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Understanding Voice over IP by Eur Ing Dr KR Duncan BSc MSc PhD CEng CITP CPhys

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. Objectives:. Describe voice telephony in a circuit switched network (e.g Public Switched Telephone Network (PSTN)

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Understanding Voice over IP by Eur Ing Dr KR Duncan BSc MSc PhD CEng CITP CPhys

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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 http://www.kenega.co.uk 02392 454623

  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) • PCM is analagous to G711 codec in VoIP

  5. 0 1 2 3 4 5 6 7 8 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 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

  6. E164 Addresses National Call Request International Call Request XXX XXXX XXXXXXX 00 0 Country Code Subscriber Number National Destination Code • E164 telephone numbers are network layer addresses • Network addresses have hierarchical structure

  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

  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

  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

  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

  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 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

  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

  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

  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)

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

  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)

  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. 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. Gateway

  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)

  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.723.1 (MP-MLQ) (ACELP) 6.3 5.3 30 30 1995 • Bit rate for voice call is determined by codec used • G711 codec is mandatory – others are optional

  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

  24. Payload Formats

  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.723.1 (MP-MLQ) (ACELP) 6.3 5.3 30 30 3.9 3.65

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

  27. Voice Media Packet using G.729/G.723.1 Codec • 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 Voice Payload RTP Header UDP Header IP Header e.g. G.729 (20mS delay) = 20 bytes G.723 (30mS delay) = 24 bytes

  28. IP Bandwidth Requirements for a Voice Call Codec Bit 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 also • cf 64 kbps for voice call over PSTN

  29. The H.323 Protocol Stack • 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) Call Signalling H.225 Capabilities Exchange H.245 Control RAS Audio Control RTCP Compressed Audio RTP TCP or UDP UDP IP

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

  31. 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:ken@kenega.co.uk(requires DNS lookup) • sip:ken@194.143.174.70 • Either can be placed directly on a Web page • Two kinds of SIP messages • Requests (from client) • Responses (from server)

  32. SIP Proxy Server Example SIP Registrar (Location) Server DNS Register • DNS lookup • kenega.co.uk • ken@kenega.co.uk • SIPSERV IP address • INVITE ken@kenega.co.uk • INVITE ken@kenega.co.uk • 200 OK • 200 OK • ACK ken@kenega.co.uk • ACK ken@kenega.co.uk Calling PC SIP proxy Server Called PC dave@cisco.com ken@kenega.co.uk dave@cisco.com wants to call ken@kenega.co.uk 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 MG2 MG1 IP Network Digit Map Voice Signalling Voice Path

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

  35. 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

  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

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

  38. 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’s • Likely to use Broadband Access to communicate with Skype server and other Skype users Skype Server Skype Terminal Skype Terminal Internet ISP ISP

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

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

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

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

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