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Voice over IP

Voice over IP. حسين كاري زاده دي ماه 1388. Agenda. ‘ old world’ voice = TDM ‘ new world’ voice packetization Quality of service Signalling Issues with NAT Security. Telephony Equipment. Basic Telephone handset. Key system Mechanical to electronic 2-10 telephone handsets is typical.

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Voice over IP

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  1. Voice over IP حسين كاري زاده دي ماه 1388

  2. Agenda • ‘old world’ voice = TDM • ‘new world’ voice packetization Quality of service Signalling Issues with NAT Security

  3. Telephony Equipment Basic Telephone handset Key system • Mechanical to electronic • 2-10 telephone handsets is typical PABX • Advanced features and call routing • 10-100’s of telephone handsets The Telephone Exchange / C.O.

  4. Analogue Telephony—Signaling • Supervisory – on-hook/off-hook “Can I make a phone call??” • Addressing - DTMF “…the dialed number…” • Call progress – ringback tone “…is the phone ringing or engaged?”

  5. BELL BELL BELL Off-hook,close loop + DC Current Switch 48v – Ring on-hookAns off-hook BELL !! AC + Ringing Switch 48v – Loop Start Signaling (FXS) On-hook,open loop Station PBX or Central Office Loop(Local or Station) + Switch 48v T – R Current sense

  6. Basic Call Progress: Idle On-Hook Open Circuit On-Hook Open Circuit Telephone Switch Local Loop Local Loop 48v

  7. Basic Call Progress: Dialing Off-Hook Closed Circuit On-Hook Open Circuit dialtone Telephone Switch Local Loop 48v DC Current Dialed Digits Pulses or Tones

  8. Basic Call Progress: Switching Address to Port Translation Off-Hook Closed Circuit On-Hook Open Circuit ? Telephone Switch Local Loop 48v

  9. 90V ACRing Signal Ring BackTone RG Basic Call Progress: Ringing Off-Hook Closed Circuit On-Hook Open Circuit Telephone Switch Local Loop Local Loop 48v

  10. RG Basic Call Progress: Talking Off-Hook Closed Circuit Off-Hook Closed Circuit Voice Energy DC Current X Voice Energy DC Current Telephone Switch Local Loop Local Loop 48v

  11. Voice Signalling Trunk Signalling PSTN PBX PBX PBX to PBX Signalling Station Loop Signalling Private Network

  12. Listener Talker Talker Echo Listener Echo Echo in Voice Networks Delay inthe network

  13. Echo Is Always Present … Too Much Echo Is Bad,but No echo is also bad!! - 50 High Loss Echo Is Unnoticeable Echo Loss (dB) Echo Is a Problem Low Loss - 10 ~200 ~20 Echo Path Delay (ms)

  14. Impedance Mismatch is here Echo Is Experienced here How Does Echo Happen? Echo Is Due to a Reflection Tx Rx RemoteExchange Local Exchange Impedance Mismatch at the 2w-4w Hybrid Is the Most Common Reason for Echo

  15. 3700Hz voice bandwidth Human Ear Response Telephone Network Speech and the Telephone Network Power / Volume 300Hz 3400Hz 4kHz 16kHz Frequency / Pitch

  16. Mean Opinion Score Source Channel Simulation Impairment Codec ‘X’ 1 2 3 4 5 “Nowadays, a chicken leg is a rare dish” Rating Speech Quality Level of Distortion 5 Excellent Imperceptible 4 Good Just perceptible but not annoying 3 Fair Perceptible and slightly annoying 2 Poor Annoying but not objectionable 1 Unsatisfactory Very annoying and objectionable 1 2 3 4 5 MOS of 4.0 = Toll Quality

  17. Summary • Analogue voice technology dates back to the late 1800s; • Analogue information exchange is based on voltage, current sense, grounding; • Echo is a fundamental component of Analogue voice and must be controlled.

  18. Agenda • ‘old world’ voice • ‘new world’ voice packetization Quality of service Signalling Issues with NAT Security

  19. Signaling Network In-/Out-of-band Sig Link Bearer facility Transport Network PBX PBX SSP SSP Phone B X2001 Phone A X1001 CO Trunks CO Trunks SS7, QSIG, Proprietary SCP STP STP Wide Area Network Switch Switch Router Router Computer A 200.1.1.1 Computer B 200.1.2.1 Ethernet Ethernet BGP, OSPF, EIGRP, RIP In-band Routing/Signaling Voice/Data Network Components

  20. X2001 X1001 PRI PRI PBX PBX Class 5 Class 5 Class 4 Class 4 R1 R2 R3 R4 X2001 10.1.2.1 X1001 10.1.1.1 10.1.1.1 10.1.2.1 Voice Voice Switch Switch Connection vs. Connectionless Connection signaled based on destination number Connection remains up for duration of call Packets are routed byhop, flow, or destination

  21. IP Phones • QoS in phones - standard 802.1p/q • Integrated Ethernet switching • Easy access to new world features IPv6 GigaEthernet Video IEEE 802.1x

  22. Inline Power: IEEE 802.3af Provides DC Power over Standard Category-5 Ethernet • IP phone are power hungry and you do not want to have a 220V power cable • => get power through the UTP cable Inline Power 10/100 Ethernet without Inline Power

  23. Agenda • ‘old world’ voice • ‘new world’ voice Packetization Quality of service Signalling Issues with NAT Security

  24. Sample rate = 2 x highest frequency Analogue to Digital Voice Pulse Code Modulation—Nyquist Theorem Sampling Stage Analogueue Audio Source 8,000 samples per second B/W = 300 to 4000Hz 1 sample = 8 bits; 8000 samples/sec = 64,000 bit/s Digital Audio Stream ...00100101111011001001...

  25. Codec Speech Compression TechniquesWhat does the Compression? Digital Signal Processor Speech Compression Voila... DSP

  26. Speech Compression TechniquesOverview • Waveform Coding • PCM • Differential Waveform Coding • DPCM, ADPCM • Source algorithms • Generic CELP, CSA-CELP

  27. Mean Opinion Scores 5 Hybrid Coders (LD-CELP & CS-ACELP) 4 Waveform Coders (ADPCM) Subjective Quality (MOS) 3 2 Vocoders (Older Technology) 1 2 4 8 16 32 64 Kbps Score Quality Description of Impairment 5 4 3 2 1 Excellent Good Fair Poor Bad Imperceptible Just Perceptible, not Annoying Perceptible and Slightly Annoying Annoying but not Objectionable Very Annoying and Objectionable Source: A.M. Kondoz, “Digital Speech Coding for Low Bit-Rate Communications Systems”, 1995

  28. Voice Activity Detection – G.729b B/W recovered - 31 dbm No Voice Traffic Sent Voice Activity (Power Level) Hang Timer - 54 dbm Speech “Spurt” Silence Speech “Spurt” Time

  29. RTP/RTCP—RFCs 1889/1890 • End-to-end network transport function Payload type identification—voice, video, compression type Sequence numbering Time stamping Delivery monitoring • RTCP (Real-Time Control Protocol) VER Payload Type M CC Sequence Number 4 Bytes RTP Timestamp 4 Bytes Synchronization Source (SSRC) ID 4 Bytes

  30. RTP (12) IP Header (20) UDP (8) Header is 40 bytes PAYLOAD : 20 Compressing RTP Header gives 4-5 PAYLOAD : 20 11 kbps of bandwidth per call Bandwidth Per IP Call 20ms @ 8kbit/s of compressed voice 26 kbps of bandwidth per call

  31. Summary • All voice over the telephone network is somewhat compressed; • DSPs allow very high compression rates while producing good quality speech • Silence suppression can deliver additional bandwidth efficiencies

  32. Agenda • ‘old world’ voice • ‘new world’ voice Packetization Quality of service Signalling Issues with NAT Security

  33. PBX PBX Delay and Voice Sender Receiver Network First Bit Transmitted Last Bit Received A A Network Transit Delay t Processing Delay Processing Delay End-to-End Delay

  34. Network Delay Variation—“Jitter” SenderA ReceiverB B A C Sender Transmits t d1 d2 C B A B Receives t D2 = d2 D1 = d1 Jitter

  35. Delay and Jitter • Delay and jitter are generated when a packet is stored and forwarded: by router and switches (frame, cell) • Delay is also generated by links 1 microsecond every 200 Km • Jitter is also caused by burst

  36. Delay in Perspective Cumulative Transmission Path Delay CB Zone Satellite Quality Fax Relay, Broadcast High Quality 0 100 200 300 400 500 600 700 800 Time (msec) Delay Target

  37. Integrated Services QoS Model Resource Reservation Protocol

  38. RSVP Agent for Dumb Phones Main Office Edge router contains an RSVP Agent, which is the RSVP signaling proxy for Cisco CallManager CallManager SIP Proxy Signaling To RSVP Agents To Establish Inter-location Reservation Remote Office #1 Remote Office #2 Reserved Path (audio stream) Phone To Agent Media – Not Reserved RSVP Agent RSVP Agent

  39. Differentiated Services Finance Manager Catalyst Switch Enforcement Remote Campus CampusBackbone Cisco Router Catalyst Switch Cisco Router Classification Classification Order Entry, Finance, Manufacturing Multimedia Training Servers

  40. Len ID offset TTL Proto FCS IP-SA IP-DA Data Packet Classification Layers 3 bits called IP Precedence for differentiated services (DiffServ may use 6 D.S. bits plus 2 for flow ctrl) Layer 3 IPV4 Version Length ToS 1 Byte 3 bits used for COS (user priority) Layer 2 802.1Q/p PREAM. SFD DA SA TAG 4 Bytes PT DATA FCS

  41. QoS Policy Enforcement CARCommitted Access Rate Admission Control CongestionManagement CongestionAvoidance Traffic Shaping WREDWeighted Random Early Detection GTSGeneric Traffic Shaping PQ Priority Queuing CBWFQ Class Based WFQ

  42. ML-PPP queueing algorithm • Fragment large packets • Let small packets: Use “normal” encapsulation Interleave with fragmented traffic Voice 2 Voice 1 Jumbogram Fragment 4 Fragment 3 Voice 2 Fragment 2 Voice 1 Fragment 1

  43. Agenda • ‘old world’ voice • ‘new world’ voice Packetization Quality of service Signalling Issues with NAT Security

  44. Config-Table: MAC add-> config 2-DHCP & TFTP MAC add-> config 3-MAC 4-Config Simple signaling: SCCP /1 Catalyst Switch The phone is powered, what next? 1-Phone looks for DHCP server 2-Phone gets IP + CM address IP Phone 3-Phone sends MAC to CM 4-CM sends configuration 1-DHCP? MCS-7835 Call Manager IP Phone IP Phone

  45. RING!! 1-“320” 4-“Direct IP connection between phones” 3-“210 is calling!” #210 = 20.10.1.1 2-CM Routing: #320 = 30.20.1.1 #430 = 40.30.1.1 Simple signaling: SCCP /2 Catalyst Switch What happens if IP Phone ‘210’ calls‘320’? 1-Phone sends ‘3’, ‘2’, ‘0’ to CM 2-CM recognizes number in routing-table #210 3-CM send call request to ‘30.20.1.1’ 4-Phone ‘320’ answers , and the phones talk directly to eachother through IP #320 #430 MCS-7835 Call Manager

  46. SIP: Session Initiated Protocol • SIP is another VoIP signaling protocol • Web like • Text format messages Similar to HTTP • Fast call setup • Run over UDP or TCP • SIP proxies are the equivalent of H.323 gatekeepers

  47. SIP Basics • SIP is a peer-to-peer protocol where end-devices (User Agents - UAs) initiate sessions • SIP defines the signaling mechanism • SIP works for voice, video, instant messaging • SIP uses IETF protocols HTTP 1.1 Session Description Protocol (SDP) media (RTP) name resolution & mobility (DHCP & DNS) application encoding (MIME) • SIP is ASCII text-based:- implementation & debugging

  48. INVITE CONNECTED BYE UNREGISTER REGISTER 1XX Information 2XX Success 3XX Redirection 4XX Client Error 5XX Server Error 6XX Global Failure SIP Commands/Responses Commands Responses

  49. INVITE 3xx Redirect INVITE to Address Returned in Contact: of 3XX response 100 Trying 180 Ringing 200 OK ACK BYE 200 OK SIP Call Flow SIP Phone SIP UA / GW Redirect Server Or SIP proxy

  50. What Is 9-1-1 (or 1-1-2 or 9-9-9)? • A simple, easy to remember telephone number that allows automated call routing to the local public safety agency, based on where you are calling from • In some jurisdictions (North America) there are many different destinations; source routed • Mostly ubiquitous for residential service • Varying degrees of deployment globally Enhanced 9-1-1 in North America European Commission current efforts to converge on 1-1-2 India currently has country-wide rollout of 1-0-8

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