
Service Provider 2
Business 2
Local Network Global Network
Local Network Management , one point of contact
Global Network Management, many points of contact
Common single private address space
Mixture of local private and public address spaces with overlapped addresses
Local QoS control No Guarantee of Qos or Service Level
Limited protocols Many protocols
February 25th, 2014 Slide 31
Voice over IPCarleton University
NAT
ALG
Private IP Address Space
Public IP Address Space
Infrastructure
Firewalls Used to keep out unwanted access Restricts flow of data both ways, including voice
Network Address Translation (NAT) Maps many internal private addresses to limited number of public IP addresses NAT is typically not application aware VoIP media and signalling may include private IP addresses in messages which will be confusing externally in public IP space
Application Level Gateway (ALG) Stateful and knowledgeable of protocol, e.g. SIP Can translate private/public addresses within messages
NAT and IPv6
NAT and ALG will not be needed Any device can access any other device in both public and private address space Truly global access- one large address space
February 25th, 2014 Slide 32
Voice over IPCarleton University
VoIP
Infrastructure
Carrier/SP
PSTN
LAN
LAN
SIP Trunk Gateway
Internet
SIP ALGLAN
Carrier2
Border Gateway
Architecture of SIP in a large carrier deployment
• SIP ALG provides IPv4 NAT and firewall functions for SIP (a.k.a. Session Border Controller (SBC))
Hosted
SIP ALG
SoftSwitch
Public IP
Private IPPrivate IP
Private IP
Public IP
Public IP
February 25th, 2014 Slide 33
Voice over IPCarleton University
Industry Trends
SIP Trunks
SIP User
Network SP provides phones
Network SP provides end-end IP
IPv6 provides everyone with a global address
SPs compete on a global scale
Infrastructure
With IPv6 all devices can be addressed globally
Removes need for NAT and SIP Proxies (ALG), making global connections possible
For example: call control in NA, gateway in Asia, IP phone in Europe!
SIP is becoming an accepted global standard for IP media device signalling
SIP and IPv6 have the potential to become disruptive technologies in displacing the current (TDM) telephone network systems
Today
February 25th, 2014 Slide 34
Voice over IPCarleton University
Agenda
Executive Summary
History
Business Case
Services
Convergence
Infrastructure
Technical Challenges
February 25th, 2014 Slide 35
Voice over IPCarleton University
Technical ChallengesMany!
There are many…
Voice Quality Delay, lost data, jitter, echo Network issues, non deterministic, connectionless Bandwidth, packet overhead, queue delays Clock synchronisation NAT and ALG for “off-net” connections Security Emergency Location E911 IP address space AND translation End points need to use the same media format, or CODEC
February 25th, 2014 Slide 36
Voice over IPCarleton University
Ps35
SLR Nfor Nc A8 -64 -70 0
Ds RLR Nfo3 2 -62
OLR Nos Ie10 -75 0
STMR Pr RLR Nor No18 35 2 -84 -61
Dr LSTR Pre OLR SLR Ro3 21 35 10 8 95
STMR No18 -61
OLR T RLR Iolr10 150 2 0.44
EL TELR Ist Is54 64 2.20 2.64
T Ro Iq150 95 0.00
STMR Qdu18 1
Ist Idte Id R
2.20 3.54 4.55 88b26
WEPL Idle110 0.84
b28
Tr Ta Idd300 150 0.16
c61
Ro
95
Techincal ChallengesVoice Quality - Metrics
To a User - It’s a Phone!
Voice Quality Metrics
Toll Quality Mean Opinion Score (MOS) of 4.0 or better E-Model with R=80 or better Output based on many inputs:
Delay Levels Echo Background noise CODEC
R=88
Continued Voice Quality is expected Continued Voice Quality is expected
February 25th, 2014 Slide 37
Voice over IPCarleton University
Technical ChallengesVoice Quality- Delay and Loss
Voice Quality
With good echo cancellation techniques End to end delays of ~150ms are tolerable 1% packet loss with good Packet Loss Concealment is also tolerable Jitter only becomes significant when it results in packet loss Jitter buffer balance between adding delay and introducing packet loss
G.711 - QoS Versus Delay and % Packet Loss
40
50
60
70
80
90
0 100 200 300 400 500
Total One Way Delay (ms)
QoSR Value
0% PL
1% PL
2% PL
3% PL
1% PLC
2% PLC
3% PLC
Satisfied
Some User Dissatisfied
Many Users Dissatisfied
Nearly All Users Dissatisfied
Not Recommended
55 dB
Far EndEcho Loss
Note: Above 200ms an additional 20ms delay is worse than 1% packet loss with PLC.
Some Delay is tolerable Some Delay is tolerable
February 25th, 2014 Slide 38
Voice over IPCarleton University
Technical ChallengesVoice Quality - Echo
De-packetisation
Packetisation
SpeechDecode
SpeechEncode
Jitter Bufferand Packet
LossConcealment
Echo Canceller
NLP
EchoPrediction
D / A
A / D
Electrical Coupling
IP Gateway End Point
De-packetisation
Packetisation
SpeechDecode
SpeechEncode
Jitter Bufferand Packet
LossConcealment
Echo Canceller
NLP
EchoPrediction
D / A
A / D
Acoustic Coupling
IP-Phone End Point
Echo is always present, even in TDM
Delays in IP make this more noticeable
IP
IP
Control of Echo is importantControl of Echo is important
February 25th, 2014 Slide 39
Voice over IPCarleton University
Technical ChallengesVoice Quality - Delay
Let’s look at where delay occurs
Fixed Delays in CODECs and filters Packet size delays to build a packet Jitter Buffer Network (which also introduces jitter)
End to End Delay = 79ms, but with 10ms jitter (router)
3ms
CODECFilters
2ms
20msPacket Creation
40msJitter Buffer
2ms
L2Switch
2ms
1-10ms
RouterQueue
1-10ms
2ms
L2Switch
2ms
2ms
CODECFilters
3ms40ms
Jitter Buffer20ms
Packet Creation
Network
Control of Delay is importantControl of Delay is important
February 25th, 2014 Slide 40
Voice over IPCarleton University
Technical ChallengesNetwork Jitter
Where does jitter come from? Serialization delay: Waiting for larger packets to transfer Lack of Priority means all data is treated equally - First in First out
Apply priority queues for voice and set MTU to cut large packets
Voice 1 Voice 2 Voice 3 Voice 1 Voice 2 Voice 3
Data
Input
O/Pw/o
MTU
Voice 1 Voice 2 Voice 3 Data Voice 1
Delay x ms
O/PwithMTU
Voice 1 Voice 2 Voice 3 Data1 Voice 1 Voice 2 Voice 3Data2 Data3
MTU Breaks up large packets
Priority mechanism to get voice into gap first
Use QoS settings to prioritize voice and minimize jitterUse QoS settings to prioritize voice and minimize jitter
February 25th, 2014 Slide 41
Voice over IPCarleton University
Technical ChallengesNetwork Jitter
Removal of jitter
Voice CODECs run at a constant rate Too much or too little will result in a gap Small gaps in voice are not discernable <60ms Small gaps in tones are discernable Jitter Buffer needed = Leaky Bucket
Packet Loss Concealment hides loss
Fill gaps with noise, silence Remove data in fixed size, during silence
Packet Arrival
Buffer Fill
Jitter Range
Jitter Buffer = ‘Leaky Bucket’
PLC Hides lost packets
Jitter Buffer = ‘Leaky Bucket’
PLC Hides lost packets
February 25th, 2014 Slide 42
Voice over IPCarleton University
Technical ChallengesClock Slip
Clock Slip
The CODEC at each end may run at 64kbits/s, but they have a tolerance No clock synchronization, therefore need to add or drop data Example of packet drop due to slip
Suppose two device, each at 50ppm (TDM tolerance) That’s 100 bits drift in 1 million bits, or 8 bits in 80,000 bits which = 1 bit every 1.25 seconds @ 64kbits/s, or 1 packet (160 bytes) every 3 minutes, 20 seconds
Clock slip buffer needs to consider this drift up and down Often, slip correction is included with jitter buffer control to minimize
media delays and complexity of multiple buffers
Clock Slip
Fast Clock
Slow Clock
Clock Slip needs to be consideredClock Slip needs to be considered
February 25th, 2014 Slide 43
Voice over IPCarleton University
Transferring tones is problematic if the jitter buffer discards
A DTMF tone need only be 75ms long. A packet loss of 20ms is significant, results in misdialed digits. Convert tones to signalling packet (RFC4733) and regenerate at edge (if
needed)
Technical ChallengesTransmitting Tones
RFC4733 ensures DTMF tones are transferred correctlyRFC4733 ensures DTMF tones are transferred correctly
IP Network
RFC4733
DTMF
February 25th, 2014 Slide 44
Voice over IPCarleton University
Technical ChallengesFAX and Modem
In band tone transmission
Other devices use in band tones, such as: FAX and MODEM
FAX will work, but only under very controlled network conditions, such as packet loss
MODEMs will work, but again under controlled conditions such as echo cancellation
Alternative CODEC for FAX is T.38 (and less often T.37)
Alternative CODEC for MODEM (V.150) is under investigation
Proposals have been made, but due to complexity there is currently little enthusiasm to include this in gateways.
Limited (proprietary) solutions are available.
FAX and MODEM need alternative CODECsFAX and MODEM need alternative CODECs
February 25th, 2014 Slide 45
Voice over IPCarleton University
Technical ChallengesPacket Size
How big a packet should be used?
20ms Packets - Good Compromise20ms Packets - Good Compromise
Packet Rate Use Advantages Disadvantages
10ms High speed network Low latency High Bandwidth and packet rate, not all codecs work
20ms Mixed network, including WAN
Acceptable latency, minimum rate for more complex codecs
Reasonable bandwidth usage
30ms Wireless access Reduced packet rate Increased latency, not all codecs work
40-60ms Lower speed links, satellite
Reduced bandwidth Increased latency, reduced end user quality of use experience
February 25th, 2014 Slide 46
Voice over IPCarleton University
Technical ChallengesCODEC
So many CODECs, which one to choose?
Balance of Voice Quality and Bandwidth usageBalance of Voice Quality and Bandwidth usage
CODEC Type Voice Quality Network Impact
G.711“The Standard”
Base CODEC. Good voice quality. PSTN compatible
High Bandwidth, for voice.
G.726(Delta Modulation)
Good Voice Quality Limited bandwidth reduction. Poor return on processing investment
G.729, G.729a(Compression)
Acceptable voice quality Much reduced bandwidth. Good for WAN access and wireless. Good return on processing investment
G.729b(Compression + Silence suppression)
Reduced voice quality. Silence detection and switching causes issues
Potential for further reduced bandwidth doesn’t materialize. Bandwidth must still be provisioned, even if not used.
G.722.1(Wideband)
Much improved voice quality (8kHz) over G.711. Good user experience
Reduced bandwidth compared to G.711. Good return on processing investment.
February 25th, 2014 Slide 47
Voice over IPCarleton University
Technical ChallengesBandwidth
How much bandwidth needed?
Payload G.711: 160 Bytes (64kbps) G.722.1: 80 Bytes (32kbps) G.729: 20 Bytes (8kbps)
Plus Overhead: RTP, UDP, IP, MAC and Ethernet + inter-packet gaps
LAN Bandwidth (Ethernet)
G.711 ~ 100kbits/s
G.722.1 ~ 65kbits/s
G.729 ~ 40kbits/s
LAN Bandwidth (Ethernet)
G.711 ~ 100kbits/s
G.722.1 ~ 65kbits/s
G.729 ~ 40kbits/s
February 25th, 2014 Slide 48
Voice over IPCarleton University
WAN/InternetLAN
Techincal ChallengesNAT and ALG (Off network connections)
Private IP Address Space Public IP
Address Space
Only translates header of message, so internal addresses are incorrect
10.10.1.1
2.3.4.55.6.7.8
SA DA Message10.10.1.1 5.6.7.8 Send Voice to 10.10.1.1
SA DA Message2.3.4.5 5.6.7.8 Send Voice to 10.10.1.1
SA DA Message2.3.4.5 5.6.7.8 Send Voice to 2.3.4.5
NAT Only
NAT and ALG Protocol Aware and translates both header IP and message content as well
NAT/ALG
February 25th, 2014 Slide 49
Voice over IPCarleton University
The ChallengesSecurity
Security:
How accessible is the equipment Put a lock on the door!
How robust is the system to attack, DOS? Harden system to cater for fault conditions as well as normal operation. Authentication (Who is this?) Authorization (Is this action allowed?) Encryption (You can’t see this, well not easily) Integrity (Did someone tamper with this?)
Phreakers gaining access for free calls, or charging others Provide separate access, e.g. separate physical connection Remove ‘backdoors’ Ring-back on MODEM
Lock the Door! Lock the Door!
February 25th, 2014 Slide 50
Voice over IPCarleton University
The ChallengesSecurity
Security
Monitoring and substitution of voice UDP has no ACK/NACK, can be substituted, redirected Encryption, use of public and private keys DES, DES-3, RC-4, AES, SSH, SSL, IP-SEC, etc. Legal issues and Intellectual Property in distribution and use of encryption
Access through firewalls Open up ports, but this makes it ‘look like a pin cushion’ Use a Session Border Controller, or Application Level Gateway, dynamically
opens ports as needed based on application VPN between sites, but not to Internet direct
Understand where data may be public and safeguard access and read rights
Understand where data may be public and safeguard access and read rights
February 25th, 2014 Slide 51
Voice over IPCarleton University
The ChallengesRules and Regulations
Emergency Location (E911)
Emergency Location (E911) requires that a person making an emergency call can be physically located within a pre-defined area
IP phones can move and be located globally These requirements are potentially in conflict New global standards and regulations are evolving to maintain this capability
IETF-ECRIT : “Framework for Emergency Calling using Internet Multimedia”
CALEA Call Tracing, Malicious call handling Wire-tapping
Charging for services
Who pays? The Internet is ‘free’ But, is it?
Local and Global rules need to be
applied
Local and Global rules need to be
applied
February 25th, 2014 Slide 52
Voice over IPCarleton University
The ChallengesIPv6
IPv4 Public Address
The current public address range has run out!
Main users are NA and Europe Insufficient for ROW Exhaustion
IANA Jan 2011 Regional Internet Regions:
April 2011
IPv6 Public Address Driver: 3G/4G wireless, internet
connected appliances Already being deployed in a
number of countries
IPv6 is here! IPv4 has run outIPv6 is here! IPv4 has run out
IPv4 Sold
February 25th, 2014 Slide 53
Voice over IPCarleton University
Finale
VoIP is mainstream Mobility and Unified Communications Business Process Improvement, rather than networking and toll bypass
Technical challenges for voice quality are being overcome The large Telecos are changing to embrace the IP changes SIP is becoming a common communication method, and feature
interaction between vendors is improving Many new providers appearing in the market place IPv6 is being implemented to provide truly global communications
SIP and IPv6 are disruptive communication technologies Many business and global changes expected because of these Many carriers providing voice, data and now IP Voice services
Thank You