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VoIP

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VoIP

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  1. VoIP Lecture 8 Paul Flynn

  2. SF RTP SJ IXC CO Network Components CO - Central Office Trunk - Switch-switch connection Loop - Line from switch to phone Tandem switch - provides switch-switch interconnection IXC - interexchange carrier PBX - Private branch exchange

  3. The PSTN: Separate Voice and Signaling Networks SS7 Voice SSP • SSP: Service Switching Point (Telephone Switch) • STP: Signaling Transfer Point (Router) • SCP: Service Control Point (Database, Logic) STP Trunk SCP SSP Signaling (Packet) Trunk (TDM) Trunk STP SSP

  4. Local Loop Speaker Listener 2 wire Switch switch 2 wire 2 wire • 2 wire from phone to switch • Tip and Ring - derived from old switchboard plugs • 4 wire used at switch • Conversion performed by hybrid Talker Echo 2 wire

  5. Local Loop (cont.)Problems with Analog Transmission Speaker Listener 2 wire Hybrid Hybrid 2 wire 2 wire • Several problems with analog • Attenuation - loss of signal power • Distortion - unequal loss at different frequencies • Noise - induced into line which is amplified along with signal by network components • Echo - due to 2/4 wire conversion • Physical impairments - bad lines, bridge taps, load coils Talker Echo 2 wire

  6. Digitizing Voice • Assumption is that human speech information is contained in the range of 300-3400 Hz • Filter & use signal below 4 kHz to prevent aliasing • Sample and quantize signal at 8kHz • encoder produces 64 kbit/sec stream of data

  7. Waveform Coders (codec) Voice ENCODER Quantizer n Bits/Sample 2n Levels Sampler 2 * Fmax Samples/Sec Low Pass Filter BW = Fmax Binary Encoder Clock Voice DeCODER Binary to Decimal Decoder Pulse Detector Filter BW = Fmax

  8. Non-Linear vs. Linear EncodingCompanding (a-law vs -law) Output Output Input Input Non- Linear Encoding Closely Follows Human Voice Characteristics High Amplitude Signals Have More Quantization Distortion (Both a- & - Law) Linear Encoding Relatively Easy to Analyze, Synthesize, and Regenerate All Amplitudes Have Roughly Equal Quantization Distortion

  9. 0001 0010 0011 0100 0101 0111 1000 1001 1010 1011 1100 1101 1110 1111 0001 0010 0011 0100 0101 0111 1000 1001 1010 1011 1100 1101 1110 1111 0001 0010 0011 0100 0101 0111 1000 1001 1010 1011 1100 1101 1110 1111 0001 0010 0011 0100 0101 0111 1000 1001 1010 1011 1100 1101 1110 1111 Linear Predictive CodingSource Coding 10 20 ms 1001 1011 Actual Code Predicted Code

  10. Bandwidth Requirements Voice Band Traffic Result Bit Rate Encoding/ Compression G.711 PCM A-Law/u-Law 64 kbps (DS0) G.726 ADPCM 16, 24, 32, 40 kbps G.729 CS-ACELP 8 kbps G.728 LD-CELP 16 kbps G.723.1 CELP 6.3/5.3 kbps Variable

  11. Voice Quality Anything Above an MOS of 4.0 Is “Toll” Quality Delay(msec) Compression Method MOS Score 64K PCM (G.711) 4.4 0.75 32K ADPCM (G.726) 4.2 1 16K LD-CELP (G.728) 3–5 4.2 8K CS-ACELP (G.729) 4.2 15 3.6 8K CS-ACELP (G.729a) 15

  12. Voice Activity Detection - 31 dbm B/W Saved Voice Activity (Power Level) Hang Timer No Voice Traffic Sent SID Buffer SID - 54 dbm Pink Noise Voice “Spurt” Silence Voice “Spurt” Time

  13. Applications of Speech Coding • Telephony, PBX • Wireless/Cellular Telephony • Internet Telephony • Speech Storage (Automated call-centers) • High-Fidelity recordings/voice • Speech Analysis/Synthesis • Text-to-speech (machine generated speech)

  14. Different Types of Signaling(when you place a call) • Supervisory - Determines state of line/trunk whether on/off-hook EM signal leads, loop open/closed • Addressing - passes digit information for call routing DTMF, MF, DNIS • Informational - indicates call progress Busy signal, dial tone, ring back

  15. Summary Page Local Loop FXS/ FXO Loopstart/ Gndstart SF RTP SJ IXC CO T1/ E1 DTMF/ MF CAS/ CCS

  16. Voice Transport Protocols

  17. Voice Transport ProtocolOverview Encoder/ Decoder T1/E1 CAS/CCS Cisco Gateway IP ATM, FR, HDLC Cisco Gateway PBX PSTN

  18. Queuing • Voice always given priority over data • Real-time queue for voice and video Data queue serviced only if nothing in Real Time queue - (Exhaustive like priority queuing) • Non-real time queue (Data) WFQ by default WFQ Disabled if Frame Relay Traffic Shaping Enabled Fancy queuing disabled if voice-encap set on interface

  19. Protocols Used • H.225.0 for Connection and Status • Q.931 ‘derived’ messages • ‘RAS’ for Endpoint-GK signaling. • H.245 for negotiating channel usage and capabilities • Media transport • RTP/RTCP -- standard payloads (RFC1889/1890) • ‘native’ uni/multicast support

  20. Our focus VoIP Camps Circuit switch engineers “We over IP” “Convergence” ITU standards Conferencing Industry Netheads “IP over Everything” H.323 SIP “Softswitch” BICC ISDN LAN conferencing I-multimedia WWW Call Agent SIP & H.323 BISDN, AIN H.xxx, SIP IP IP IP “any packet”

  21. IP SIP Phones and Adaptors 1 • Are true Internet hosts • Choice of application • Choice of server • IP appliances • Implementations • 3Com (3) • Columbia University • MIC WorldCom (1) • Mediatrix (1) • Nortel (4) • Siemens (5) Analog phone adaptor 2 3 Palm control 4 4 5

  22. PBX Gateway PSTN Internal T1/CAS (Ext:7130-7139) External T1/CAS Call 9397134 Call 7134 Ethernet 1 2 4 5 3 Regular phone (internal) SIP server SQL database sipd sipc Bob’s phone 7134 => bob PSTN to IP Call

  23. PBX Gateway (10.0.2.3) PSTN External T1/CAS Internal T1/CAS Call 5551212 Call 85551212 Ethernet 4 5 2 3 1 5551212 Bob calls 5551212 Regular phone (internal, 7054) SIP server SQL database sipd sipc Use sip:85551212@10.0.2.3 IP to PSTN Call

  24. End-to-End Delay Sender Receiver Network First BitTransmitted Last BitReceived A A t Network Transit Delay Processing Delay Processing Delay End-to-End Delay

  25. Fixed Delay Components Propagation Delay Serialization Delay—Buffer to Serial Link Processing Delay • Propagation—six microseconds per kilometer • Serialization • Processing Coding/compression/decompression/decoding Packetization

  26. Variable Delay Components Queuing Delay Queuing Delay Queuing Delay • Queuing delay • Dejitter buffers • Variable packet sizes DejitterBuffer

  27. D3 = D2 Delay Variation—“Jitter” Sender Receiver Network B C A Sender Transmits t A B C Sink Receives D1 D2 = D1 t 85

  28. Network QoS Toolkit

  29. Call Leg 2 Call Leg 1 IP Cloud Call Leg 3 Call Leg 4 Logical Connections