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© Kenega Training Ltd 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
© Kenega Training Ltd 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
© Kenega Training Ltd 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
© Kenega Training Ltd Corporate Telephony using a Digital 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 ASwitch B Signalling between user and network (Q931) PCM is analagous to G711 codec in VoIP
© Kenega Training Ltd Connection between User and Network PRI using CCS 012345678 910111213141516171819202122232425262728293031 G.704 Frame Structure – 32 x 64 kbps timeslots Q 931 MessageFCSAddressing/Control7E Signalling Timeslot carries Q921 Frames Time slots 1 – 15 and 17 – 31 are bearer channels Time slot 16 carries byte samples of packetised voice signalling
© Kenega Training Ltd E164 Addresses XXX 000 XXXXXXXXXXX International Call Request National Call Request Country Code National Destination Code Subscriber Number E164 telephone numbers are network layer addresses Network addresses have hierarchical structure
© Kenega Training Ltd 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 PSTN SS7 signalling can be backhauled into a Data Network
© Kenega Training Ltd Performance of the Voice Network 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 PSTN
© Kenega Training Ltd 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
© Kenega Training Ltd 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
© Kenega Training Ltd The Internet Protocol (IP) Source Host Host to Host Internet Process Network Access Destination Host Host to Host Internet Process Network Access Internetwork payload IP Header IP datagram routed through connectionless, unreliable internetwork using destination IP address in IP header
© Kenega Training Ltd Network Access Headers within TCP/IP Host to Host Internet Process TCP/IP Stack Application Data Process Header TCP or UDP Header IP Header Network Access Layer Header FCS e.g. FTP TFTP TELNET or VOICE PROTOCOLS e.g. PPP Frame Relay Ethernet
© Kenega Training Ltd 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
© Kenega Training Ltd 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
© Kenega Training Ltd 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)
© Kenega Training Ltd Data Voice IP Voice & Data Convergence Convergence of voice and data networks Reduce rising communications costs Real-time voice over IP
© Kenega Training Ltd 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: lInternet Architecture Board (IAB) lInternet Corporation for Assigned Names & Numbers (ICANN) lSIGTRAN lSoft Switch Consortium (SSC) lForums (SIP, H.323, etc)
© Kenega Training Ltd 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.
© Kenega Training Ltd Gateway Packet SwitchedCircuit 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.
© Kenega Training Ltd Voice Conversion Internet or Private IP Network PABX (Gateway) Analogue (or Digital) Packets IP PCM samples are delayed, optionally compressed, and carried across the IP network in IP packets
© Kenega Training Ltd Three Styles of Call Phone to phone Phone to PC /PC to Phone PC to PC
© Kenega Training Ltd Voice Coding Compression Method Bit Rate (kbps) Frame Size (mS) Year Finalised G.711 (PCM)640.1251972 G.726 (ADPCM)40,32,24,160.1251988 G.728 (LD-CELP)160.6251992 G.729 (CS-ACELP)8101995 G.723.1 (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
© Kenega Training Ltd 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
© Kenega Training Ltd Payload Formats PTIENCODINGMEDIACLOCK (Hz) 0PCM (µ-Law)Audio8000 3GSMAudio8000 8PCM (A-Law)Audio8000 9G.722Audio8000 15G.728Audio8000 18G.729Audio8000 31H.261Video90000 34H.263Video90000 101NTEdtmf tonesn/a 96 – 127 (dyn)GSM-HRAudio8000 96 – 127 (dyn)GSM-EFRAudio8000
© Kenega Training Ltd Mean Opinion Score (MOS) Compression Method Bit Rate (kbps) Frame Size (mS) MOS G.711 (PCM)640.1254.1 G.726 (ADPCM)40,32,24,160.1253.85 G.728 (LD-CELP)160.6253.61 G.729 (CS-ACELP)8103.92 G.723.1 (MP-MLQ) (ACELP) 6.3 5.3 30 3.9 3.65
© Kenega Training Ltd 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
© Kenega Training Ltd 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
© Kenega Training Ltd IP Bandwidth Requirements for a Voice Call Codec Bit Rate (kbps) Delay (mS) IP Bandwidth G.711 (PCM)640 2.6 Mbps G.711(PCM)641096 kbps G.711(PCM)642080 kbps G.711 (PCM)6430 74 kbps G.729 (82024 kbps Layer 2 overhead needs to be accounted for also cf 64 kbps for voice call over PSTN
© Kenega Training Ltd The H.323 Protocol Stack IP TCP or UDPUDP RTP Compressed Audio Control RTCP Control RAS Capabilities Exchange H.245 Call Signalling H.225 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)
© Kenega Training Ltd The SIP Protocol Stack IP UDP (or TCP)UDP RTP Compressed Audio Control RTCP Capabilities Exchange SDP (SIP) Call Signalling SIP Similar to HTTP and SMTP – text based protocol Highly scalable – utilises DNS Classic client/server Internet Model Uses well known signalling port 5060 (UDP)
© Kenega Training Ltd 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:firstname.lastname@example.org (requires DNS lookup) sip:email@example.com Either can be placed directly on a Web page Two kinds of SIP messages Requests (from client) Responses (from server)
© Kenega Training Ltd firstname.lastname@example.org email@example.com SIP Proxy Server Example DNS lookup INVITE ken @kenega.co.uk kenega.co.uk firstname.lastname@example.org 200 OK INVITE ken @kenega.co.uk 200 OK ACK email@example.com ACK firstname.lastname@example.org SIPSERV IP address DNS SIP Registrar (Location) Server Called PCSIP proxy Server Calling PC email@example.com wants to call firstname.lastname@example.org but he has gone to Eurotech for the day RegisterRegister
© Kenega Training Ltd Call Connection with MGCP Notify CreateConnection ModifyConnection IP Network MG1 MG2 Call Agent Digit Map Voice Signalling Voice Path Notify DeleteConnection Call Setup Call Teardown ModifyConnection
© Kenega Training Ltd IP TCPUDP RTP Compressed Audio /Video Audio Control RTCP Cisco ‘Skinny’ Protocol 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
© Kenega Training Ltd LANs and WANs lLANs: lTraditional Ethernet, 10Mbps, no QoS support, legacy technology lFast Ethernet, 100 Mbps, supports QoS, standard access switch lGigabit Ethernet, 1000 Mbps, point-to-point, supports QoS l10 Gb Ethernet, 10000 Mbps, supports QoS, work in progress lWANs: lX.25 - up to 256kbps, old technology, but still widely used, no QoS support lFrame –Relay – up to 2 Mbps, used for WAN interconnect, limited QoS support lISDN – up to 128Kbps (BRI) or 2 Mbps (PRI), used for backup and remote working, no QoS support lxDSL – up to 16Mbps, used for SOHO internet connections, backup and remote working, no QoS support lATM – up to 2Gbps (155/622 Mbps more normal), extensive QoS support, slowly losing favour but large installed base lMPLS – extensive QoS support and will be covered in QoS chapter
© Kenega Training Ltd 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
© Kenega Training Ltd H.323/SIP Terminal ISP Internet VoIP Implementation (Domestic) - 1 International voice calls at local call rates. Likely to be used with broadband access.
© Kenega Training Ltd 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 Terminal ISP Internet Skype Server
© Kenega Training Ltd 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) Broadband Router DSLAM Internet FXS Interface Communicate via soft switch (see later)
© Kenega Training Ltd VoIP Implementation (Corporate) - 1 This is a typical H323 implementation Gatekeeper gives Call Admission Control PBX PSTN Router Gateway WAN Gatekeeper
© Kenega Training Ltd 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
© Kenega Training Ltd RTP SS7 SIP MGCP H.323 RTP Gateway IP Network Accounting PSTN Call agent Proxy Server GK Soft Switch VoIP Implementation (Carrier/Large Corporate) Voice Signalling Voice Path Signalling Gateway
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