1. Understanding Voice over IPby
Eur Ing Dr KR Duncan BSc MSc PhD CEng CITP CPhys
Kenega Training Ltd, Havant, Hants

2. 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 voice
Discuss main VoIP signalling and transport protocols – SIP, H323 and RTP
Describe packetisation, codecs and bandwidth requirements for VoIP
Present some typical VoIP deployments

3. 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 technologies
LE switch filters analogue voice and digitises using PCM
Analogue voice carried across digital network in 64 kbps channel

4. Corporate Telephony using a Digital PBX
Signalling sets up PCM bandwidth for conversation (64 kbps) – in a TDM timeslot
PBX
PBX
Switch A
Switch B
PSTN
Signalling between user and network (Q931)
Signalling between user and network (Q931)
Signalling within the network (SS7)
Notes
PCM is analagous to G711 codec in VoIP

5. Connection between User and Network
PRI using CCS
G.704 Frame Structure – 32 x 64 kbps timeslots 31 30 29 28 27 26 25 24 23 22 21 20 19 18 17 16 15 14 13 12 11 10 9 1 2 3 4 5 6 7 8
Signalling Timeslot carries Q921 Frames 7E
FCS
Q 931 Message
Addressing/Control 7E
Time slots 1 – 15 and 17 – 31 are bearer channels
Time slot 16 carries byte samples of packetised voice signalling
Notes

6. E164 Addresses International Call Request National Call Request XXX
XXXX
XXXXXXX 00
Country Code
National Destination Code
Subscriber Number
E164 telephone numbers are network layer addresses
Network addresses have hierarchical structure
Notes

7. Signalling Protocol Stacks
Part of SS7 Protocol Stack
7 Layer OSI Model
TUP / ISUP
Application
Presentation
Session
Transport
MTP 3
Network
MTP 2
Datalink
MTP 1
Physical
SS7 signalling is already packetised in PSTN
SS7 signalling can be backhauled into a Data Network
Notes

8. Performance of the Voice Network
PSTN
ITU G.114 emphasises the need to consider delay
One way end-to-end delay no more than 150 mS
Post-dial delay less than 2 seconds
Notes

9. Voice over the PSTN - pros
Service Guarantees
Low Delay
Low Jitter
Uses Admission Control
Call only accepted if sufficient resources exist in network
Each call receives a dedicated bandwidth
Notes

10. Voice over the PSTN - Cons
High bandwidth requirement due to legacy standards
Each call requires 64kbps of bandwidth – from PCM
New Codecs utilise only 8k
Inefficient usage of bandwidth
Bandwidth wasted during gaps in the conversation
Notes

11. The Internet Protocol (IP)
Process
Process
Host to Host
Host to Host
Internet
Internet
Destination Host
Network
Access
Source Host
Network
Access
Internetwork
payload IP
Header
Notes
IP datagram routed through connectionless, unreliable internetwork using destination IP address in IP header

12. Headers within TCP/IP TCP/IP Stack Process Host to Host Internet
Network Access
FCS
Application
Data
Process
Header
TCP or UDP
Header IP
Header
Network Access
Layer Header
e.g. FTP
TFTP
TELNET or
VOICE
PROTOCOLS
e.g. PPP
Frame Relay
Ethernet
Notes

13. 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
Notes

14. 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
Notes

15. Voice over Data Networks - Cons
No 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)
Notes

16. Voice & Data Convergence
Convergence of voice and data networks
Reduce rising communications costs
Real-time voice over IP
Voice
Data IP
Notes

17. 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 IETF
Other Organisations involved:
Internet Architecture Board (IAB)
Internet Corporation for Assigned Names & Numbers (ICANN)
SIGTRAN
Soft Switch Consortium (SSC)
Forums (SIP, H.323, etc)
Notes

18. 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.

19. Gateway Circuit Switched Packet Switched Gateway
To communicate to a PSTN user, a gateway is required
Provides an interface between:
circuit switched telephone networks (PSTN and GSM) and packet switched IP data networks.
Notes

20. Voice Conversion
PCM samples are delayed, optionally compressed, and carried across the IP network in IP packets
PABX
(Gateway)
Internet or
Private IP Network IP
Packets
Analogue (or Digital)
Notes

21. Three Styles of Call Phone to phone Phone to PC /PC to Phone PC to PC
Notes

22. Voice Coding Compression Method Bit Rate (kbps) Frame Size (mS) Year
Finalised
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 (MP-MLQ)
(ACELP)
6.3
5.3 30
1995
Bit rate for voice call is determined by codec used
G711 codec is mandatory – others are optional
Notes

23. 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 control
Sequences and Timestamps packets for proper replay
Indicates codec used in RTP header
Port 5004 (UDP) registered by IETF – but voice software can negotiate dynamic port
Notes

24. Payload Formats PTI ENCODING MEDIA CLOCK (Hz) PCM (µ-Law) Audio 8000 3
PCM (µ-Law)
Audio
8000 3
GSM 8
PCM (A-Law) 9
G.722 15
G.728 18
G.729 31
H.261
Video
90000 34
H.263
101
NTE
dtmf tones
n/a
96 – 127 (dyn)
GSM-HR
GSM-EFR
Notes

25. Mean Opinion Score (MOS)
Compression
Method
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 (MP-MLQ)
(ACELP)
6.3
5.3 30
3.9
3.65
Notes

26. Voice Media Packet using G.711 Codec
Voice Payload
RTP
Header
UDP
Header IP
Header
e.g. G.711 (20mS delay) = 160 bytes
G711 codec is mandatory in VoIP implementations
IP packet size around 200 bytes
Notes

27. Voice Media Packet using G.729/G.723.1 Codec
Payload
RTP
Header
UDP
Header IP
Header
e.g. G.729 (20mS delay) = 20 bytes
G.723 (30mS delay) = 24 bytes
Compresses voice payload to reduce bandwidth for call
Additional processing degrades quality and adds delay
G.729 used by Main vendors such as Cisco and Nortel
IP packet size around 60 bytes
Notes

28. IP Bandwidth Requirements for a Voice Call
Codec
Bit Rate
(kbps)
Delay
(mS)
IP Bandwidth
G.711 (PCM) 64
­ 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 also
cf 64 kbps for voice call over PSTN
Notes

29. The H.323 Protocol Stack H.225 H.245 Control RAS Audio Control RTCP
Call
Signalling
H.225
Capabilities
Exchange
H.245
Control
RAS
Audio
Control
RTCP
Compressed
Audio
RTP
TCP or UDP
UDP IP
Deployed extensively in corporate environment
Gatekeeper offers admission control and bandwidth management
Originally designed for LAN – poor scalability
Uses well known signalling port 1720 (TCP or UDP)
Notes

30. The SIP Protocol Stack SIP SDP (SIP) Audio Control RTCP RTP
Call
Signalling
SIP
Capabilities
Exchange
SDP (SIP)
Audio
Control
RTCP
Compressed
Audio
RTP
UDP (or TCP)
UDP IP
Similar to HTTP and SMTP – text based protocol
Highly scalable – utilises DNS
Classic client/server Internet Model
Uses well known signalling port 5060 (UDP)
Notes

31. SIP Components SIP components Addressing and naming
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
(requires DNS lookup)
Either can be placed directly on a Web page
Two kinds of SIP messages
Requests (from client)
Responses (from server)
Notes

32. SIP Proxy Server Example
SIP Registrar (Location) Server
DNS
Register
DNS lookup
kenega.co.uk
SIPSERV IP address
INVITE
INVITE
200 OK
200 OK
ACK
ACK
Calling PC
SIP proxy Server
Called PC
Notes
wants to call but he has gone to Eurotech for the day

33. Call Connection with MGCP
Notify
ModifyConnection
CreateConnection
Call Setup
CreateConnection
ModifyConnection
Notify
DeleteConnection
Call Teardown
DeleteConnection
Call Agent
MG1
MG2
IP Network
Digit Map
Notes
Voice Signalling
Voice Path

34. Skinny Client Control Protocol (SCCP)
Cisco
‘Skinny’
Protocol
Audio
Control
RTCP
Compressed
Audio /Video
RTP
TCP
UDP IP
Used for communication between Cisco IP telephones and Cisco Callmanager Server
Proprietary Voice Signalling Protocol
Uses TCP port 2000 for voice signalling messages
Notes

35. LANs and WANs LANs: WANs:
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, 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
Notes

36. VoIP in the WAN - Packet or Circuit Switched?
Connection oriented (ATM or MPLS) or connectionless (IP)
A queue of queues
Packet switching gives large variable delay (jitter) making it unsuitable for delay sensitive data like voice
Circuit Switching gives less jitter and is more suitable for voice
Notes

37. VoIP Implementation (Domestic) - 1
H.323/SIP Terminal
H.323/SIP Terminal
Internet
ISP
ISP
International voice calls at local call rates.
Likely to be used with broadband access.
Notes

38. VoIP Implementation (Domestic) - 2
Skype Server
Skype Terminal
Skype Terminal
Internet
ISP
ISP
Use of Skype or other VoIP Provider for free Internet calls and cheap rate to PSTN – using Skype out
Skype software loaded onto PC’s
Likely to use Broadband Access to communicate with Skype server and other Skype users

39. VoIP Implementations (Domestic/Home Office)
PSTN
(circuit switched)
Communicate via soft switch (see later)
FXS Interface
DSLAM
Broadband
Router
Third Party
Data Network
( e.g. BT)
Internet
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

40. VoIP Implementation (Corporate) - 1
PBX
PBX
PSTN
Gateway
Gateway
WAN
Gatekeeper
Router
Router
Notes
This is a typical H323 implementation
Gatekeeper gives Call Admission Control

41. VoIP Implementations (Corporate) - 2
DSLAM
PSTN
(circuit switched)
FXO interface on SIP PBX
Internet
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

42. VoIP Implementation (Carrier/Large Corporate)
Soft Switch
SIP
Call agent
Proxy Server
H.323 GK
Accounting
RTP
RTP
SS7
IP Network
Signalling Gateway
MGCP
Notes
Voice Signalling
PSTN
Gateway
Voice Path