VoIP Lecture 8 Paul Flynn

2 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 CO IXC SJ SF RTP

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

Local Loop 2 wire from phone to switch Tip and Ring - derived from old switchboard plugs 4 wire used at switch Conversion performed by hybrid 2 wire SwitchSwitch switchswitch SpeakerListener Talker Echo

Local Loop (cont.) Problems with Analog Transmission 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 2 wire HybridHybrid HybridHybrid SpeakerListener Talker Echo

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

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

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

Linear Predictive Coding Source Coding Actual Code Predicted Code ms

Bandwidth Requirements Voice Band Traffic Encoding/Compression Result Bit Rate 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 CELP 6.3/5.3 kbps Variable

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

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

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

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

16 Voice Transport Protocols

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

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

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20 Protocols Used H 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

Rensselaer Polytechnic Institute 21 VoIP Camps ISDN LAN conferencing IP H.323 I-multimedia WWW IP SIP Call Agent SIP & H.323 IP “Softswitch” BISDN, AIN H.xxx, SIP “any packet” BICC Conferencing Industry Netheads “IP over Everything” Circuit switch engineers “We over IP” “Convergence” ITU standards Our focus

Rensselaer Polytechnic Institute 22 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) 4 IP SIP Phones and Adaptors 1 3 Analog phone adaptor Palm control 2 54

Rensselaer Polytechnic Institute 23 PSTN to IP Call PBX PSTN External T1/CAS Regular phone (internal) Call SIP server sipd Ethernet 3 SQL database => bob sipc 5 Bob’s phone Gateway Internal T1/CAS (Ext: ) Call

Rensselaer Polytechnic Institute 24 IP to PSTN Call Gateway ( ) 3 SQL database 2 Use Ethernet SIP server sipd sipc 1 Bob calls PSTN External T1/CAS Call PBX Internal T1/CAS Call Regular phone (internal, 7054)

25 End-to-End Delay SenderReceiver Network Transit Delay t A A A A Network Last Bit Received First Bit Transmitted Processing Delay Processing Delay End-to-End Delay

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

Variable Delay Components Queuing delay Dejitter buffers Variable packet sizes Dejitter Buffer Queuing Delay

28 Delay Variation—“Jitter” t t Sender Transmits Sink Receives A A B B C C A A B B C C D1D1 D 2 = D 1 SenderReceiver D 3 = D 2 Network 85

29 Network QoS Toolkit

30 Logical Connections Call Leg 3 Call Leg 1 IP Cloud Call Leg 2 Call Leg 4