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Understanding Voice over IP

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1 Understanding Voice over IP
by 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

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 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

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