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Understanding Voice over IP
by
Eur Ing Dr KR Duncan BSc MSc PhD CEng CITP CPhys
Kenega Training Ltd, Havant, Hants
http://www.kenega.co.uk
02392 454623
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Objectives:
Describe voice telephony in a circuit switched network (e.g Public Switched Telephone Network (PSTN)
Describe requirements on bandwidth and delay in the PSTN Describe suitability of Data Networks for transporting voiceDiscuss main VoIP signalling and transport protocols – SIP,
H323 and RTPDescribe packetisation, codecs and bandwidth requirements
for VoIPPresent some typical VoIP deployments
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Analogue Connection to a Local Exchange Switch
PSTN(circuit switched digital voice – 64 kbps per call)
Copper loop – typically uses loop start signalling for voice
Copper loop can be analogue or digital (BRI)Copper loop mainly analogue for xDSL technologiesLE switch filters analogue voice and digitises using PCMAnalogue voice carried across digital network in 64 kbps channel
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Corporate Telephony using a Digital PBX
PBX PBX
PSTN
Signalling between user and network (Q931)
Signalling within the network (SS7)
Signalling sets up PCM bandwidth for conversation (64 kbps) – in a TDM timeslot
Switch A Switch B
Signalling between user and network (Q931)
PCM is analagous to G711 codec in VoIP
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Connection between User and Network
PRI using CCS
0 1 234 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 2223 24 25 26 27 28 29 3031
G.704 Frame Structure – 32 x 64 kbps timeslots
Q 931 MessageFCS Addressing/Control 7E7E
Signalling Timeslot carries Q921 Frames
Time slots 1 – 15 and 17 – 31 are bearer channelsTime slot 16 carries byte samples of packetised voice signalling
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E164 Addresses
XXX00 0 XXXX XXXXXXX
International Call Request
National Call Request
Country Code
National Destination
Code
Subscriber Number
E164 telephone numbers are network layer addressesNetwork addresses have hierarchical structure
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Signalling Protocol Stacks
Application
Presentation
Session
Transport
Network
Datalink
Physical
7 Layer OSI Model
MTP 3
MTP 2
MTP 1
TUP / ISUP
Part of SS7 Protocol Stack
SS7 signalling is already packetised in PSTNSS7 signalling can be backhauled into a Data Network
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Performance of the Voice Network
ITU G.114 emphasises the need to consider delayOne way end-to-end delay no more than 150 mSPost-dial delay less than 2 seconds
PSTN
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Voice over the PSTN - pros
Service GuaranteesLow DelayLow JitterUses Admission Control Call only accepted if sufficient resources exist in network
Each call receives a dedicated bandwidth
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Voice over the PSTN - Cons
High bandwidth requirement due to legacy standardsEach call requires 64kbps of bandwidth – from PCMNew Codecs utilise only 8k Inefficient usage of bandwidthBandwidth wasted during gaps in the conversation
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The Internet Protocol (IP)
Source Host
Host to Host
Internet
Process
Network Access
Destination Host
Host to Host
Internet
Process
Network Access
Internetwork
payload IPHeader
IP datagram routed through connectionless, unreliable internetwork using destination IP address in IP header
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Network Access
Headers within TCP/IP
Host to Host
Internet
Process
TCP/IP Stack
ApplicationData
ProcessHeader
TCP or UDPHeader
IPHeader
Network AccessLayer Header
FCS
e.g. FTP
TFTP
TELNET
or
VOICE
PROTOCOLS
e.g. PPP
Frame Relay
Ethernet
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Transport (Host to Host) Protocols
Transmission Control Protocol (TCP) Ports numbers point to software application End to end reliability Connection oriented
Stream Control Transmission Protocol (SCTP) Alternative to TCP for backhauling multiple signalling messages
User Datagram Protocol (UDP) Port numbers point to software application Used to carry voice media packets Connectionless Unreliable
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Voice over Data Networks - Pros Packet Switched not Circuit Switched
Packet switching has greater resilience
Call Bandwidth Flexibility Reduced bandwidth per call when using more efficient coding
scheme Bandwidth can be increased on a needs basis
Efficient bandwidth utilisation Available bandwidth can be shared amongst various traffic types
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Voice over Data Networks - ConsNo Service Guarantees (no per call state)
Packets may be queued by Routers Packets may follow different paths
Unpredictable Quality of Service Traffic is sent ‘best-effort’ by default
No admission control Connectionless (Unless controlled by another protocol)
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Data
Voice
IP
Voice & Data Convergence
Convergence of voice and data networksReduce rising communications costsReal-time voice over IP
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Standards Organisations in VoIP
H.323 VoIP Solution - International Telecommunications Union (ITU)
SIP VoIP Solution - Internet Engineering Task Force (IETF)Soft Switching VoIP Solution – ITU and IETFOther Organisations involved:
Internet Architecture Board (IAB) Internet Corporation for Assigned Names & Numbers (ICANN) SIGTRAN Soft Switch Consortium (SSC) Forums (SIP, H.323, etc)
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Analogue Equipment can be used in VoIP
Analogue telephones connect via Foreign Exchange Subscriber (FXS) interface.
FXS interface provides dial tone, battery current and ring voltage to the analogue telephone.
FXS can be an Analogue Telephone Adapter (ATA) or a voice card in a router or server.
Analogue trunk lines can be connected via a Foreign Exchange Office (FXO) interface.
FXO receives POTS from a switch in the Local Exchange and provides on-hook/off-hook indication to switch.
FXO is typically a voice card in a router or server.
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Gateway
Packet SwitchedCircuit Switched
Gateway
To communicate to a PSTN user, a gateway is requiredProvides an interface between:
circuit switched telephone networks (PSTN and GSM) and packet switched IP data networks.
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Voice Conversion Internet or
Private IP Network
PABX(Gateway)
Analogue (or Digital)
PacketsIP
PCM samples are delayed, optionally compressed, and carried across the IP network in IP packets
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Three Styles of Call
Phone to phonePhone to PC /PC to PhonePC to PC
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Voice Coding
CompressionMethod
Bit Rate(kbps)
Frame Size(mS)
YearFinalised
G.711 (PCM) 64 0.125 1972
G.726 (ADPCM) 40,32,24,16 0.125 1988
G.728 (LD-CELP) 16 0.625 1992
G.729 (CS-ACELP) 8 10 1995
G.723.1 (MP-MLQ) (ACELP)
6.35.3
3030
1995
Bit rate for voice call is determined by codec usedG711 codec is mandatory – others are optional
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Real-time Transport Protocol (RTP)
RTP V2 is defined in IETF RFC 1889, along with a profile for carrying audio and video over RTP in RFC 1890
RTP carries voice or video Does not offer any form of reliability or a protocol-defined
flow/congestion controlSequences and Timestamps packets for proper replay Indicates codec used in RTP headerPort 5004 (UDP) registered by IETF – but voice software
can negotiate dynamic port
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Payload Formats
PTI ENCODING MEDIA CLOCK (Hz)
0 PCM (µ-Law) Audio 8000
3 GSM Audio 8000
8 PCM (A-Law) Audio 8000
9 G.722 Audio 8000
15 G.728 Audio 8000
18 G.729 Audio 8000
31 H.261 Video 90000
34 H.263 Video 90000
101 NTE dtmf tones n/a
96 – 127 (dyn) GSM-HR Audio 8000
96 – 127 (dyn) GSM-EFR Audio 8000
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Mean Opinion Score (MOS)
CompressionMethod
Bit Rate(kbps)
Frame Size(mS)
MOS
G.711 (PCM) 64 0.125 4.1
G.726 (ADPCM) 40,32,24,16 0.125 3.85
G.728 (LD-CELP) 16 0.625 3.61
G.729 (CS-ACELP) 8 10 3.92
G.723.1 (MP-MLQ) (ACELP)
6.35.3
3030
3.93.65
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Voice Media Packet using G.711 Codec
G711 codec is mandatory in VoIP implementations IP packet size around 200 bytes
Voice Payload RTPHeader
UDPHeader
IPHeader
e.g. G.711 (20mS delay) = 160 bytes
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Voice Media Packet using G.729/G.723.1 Codec
Compresses voice payload to reduce bandwidth for callAdditional processing degrades quality and adds delayG.729 used by Main vendors such as Cisco and Nortel IP packet size around 60 bytes
VoicePayload
RTPHeader
UDPHeader
IPHeader
e.g. G.729 (20mS delay) = 20 bytes
G.723 (30mS delay) = 24 bytes
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IP Bandwidth Requirements for a Voice Call
CodecBit Rate(kbps)
Delay(mS)
IP Bandwidth
G.711 (PCM) 64 0 2.6 Mbps
G.711(PCM) 64 10 96 kbps
G.711(PCM) 64 20 80 kbps
G.711 (PCM) 64 30 74 kbps
G.729 ( 8 20 24 kbps
Layer 2 overhead needs to be accounted for alsocf 64 kbps for voice call over PSTN
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The H.323 Protocol Stack
IP
TCP or UDP UDP
RTP
CompressedAudioAudio
Control
RTCP
Control
RAS
CapabilitiesExchange
H.245
CallSignalling
H.225
Deployed extensively in corporate environmentGatekeeper offers admission control and bandwidth managementOriginally designed for LAN – poor scalabilityUses well known signalling port 1720 (TCP or UDP)
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The SIP Protocol Stack
IP
UDP (or TCP) UDP
RTP
CompressedAudioAudio
Control
RTCP
CapabilitiesExchange
SDP (SIP)
CallSignalling
SIP
Similar to HTTP and SMTP – text based protocolHighly scalable – utilises DNSClassic client/server Internet ModelUses well known signalling port 5060 (UDP)
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SIP Components
• SIP components• User Agent Client (UAC) — Makes calls
• User Agent Server (UAS) — Answers or rejects calls
• SIP servers (several types) — Locate called parties
• Proxy server
• Redirect server
• Registrar/Location server
• Addressing and naming• sip:[email protected] (requires DNS lookup)
• sip:[email protected]
• Either can be placed directly on a Web page
• Two kinds of SIP messages• Requests (from client)
• Responses (from server)
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[email protected] [email protected]
SIP Proxy Server Example
DNS lookup
INVITE [email protected]
200 OK
INVITE [email protected]
200 OK ACK [email protected] ACK [email protected]
SIPSERV IP address
DNS
SIP Registrar (Location) Server
Called PCSIP proxy Server
Calling PC
[email protected] wants to call [email protected] but he has gone to Eurotech for the day
Register
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Call Connection with MGCP
Notify
CreateConnectionCreateConnectionModifyConnection
IP Network
MG1 MG2
Call Agent
Digit Map
Voice SignallingVoice Signalling
Voice PathVoice Path
NotifyDeleteConnection DeleteConnection
Call Setup
Call Teardown
ModifyConnection
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IP
TCP UDP
RTP
CompressedAudio /VideoAudio
Control
RTCP
Cisco ‘Skinny’
Protocol
Skinny Client Control Protocol (SCCP)
Used for communication between Cisco IP telephones and Cisco Callmanager Server
Proprietary Voice Signalling ProtocolUses TCP port 2000 for voice signalling messages
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LANs and WANs
LANs: Traditional Ethernet, 10Mbps, no QoS support, legacy technology Fast Ethernet, 100 Mbps, supports QoS, standard access switch Gigabit Ethernet, 1000 Mbps, point-to-point, supports QoS 10 Gb Ethernet, 10000 Mbps, supports QoS, work in progress
WANs: X.25 - up to 256kbps, old technology, but still widely used, no QoS support Frame –Relay – up to 2 Mbps, used for WAN interconnect, limited QoS
support ISDN – up to 128Kbps (BRI) or 2 Mbps (PRI), used for backup and
remote working, no QoS support xDSL – up to 16Mbps, used for SOHO internet connections, backup and
remote working, no QoS support ATM – up to 2Gbps (155/622 Mbps more normal), extensive QoS support,
slowly losing favour but large installed base MPLS – extensive QoS support and will be covered in QoS chapter
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VoIP in the WAN - Packet or Circuit Switched?
Connection oriented (ATM or MPLS) or connectionless (IP)A queue of queuesPacket switching gives large variable delay (jitter) making it
unsuitable for delay sensitive data like voiceCircuit Switching gives less jitter and is more suitable for voice
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H.323/SIP TerminalH.323/SIP Terminal
ISPISP
Internet
VoIP Implementation (Domestic) - 1
International voice calls at local call rates.Likely to be used with broadband access.
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VoIP Implementation (Domestic) - 2
Use of Skype or other VoIP Provider for free Internet calls and cheap rate to PSTN – using Skype out
Skype software loaded onto PC’sLikely to use Broadband Access to communicate with
Skype server and other Skype users
Skype TerminalSkype Terminal
ISPISP
Internet
Skype Server
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VoIP Implementations (Domestic/Home Office)
FXS allows analogue telephones to be used for VoIP Dial code allows calls to be carried across Internet Soft phone could be installed on PC
PSTN(circuit switched)
Third Party Data Network
( e.g. BT)
BroadbandRouter
DSLAM
Internet
FXS Interface
Communicate via soft switch
(see later)
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VoIP Implementation (Corporate) - 1
This is a typical H323 implementation Gatekeeper gives Call Admission Control
PBX PBX
PSTN
RouterRouter
GatewayGateway
WANGatekeeper
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VoIP Implementations (Corporate) - 2
FXO allows analogue lines (PSTN) to integrate with VoIP IP telephones can communicate over Internet through IPSec tunnels IP telephones can ‘break out’ to PSTN via FXO interface on SIP PBX
PSTN(circuit switched)
DSLAM
Internet
FXO interface on SIP PBX
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RTP
SS7
SIP
MGCP
H.323
RTP
Gateway
IP Network
Accounting
PSTN
Callagent
Proxy Server
GK
Soft SwitchSoft Switch
VoIP Implementation (Carrier/Large Corporate)
Voice SignallingVoice Signalling
Voice PathVoice Path
Signalling Signalling GatewayGateway
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