Telephony & Internet Telephony

Save this PDF as:
 WORD  PNG  TXT  JPG

Size: px
Start display at page:

Download "Telephony & Internet Telephony"

Transcription

1 Telephony & Internet Telephony Based upon slides of Henning Schulzrinne (Columbia) 1

2 Telephony 2

3 Telephone Network: What is It? Specialized to carry voice traffic Aggregates like T1, SONET OC-N can also carry data Also carries Telemetry, video, fax, modem calls Internally, uses digital samples Switches and switch controllers are special purpose computers Pieces: 1. End systems 2. Transmission 3. Switching 4. Signaling 3

4 Telephone Network: What is It? Single basic service: two-way voice low end-to-end delay guarantee that an accepted call will run to completion Endpoints connected by a circuit, like an electrical circuit Signals flow both ways (full duplex) Associated with reserved bandwidth and buffer resources 4

5 Public Telephony (PSTN) History 1876 invention of telephone 1915 first transcontinental telephone (NY SF) 1920 s first automatic switches 1956 TAT-1 transatlantic cable (35 lines) 1962 digital transmission (T1) ESS analog switch 1974 Internet packet voice ESS digital switch 1980s Signaling System #7 (out-of-band) 1990s Advanced Intelligent Network (AIN) 5

6 Telephone System Overview Analog narrowband circuits: home-> central office 64 kb/s continuous transmission, with compression across oceans µ-law: 12-bit linear range -> 8-bit bytes Everything clocked a multiple of 125 s Clock synchronization framing errors AT&T: 136 toll switches in U.S. Interconnected by T1, T3 lines & SONET rings Call establishment out-of-band using packetswitched signaling system (SS7) 6

7 Telephony: Multiplexing Telephone Trunks between central offices carry hundreds of conversations: Can t run thick bundles! Send many calls on the same wire: multiplexing Analog multiplexing bandlimit call to 3.4 KHz and frequency shift onto higher bandwidth trunk Digital multiplexing: convert voice to samples 8000 samples/sec => call = 64 Kbps 7

8 Telephone Network Design Fully connected core simple routing telephone number is a hint about how to route a call But not for 800/888/700/900 numbers: these are pointers to a directory that translates them into regular numbers hierarchically allocated telephone number space 8

9 Telephone Network Design 9

10 Telephone Pieces: End Systems 10

11 Telephone Pieces: End Systems Transducers: key to carrying voice on wires Dialer Ringer Switch-hook 11

12 Last-Mile Transmission Environment Wire gauges:19, 22, 24, 26 gauge(smaller better) Diameters: 0.8, 0.6, 0.5, 0.4 mm (larger better) Various forms of noise: (twisting reduces noise) Bridged-tap noise: bit-energy diverted to extension phone sockets Crosstalk Ham radio AM broadcast Insertion loss: -140 dbm noise floor 100 million times more sensitive than normal modems Bandwidth range = 600 khz Notch effects in insertion loss due to bridged-taps Transmission PSD = -40dBm => 90 dbm budget 12

13 Both trans & reception circuits need two wires 2-wire vs 4-wire: Sidetones and Echoes 4 wires from every central office to home Alternative: Use same pair of wires for both transmission and reception Signal from transmission flows to receiver: sidetone Reverse Effect: received signal at end-system bounces back to CO (esp if delay > 20 ms): echo Solutions: balance circuit (attenuate side-tone) + echocancellation circuit (cancel echoes). 13

14 Pulse Dialing sends a pulse per digit collected by central office (CO) Interpreted by CO switching system to place call or activate special features (eg: call forwarding, prepaidcalls etc) Tone key press (feep) sends a pair of tones = digit also called Dual Tone Multifrequency (DTMF) CO supplies the power for ringing the bell. Standardized interface between CO and end-system => digital handsets, cordless/cellular phones 14

15 Telephone Pieces: Transmission Muxing Trunks between central offices carry hundreds of conversations Can t run thick bundles! Instead, send many calls on the same wire Multiplexing (a.ka. Sharing) Analog multiplexing Band-limit call to 3.4 KHz and frequency shift onto higher bandwidth trunk obsolete Digital multiplexing first convert voice to samples 1 sample = 8 bits of voice 8000 samples/sec => call = 64 Kbps 15

16 Transmission Multiplexing (contd) How to choose a sample? 256 quantization levels, logarithmically spaced (why?) sample value = amplitude of nearest quantization level Two choices of levels (µ law and A law) Time division multiplexing Trunk carries bits at a faster bit rate than inputs n input streams, each with a 1-byte buffer Output interleaves samples Need to serve all inputs in the time it takes one sample to arrive => output runs n times faster than input Overhead bits mark end of frame (why?) 16

17 Transmission Multiplexing Multiplexed trunks can be multiplexed further Need a standard! (why?) US/Japan standard is called Digital Signaling hierarchy (DS) Digital Signal Number Number of previous level circuits Number of voice circuits Bandwidth DS Kbps DS Mbps DS Mbps DS Mbps 17

18 Telephone Pieces: Switching 18

19 Telephone Pieces: Switching Problem: each user can potentially call any other user can t have (a billion) direct lines! Switches establish temporary circuits Switching systems come in two parts: switch and switch controller 19

20 Switching System Components 20

21 Switch: What does it do? Transfers data from an input to an output many ports (up to 200,000 simultaneous calls) need high speeds Some ways to switch: 1. space division switching: eg: crossbar if inputs (or crosspoints) are multiplexed, need a schedule (why?) 21

22 Crossbar Switching Elements 22

23 Switching (Contd) Another way to switch time division (time slot interchange or TSI) also needs a service schedule (why?) To build larger switches we combine space and time division switching elements 23

24 Telephone pieces: Signaling A switching system has a switch and a switch controller Switch controller is in the control plane does not touch voice samples Manages the network call routing (collect dialstring and forward call) alarms (ring bell at receiver) billing directory lookup (for 800/888 calls) 24

25 Signaling Switch controllers are special purpose computers Linked by their own internal computer network Common Channel Interoffice Signaling (CCIS) network Earlier design used in-band tones, but was hacked Also was very rigid (why?) Messages on CCIS conform to Signaling System 7 (SS7) 25

26 Signaling (contd) One of the main jobs of switch controller: keep track of state of every endpoint Key is state transition diagram 26

27 Telephony Routing of Signaled Calls Circuit-setup (I.e. the signaling call) is what is routed. Voice then follows route, and claims reserved resources. 3-level hierarchy, with a fully-connected core AT&T: 135 core switches with nearly 5 million circuits LECs may connect to multiple cores 27

28 Telephony Routing algorithm If endpoints are within same CO, directly connect If call is between COs in same LEC, use one-hop path between COs Otherwise send call to one of the cores Only major decision is at toll switch one-hop or two-hop path to the destination toll switch. Essence of telephony routing problem: which two-hop path to use if one-hop path is full (almost a static routing problem ) 28

29 Features of telephone routing Resource reservation aspects: Resource reservation is coupled with path reservation Connections need resources (same 64kbps) Signaling to reserve resources and the path Stable load Network built for voice only. Can predict pairwise load throughout the day Can choose optimal routes in advance Technology and economic aspects: Extremely reliable switches Why? End-systems (phones) dumb because computation was non-existent in early 1900s. Downtime is less than a few minutes per year => topology does not change dynamically 29

30 Features of telephone routing Source can learn topology and compute route Can assume that a chosen route is available as the signaling proceeds through the network Component reliability drove system reliability and hence acceptance of service by customers Simplified topology: Very highly connected network Hierarchy + full mesh at each level: simple routing High cost to achieve this degree of connectivity Organizational aspects: Single organization controls entire core Afford the scale economics to build expensive network Collect global statistics and implement global changes => Source-based, signaled, simple alternate-path routing 30

31 Telecommunications Regulation History FCC regulations cover telephony, cable, broadcast TV, wireless etc Common Carrier : provider offers conduit for a fee and does not control the content Customer controls content/destination of transmission & assumes criminal/civil responsibility for content Local monopolies formed by AT&T s acquisition of independent telephone companies in early 20 th century Regulation forced because they were deemed natural monopolies (only one player possible in market due to enormous sunk cost) FCC regulates interstate calls and state commissions regulate intra-state and local calls Bells independents interconnected & expanded FCC rulemaking process: Intent to act, solicitation of public comment etc 31

32 Deregulation of telephony 1960s-70s: gradual de-regulation of AT&T due to technological advances Terminal equipment could be owned by customers (CPE) => explosion in PBXs, fax machines, handsets Modified final judgement (MFJ): breakup of AT&T into ILECs (incumbent local exchange carrier) and IXC (inter-exchange carrier) part Long-distance opened to competition, only the local part regulated Equal access for IXCs to the ILEC network 1+ long-distance number introduced then 800-number portability: switching IXCs => retain 800 number 1995: removed price controls on AT&T 32

33 Telecom Act of 1996 Required ILECs to open their markets through unbundling of network elements (UNE-P), facilities ownership of CLECs. Today UNE-P is one of the most profitable for AT&T and other long-distance players in the local market: due to apparently below-cost regulated prices ILECs could compete in long-distance after demonstrating opening of markets Only now some ILECs are aggressively entering long distance markets CLECs failed due to a variety of reasons But long-distance prices have dropped precipitously (AT&T s customer unit revenue in 2002 was $11.3 B compared to 1999 rev of $23B) ILECs still retain over 90% of local market Wireless substitution has caused ILECs to develop wireless business units 33

34 US Telephone Network Structure (after 1984) 34

35 Exchange Area Network 35

36 IP Telephony, VoIP etc 36

37 IP Telephony: Overview IP Telephony: Why? Adding interactive multimedia to the web Being able to do basic telephony on IP with a variety of devices Basic IP telephony model Protocols: SIP, H.323, RTP, Coding schemes, MGCP, RTSP Future: Invisible IP telephony and control of appliances 37

38 Telephone Service Penetration in the US AT&T Divestiture 38

39 Trends: Price of Phone Calls: NY - London AT&T Divestiture 39

40 Trends: Data vs Voice Traffic Since we are building future networks for data, can we slowly junk the voice infrastructure and move over to IP? 40

41 Trends: Phone vs Data Revenues 41

42 Private Branch Exchange (PBX) Post-divestiture phenomenon External line Corporate/Campus Private Branch Exchange Telephone switch Another switch 7043 Corporate/Campus LAN Internet 42

43 IP Telephony: PBX Replacement 7040 Corporate/Campus External line Another campus PBX PBX LAN Internet LAN 43

44 Voice over Packet Market Forecast North America 44

45 Invisible Internet Telephony VoIP technology will appear in... Internet appliances home security cameras, web cams 3G mobile terminals fire alarms chat/im tools interactive multiplayer games 45

46 IPtel for appliances: Presence 46

47 Taxonomy of Speech Coders Speech Coders Waveform Coders Source Coders Time Domain: PCM, ADPCM Frequency Domain: e.g. Sub-band coder, Adaptive transform coder 47 Linear Predictive Coder Vocoder Waveform coders: attempts to preserve the signal waveform not speech specific (I.e. general A-to-D conv) PCM 64 kbps, ADPCM 32 kpbs, CVSDM 32 kbps Vocoders: Analyse speech, extract and transmit model parameters Use model parameters to synthesize speech LPC-10: 2.4 kbps Hybrids: Combine best of both Eg: CELP (used in GSM)

48 Speech Quality of Various Coders 48

49 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) 49

50 Pulse Amplitude Modulation (PAM) 50

51 Pulse Code Modulation (PCM) * PCM = PAM + quantization 51

52 Quantization 52

53 Companded PCM Small quantization intervals to small samples and large intervals for large samples Excellent quality for BOTH voice and data Moderate data rate (64 kbps) Moderate cost: used in T1 lines etc 53

54 Companding 54

55 How it works for T1 Lines Companding blocks are shared by all 16 channels 55

56 Adaptive Gain Encoding Automatic Gain control (AGC), but accounting for silence periods 56

57 Time Waveform of Voiced/Unvoiced Sound High correlation (0.85) between samples, cycles, pitch intervals etc 57

58 Differential PCM Exploits sample-to-sample correlation (0.85) => differences require fewer bits; feedback avoids cascading quantization errors 58

59 Delta Modulation Used in first-generation PBXs (switching was more sensitive to Digital conversion cost and less sensitive to quality or data rate) 59

60 Adaptive Predictive Coding Adapt both the prediction coefficients (alphas) and the estimates Based upon past or present samples => 20 db prediction gain 60

61 Subband Coding Frequency domain analysis of input instead of time-domain Analysis: adjust quantization based upon energy level of each band Eg: G.722 coder: 7kHz voice w/ 64 kbps 61

62 G.722 (7 khz) audio Codec 62

63 Recall: Taxonomy of Speech Coders Speech Coders Waveform Coders Source Coders Time Domain: PCM, ADPCM Frequency Domain: e.g. Sub-band coder, Adaptive transform coder 63 Linear Predictive Coder Vocoder Waveform coders: attempts to preserve the signal waveform not speech specific. PCM 64 kbps, ADPCM 32 kpbs, CVSDM 32 kbps Vocoders: Analyse speech, extract and transmit model parameters Use model parameters to synthesize speech LPC-10: 2.4 kbps Hybrids: Combine best of both Eg: CELP

64 Vocoders Encode only perceptually important aspects of speech w/ fewer bits than waveform coders: eg: power spectrum vs time-domain Rensselaer accuracy Polytechnic Institute 64

65 LPC Analysis/Synthesis 65

66 Speech Generation in LPC 66

67 Multi-pulse LPC 67

68 CELP Encoder 68

69 Example: GSM Digital Speech Coding PCM: 64kbps too wasteful for wireless Regular Pulse Excited -- Linear Predictive Coder (RPE--LPC) with a Long Term Predictor loop. Subjective speech quality and complexity (related to cost, processing delay, and power) Information from previous samples used to predict the current sample: linear function. The coefficients, plus an encoded form of the residual (predicted - actual sample), represent the signal. 20 millisecond samples: each encoded as 260 bits =>13 kbps (Full-Rate coding). 69

70 Speech Quality of Various Coders 70

71 Speech Quality (Contd) 71

72 VoIP Camps Conferencing Industry Netheads IP over Everything Circuit switch engineers We over IP Convergence ITU standards 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 Our focus 72

73 Internet Multimedia Protocol Stack 73

74 IP Telephony Protocols: SIP, RTP Session Initiation Protocol - SIP Contact office.com asking for bob Locate Bob s current phone and ring Bob picks up the ringing phone Real time Transport Protocol - RTP Send and receive audio packets 74

75 Internet Telephony Protocols: H

76 H.323 (contd) Terminals, Gateways, Gatekeepers, and Multipoint Control Units (MCUs) 76

77 H.323 vs SIP Typical UserAgent Protocol stack for Internet Terminal Control/Devices Codecs Q.931 H.245 RAS RTCP RTP TPKT TCP UDP IP and lower layers Terminal Control/Devices Codecs SIP SDP RTCP RTP Transport Layer 77

78 SIP vs H.323 Text based request response SDP (media types and media transport address) Server roles: registrar, proxy, redirect Binary ASN.1 PER encoding Sub-protocols: H.245, H.225 (Q.931, RAS, RTP/RTCP), H.450.x... H.323 Gatekeeper - Both use RTP/RTCP over UDP/IP - H.323 perceived as heavyweight 78

79 Light-weight signaling: Session Initiation Protocol (SIP) IETF MMUSIC working group Light-weight generic signaling protocol Part of IETF conference control architecture: SAP for Internet TV Guide announcements RTSP for media-on-demand SDP for describing media others: malloc, multicast, conference bus,... Post-dial delay: 1.5 round-trip time (with UDP) Network-protocol independent: UDP or TCP (or AAL5 or X.25) 79

80 SDP: Session Description Protocol Not really a protocol describes data carried by other protocols Used by SAP, SIP, RTSP, H.332, PINT. Eg: v=0 o=g.bell IN IP s=come here, Watson! u= c=in IP b=ct:64 t= k=clear:manhole cover m=audio 3456 RTP/AVP 96 a=rtpmap:96 VDVI/8000/1 m=video 3458 RTP/AVP 31 m=application udp wb 80

81 SIP functionality IETF-standardized peer-to-peer signaling protocol (RFC 2543): Locate user given -style address Setup session (call) (Re)-negotiate call parameters Manual and automatic forwarding Personal mobility: different terminal, same identifier Call center: reach first (load distribution) or reach all (department conference) Terminate and transfer calls 81

82 SIP Addresses Food Chain 82

83 SIP components UAC: user-agent client (caller application) UAS: user-agent server à accept, redirect, refuse call redirect server: redirect requests proxy server: server + client registrar: track user locations user agent = UAC + UAS often combine registrar + (proxy or redirect server) 83

84 IP SIP Phones and Adaptors 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 5 Palm control

85 Telephone Telephone switch T1/E1 RTP/SIP Cisco 2600 gateway sipc SIP-based Architecture sipconf SIP conference server sipd SIP proxy, redirect server rtspd RTSP media server sipum SIP/RTSP Unified messaging SQL database RTSP e*phone Quicktime RTSP clients Web server Hardware Internet (SIP) phones Web based configuration Software SIP user agents sip323 SIPH.32 3 converto r 85 H.323 NetMeeting

86 Example Call Bob signs up for the service from the web as He registers from multiple phones Alice tries to reach Bob INVITE Call Bob sipd SIP proxy, redirect server sipd canonicalizes the destination to sipd rings both e*phone and sipc Bob accepts the call from sipc and starts talking SQL database Web server e*phone Web based configuration sipc ecse.rpi.edu Hardware Internet (SIP) phones Software SIP user agents 86

87 PSTN to IP Call PSTN External T1/CAS PBX Internal T1/CAS (Ext: ) Gateway 1 Call Call 7134 Ethernet Regular phone (internal) 5 3 SIP server SQL database sipc Bob s phone sipd => bob 87

88 IP to PSTN Call PSTN 5 External T1/CAS Call PBX Internal T1/CAS Call Gateway ( ) 3 Ethernet Regular phone (internal, 7054) 1 Bob calls SIP server SQL database sipc 2 sipd Use 88

89 Traditional voice mail system Dial Alice Phone is ringing.. The person is not available now please leave a message Your voice message... Disconnect Bob Bob can listen to his voice mails by dialing some number. 89

90 SIP-based Voic Architecture Bob INVITE INVITE phone1.office.com Alice REGISTER INVITE 90 vm.office.com The voice mail server registers with the SIP proxy, sipd Alice calls through SIP proxy. SIP proxy forks the request to Bob s phone as well as to a voic server.

91 Voic Architecture Bob phone1.office.com; CANCEL Alice 200 OK 200 OK RTP/RTCP After 10 seconds vm contacts the RTSP server for recording. vm accepts the call. Sipd cancels the other branch and......accepts the call from Alice. Now user message gets recorded 91 v-mail vm.office.com; SETUP rtspd

92 SIP-H.323: Interworking Problems Eg: Call setup translation H.323 SIP Q.931 SETUP Q.931 CONNECT Terminal Capabilities Terminal Capabilities Open Logical Channel Open Logical Channel Destination address Media capabilities (audio/video) Media transport address (RTP/RTCP receive) INVITE 200 OK ACK H.323: Multi-stage dialing 92

93 MGCP and Megaco Media Gateway Controller Protocol (RFC 2705) Controlling Telephony Gateways from external call control elements called media gateway controllers (MGC) or call agents Gateways: Eg: RGW : physical interfaces between VoIP network and residences Call control "intelligence" is outside the gateways and handled by external call control elements Goal: scalable gateways between IP telephony and PSTN Successor to MGCP: H.248/Megaco 93

94 MGCP Architecture Goal: large-scale phone-to-phone VoIP deployments RGW: Residential Gateway TGW: Trunk Gateway 94

95 Summary Telephony and IP Telephony Protocols: SIP, SDP, H.323, MCGP Example operation and services: Calls, voice mail etc Future: Integration with Web and long-term replacement for current telephone systems 95

The Telephone Network. An Engineering Approach to Computer Networking

The Telephone Network. An Engineering Approach to Computer Networking The Telephone Network An Engineering Approach to Computer Networking Is it a computer network? Specialized to carry voice Also carries telemetry video fax modem calls Internally, uses digital samples Switches

More information

Towards Junking the PBX: Deploying IP Telephony. What is a PBX?

Towards Junking the PBX: Deploying IP Telephony. What is a PBX? Towards Junking the : Deploying IP Wenyu Jiang, Jonathan Lennox, Henning Schulzrinne and Kundan Singh Columbia University {wenyu,lennox,hgs,kns10}@cs.columbia.edu We describe our departmental IP telephony

More information

SIP Trunking and Voice over IP

SIP Trunking and Voice over IP SIP Trunking and Voice over IP Agenda What is SIP Trunking? SIP Signaling How is Voice encoded and transported? What are the Voice over IP Impairments? How is Voice Quality measured? VoIP Technology Confidential

More information

Voice over IP Quality of Service and Reliability. Course Outline

Voice over IP Quality of Service and Reliability. Course Outline Voice over IP Quality of Service and Reliability David Tipper Associate Professor Department of Information Science and Telecommunications University of Pittsburgh tipper@tele.pitt.edu http://www.sis.pitt.edu/~dtipper/tipper.html

More information

Three Network Technologies

Three Network Technologies Three Network Technologies Network The largest worldwide computer network, specialized for voice ing technique: Circuit-switching Internet The global public information infrastructure for data ing technique:

More information

Session Initiation Protocol (SIP) The Emerging System in IP Telephony

Session Initiation Protocol (SIP) The Emerging System in IP Telephony Session Initiation Protocol (SIP) The Emerging System in IP Telephony Introduction Session Initiation Protocol (SIP) is an application layer control protocol that can establish, modify and terminate multimedia

More information

Overview of Voice Over Internet Protocol

Overview of Voice Over Internet Protocol Overview of Voice Over Internet Protocol Purva R. Rajkotia, Samsung Electronics November 4,2004 Overview of Voice Over Internet Protocol Presentation Outline History of VoIP What is VoIP? Components of

More information

Voice over IP. Abdus Salam ICTP, February 2004 School on Digital Radio Communications for Research and Training in Developing Countries

Voice over IP. Abdus Salam ICTP, February 2004 School on Digital Radio Communications for Research and Training in Developing Countries Voice over IP Abdus Salam ICTP, February 2004 School on Digital Radio Communications for Research and Training in Developing Countries Ermanno Pietrosemoli Latin American Networking School (Fundación EsLaRed)

More information

An Introduction to VoIP Protocols

An Introduction to VoIP Protocols An Introduction to VoIP Protocols www.netqos.com Voice over IP (VoIP) offers the vision of a converged network carrying multiple types of traffic (voice, video, and data, to name a few). To carry out this

More information

Indepth Voice over IP and SIP Networking Course

Indepth Voice over IP and SIP Networking Course Introduction SIP is fast becoming the Voice over IP protocol of choice. During this 3-day course delegates will examine SIP technology and architecture and learn how a functioning VoIP service can be established.

More information

Voice over IP Basics for IT Technicians

Voice over IP Basics for IT Technicians Voice over IP Basics for IT Technicians White Paper Executive summary The IP phone is coming or has arrived on desk near you. The IP phone is not a PC, but does have a number of hardware and software elements

More information

Need for Signaling and Call Control

Need for Signaling and Call Control Need for Signaling and Call Control VoIP Signaling In a traditional voice network, call establishment, progress, and termination are managed by interpreting and propagating signals. Transporting voice

More information

Course 4: IP Telephony and VoIP

Course 4: IP Telephony and VoIP Course 4: IP Telephony and VoIP Telecommunications Technical Curriculum Program 3: Voice Knowledge 6/9/2009 1 Telecommunications Technical Curriculum Program 1: General Industry Knowledge Course 1: General

More information

Internet Technology Voice over IP

Internet Technology Voice over IP Internet Technology Voice over IP Peter Gradwell BT Advert from 1980s Page 2 http://www.youtube.com/v/o0h65_pag04 Welcome to Gradwell Gradwell provides technology for every line on your business card Every

More information

Voice over IP (VoIP) Basics for IT Technicians

Voice over IP (VoIP) Basics for IT Technicians Voice over IP (VoIP) Basics for IT Technicians VoIP brings a new environment to the network technician that requires expanded knowledge and tools to deploy and troubleshoot IP phones. This paper provides

More information

Understanding Voice over IP Protocols

Understanding Voice over IP Protocols Understanding Voice over IP Protocols Cisco Systems Service Provider Solutions Engineering February, 2002 1 Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP

More information

Encapsulating Voice in IP Packets

Encapsulating Voice in IP Packets Encapsulating Voice in IP Packets Major VoIP Protocols This topic defines the major VoIP protocols and matches them with the seven layers of the OSI model. Major VoIP Protocols 15 The major VoIP protocols

More information

Receiving the IP packets Decoding of the packets Digital-to-analog conversion which reproduces the original voice stream

Receiving the IP packets Decoding of the packets Digital-to-analog conversion which reproduces the original voice stream Article VoIP Introduction Internet telephony refers to communications services voice, fax, SMS, and/or voice-messaging applications that are transported via the internet, rather than the public switched

More information

Data Communications & Computer Networks. Circuit and Packet Switching

Data Communications & Computer Networks. Circuit and Packet Switching Data Communications & Computer Networks Chapter 9 Circuit and Packet Switching Fall 2008 Agenda Preface Circuit Switching Softswitching Packet Switching Home Exercises ACOE312 Circuit and packet switching

More information

Understanding Voice over IP

Understanding Voice over IP Introduction Understanding Voice over IP For years, many different data networking protocols have existed, but now, data communications has firmly found its home in the form of IP, the Internet Protocol.

More information

ATA: An Analogue Telephone Adapter is used to connect a standard telephone to a high-speed modem to facilitate VoIP and/or calls over the Internet.

ATA: An Analogue Telephone Adapter is used to connect a standard telephone to a high-speed modem to facilitate VoIP and/or calls over the Internet. KEY VOIP TERMS 1 ACD: Automatic Call Distribution is a system used to determine how incoming calls are routed. When the ACD system receives an incoming call it follows user-defined specifications as to

More information

B12 Troubleshooting & Analyzing VoIP

B12 Troubleshooting & Analyzing VoIP B12 Troubleshooting & Analyzing VoIP Phillip Sherlock Shade, Senior Forensics / Network Engineer Merlion s Keep Consulting phill.shade@gmail.com Phillip Sherlock Shade (Phill) phill.shade@gmail.com Phillip

More information

Packetized Telephony Networks

Packetized Telephony Networks Packetized Telephony Networks Benefits of Packet Telephony Networks Traditionally, the potential savings on long-distance costs was the driving force behind the migration to converged voice and data networks.

More information

159.334 Computer Networks. Voice over IP (VoIP) Professor Richard Harris School of Engineering and Advanced Technology (SEAT)

159.334 Computer Networks. Voice over IP (VoIP) Professor Richard Harris School of Engineering and Advanced Technology (SEAT) Voice over IP (VoIP) Professor Richard Harris School of Engineering and Advanced Technology (SEAT) Presentation Outline Basic IP phone set up The SIP protocol Computer Networks - 1/2 Learning Objectives

More information

Voice over IP (VoIP) Overview. Introduction. David Feiner ACN 2004. Introduction VoIP & QoS H.323 SIP Comparison of H.323 and SIP Examples

Voice over IP (VoIP) Overview. Introduction. David Feiner ACN 2004. Introduction VoIP & QoS H.323 SIP Comparison of H.323 and SIP Examples Voice over IP (VoIP) David Feiner ACN 2004 Overview Introduction VoIP & QoS H.323 SIP Comparison of H.323 and SIP Examples Introduction Voice Calls are transmitted over Packet Switched Network instead

More information

Media Gateway Controller RTP

Media Gateway Controller RTP 1 Softswitch Architecture Interdomain protocols Application Server Media Gateway Controller SIP, Parlay, Jain Application specific Application Server Media Gateway Controller Signaling Gateway Sigtran

More information

Troubleshooting Voice Over IP with WireShark

Troubleshooting Voice Over IP with WireShark Hands-On Course Description Voice over IP is being widely implemented both within companies and across the Internet. The key problems with IP voice services are maintaining the quality of the voice service

More information

Voice over IP Protocols And Compression Algorithms

Voice over IP Protocols And Compression Algorithms University of Tehran Electrical and Computer Engineering School SI Lab. Weekly Presentations Voice over IP Protocols And Compression Algorithms Presented by: Neda Kazemian Amiri Agenda Introduction to

More information

Introduction to Packet Voice Technologies and VoIP

Introduction to Packet Voice Technologies and VoIP Introduction to Packet Voice Technologies and VoIP Cisco Networking Academy Program Halmstad University Olga Torstensson 035-167575 olga.torstensson@ide.hh.se IP Telephony 1 Traditional Telephony 2 Basic

More information

VoIP Analysis Fundamentals with Wireshark. Phill Shade (Forensic Engineer Merlion s Keep Consulting)

VoIP Analysis Fundamentals with Wireshark. Phill Shade (Forensic Engineer Merlion s Keep Consulting) VoIP Analysis Fundamentals with Wireshark Phill Shade (Forensic Engineer Merlion s Keep Consulting) 1 Phillip D. Shade (Phill) phill.shade@gmail.com Phillip D. Shade is the founder of Merlion s Keep Consulting,

More information

5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues.

5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues. 5. DEPLOYMENT ISSUES Having described the fundamentals of VoIP and underlying IP infrastructure, let s address deployment issues. 5.1 LEGACY INTEGRATION In most cases, enterprises own legacy PBX systems,

More information

Integrate VoIP with your existing network

Integrate VoIP with your existing network Integrate VoIP with your existing network As organisations increasingly recognise and require the benefits voice over Internet Protocol (VoIP) offers, they stop asking "Why?" and start asking "How?". A

More information

TSIN02 - Internetworking

TSIN02 - Internetworking TSIN02 - Internetworking Lecture 9: SIP and H323 Literature: Understand the basics of SIP and it's architecture Understand H.323 and how it compares to SIP Understand MGCP (MEGACO/H.248) SIP: Protocol

More information

Cisco Networks (ONT) 2006 Cisco Systems, Inc. All rights reserved.

Cisco Networks (ONT) 2006 Cisco Systems, Inc. All rights reserved. Optimizing Converged Cisco Networks (ONT) reserved. Lesson 2.4: Calculating Bandwidth Requirements for VoIP reserved. Objectives Describe factors influencing encapsulation overhead and bandwidth requirements

More information

Network Technologies

Network Technologies Network Technologies Telephone Networks IP Networks ATM Networks Three Network Technologies Telephone Network The largest worldwide computer network, specialized for voice ing technique: Circuit-switching

More information

VIDEOCONFERENCING. Video class

VIDEOCONFERENCING. Video class VIDEOCONFERENCING Video class Introduction What is videoconferencing? Real time voice and video communications among multiple participants The past Channelized, Expensive H.320 suite and earlier schemes

More information

White paper. SIP An introduction

White paper. SIP An introduction White paper An introduction Table of contents 1 Introducing 3 2 How does it work? 3 3 Inside a normal call 4 4 DTMF sending commands in sip calls 6 5 Complex environments and higher security 6 6 Summary

More information

ABSTRACT. Keywords: VoIP, PSTN/IP interoperability, SIP, H.323, RTP, PBX, SDP, MGCP, Westplan. 1. INTRODUCTION

ABSTRACT. Keywords: VoIP, PSTN/IP interoperability, SIP, H.323, RTP, PBX, SDP, MGCP, Westplan. 1. INTRODUCTION Implementing a Voice Over Internet (Voip) Telephony System Md. Manzoor Murshed Final Project Report for the course CprE550: Distributed Systems and Middleware ABSTRACT This Project is to describe the architecture

More information

4. H.323 Components. VOIP, Version 1.6e T.O.P. BusinessInteractive GmbH Page 1 of 19

4. H.323 Components. VOIP, Version 1.6e T.O.P. BusinessInteractive GmbH Page 1 of 19 4. H.323 Components VOIP, Version 1.6e T.O.P. BusinessInteractive GmbH Page 1 of 19 4.1 H.323 Terminals (1/2)...3 4.1 H.323 Terminals (2/2)...4 4.1.1 The software IP phone (1/2)...5 4.1.1 The software

More information

Voice Over Internet Protocol (VoIP) Issues and Challenges William McCrum mccrum.william@ic.gc.ca

Voice Over Internet Protocol (VoIP) Issues and Challenges William McCrum mccrum.william@ic.gc.ca Voice Over Internet Protocol (VoIP) Issues and Challenges William McCrum Phone: +1 613-990-4493 Fax: Email: +1 613-957-8845 mccrum.william@ic.gc.ca Content Network Evolution and drivers VoIP Realizations

More information

Lehrstuhl für Informatik 4 Kommunikation und verteilte Systeme

Lehrstuhl für Informatik 4 Kommunikation und verteilte Systeme Chapter 2: Representation of Multimedia Data Chapter 3: Multimedia Systems Communication Aspects and Services Multimedia Applications and Communication Protocols Quality of Service and Resource Management

More information

Optimizing Converged Cisco Networks (ONT)

Optimizing Converged Cisco Networks (ONT) Optimizing Converged Cisco Networks (ONT) Module 2: Cisco VoIP Implementations (Deploy) Calculating Bandwidth Requirements for VoIP Objectives Describe factors influencing encapsulation overhead and bandwidth

More information

Special Module on Media Processing and Communication

Special Module on Media Processing and Communication Special Module on Media Processing and Communication Multimedia Communication Fundamentals Dayalbagh Educational Institute (DEI) Dayalbagh Agra PHM 961 Indian Institute of Technology Delhi (IITD) New Delhi

More information

White Paper: Voice Over IP Networks

White Paper: Voice Over IP Networks FREE FREE One One Hour Hour VoIPonline VoIPonline Seminar TM Seminar TM For additional information contact: Terry Shugart - tshugart@analogic.com http://www.analogic.com/cti TEL: 978-977-3000 FAX: 978-977-6813

More information

Combining Voice over IP with Policy-Based Quality of Service

Combining Voice over IP with Policy-Based Quality of Service TechBrief Extreme Networks Introduction Combining Voice over IP with Policy-Based Quality of Service Businesses have traditionally maintained separate voice and data networks. A key reason for this is

More information

PacketizerTM. Overview of H.323 http://www.packetizer.com/voip/h323/papers/ Paul E. Jones. Rapporteur, ITU-T Q2/SG16 paulej@packetizer.

PacketizerTM. Overview of H.323 http://www.packetizer.com/voip/h323/papers/ Paul E. Jones. Rapporteur, ITU-T Q2/SG16 paulej@packetizer. A resource for packet-switched conversational protocols Overview of H.323 http:///voip/h323/papers/ Paul E. Jones Rapporteur, ITU-T Q2/SG16 paulej@packetizer.com June 2004 Copyright 2004 Executive Summary

More information

T1 Networking Made Easy

T1 Networking Made Easy T1 Networking Made Easy 1 THE T1 CARRIER 3 WHAT DOES A T1 LOOK LIKE? 3 T1 BANDWIDTH 3 T1 PHYSICAL CHARACTERISTICS 4 T1 FRAMING 5 LINE CODE 6 T1 NETWORKING 6 TELCOS 6 PSTN ACCESS WITH A T1 8 SUMMARY OF

More information

Contents Introduction Why Fax over IP? How Real-time Fax over IP works Implementation with MessagePlus/Open Summary. About this document

Contents Introduction Why Fax over IP? How Real-time Fax over IP works Implementation with MessagePlus/Open Summary. About this document Fax over IP Contents Introduction Why Fax over IP? How Real-time Fax over IP works Implementation with MessagePlus/Open Summary About this document This document describes how Fax over IP works in general

More information

Hands on VoIP. Content. Tel +44 (0) 845 057 0176 enquiries@protelsolutions.co.uk. Introduction

Hands on VoIP. Content. Tel +44 (0) 845 057 0176 enquiries@protelsolutions.co.uk. Introduction Introduction This 4-day course offers a practical introduction to 'hands on' VoIP engineering. Voice over IP promises to reduce your telephony costs and provides unique opportunities for integrating voice

More information

Applied Networks & Security

Applied Networks & Security Applied Networks & Security VoIP with Critical Analysis http://condor.depaul.edu/~jkristof/it263/ John Kristoff jtk@depaul.edu IT 263 Spring 2006/2007 John Kristoff - DePaul University 1 Critical analysis

More information

Voice---is analog in character and moves in the form of waves. 3-important wave-characteristics:

Voice---is analog in character and moves in the form of waves. 3-important wave-characteristics: Voice Transmission --Basic Concepts-- Voice---is analog in character and moves in the form of waves. 3-important wave-characteristics: Amplitude Frequency Phase Voice Digitization in the POTS Traditional

More information

Contents. Specialty Answering Service. All rights reserved.

Contents. Specialty Answering Service. All rights reserved. Contents 1. Introduction to Session Internet Protocol... 2 2. History, Initiation & Implementation... 3 3. Development & Applications... 4 4. Function & Capability... 5 5. SIP Clients & Servers... 6 5.1.

More information

Traditional Telephony

Traditional Telephony Traditional Telephony Basic Components of a Telephony Network This topic introduces the components of traditional telephony networks. Basic Components of a Telephony Network 3 A number of components must

More information

VoIP. Overview. Jakob Aleksander Libak jakobal@ifi.uio.no. Introduction Pros and cons Protocols Services Conclusion

VoIP. Overview. Jakob Aleksander Libak jakobal@ifi.uio.no. Introduction Pros and cons Protocols Services Conclusion VoIP Jakob Aleksander Libak jakobal@ifi.uio.no 1 Overview Introduction Pros and cons Protocols Services Conclusion 2 1 Introduction Voice over IP is routing of voice conversations over the internet or

More information

IP Telephony Deployment Models

IP Telephony Deployment Models CHAPTER 2 Sections in this chapter address the following topics: Single Site, page 2-1 Multisite Implementation with Distributed Call Processing, page 2-3 Design Considerations for Section 508 Conformance,

More information

Curso de Telefonía IP para el MTC. Sesión 2 Requerimientos principales. Mg. Antonio Ocampo Zúñiga

Curso de Telefonía IP para el MTC. Sesión 2 Requerimientos principales. Mg. Antonio Ocampo Zúñiga Curso de Telefonía IP para el MTC Sesión 2 Requerimientos principales Mg. Antonio Ocampo Zúñiga Factors Affecting Audio Clarity Fidelity: Audio accuracy or quality Echo: Usually due to impedance mismatch

More information

Voice Over IP. Priscilla Oppenheimer www.priscilla.com

Voice Over IP. Priscilla Oppenheimer www.priscilla.com Voice Over IP Priscilla Oppenheimer www.priscilla.com Objectives A technical overview of the devices and protocols that enable Voice over IP (VoIP) Demo Packet8 and Skype Discuss network administrator

More information

Unit 23. RTP, VoIP. Shyam Parekh

Unit 23. RTP, VoIP. Shyam Parekh Unit 23 RTP, VoIP Shyam Parekh Contents: Real-time Transport Protocol (RTP) Purpose Protocol Stack RTP Header Real-time Transport Control Protocol (RTCP) Voice over IP (VoIP) Motivation H.323 SIP VoIP

More information

Simple Voice over IP (VoIP) Implementation

Simple Voice over IP (VoIP) Implementation Simple Voice over IP (VoIP) Implementation ECE Department, University of Florida Abstract Voice over IP (VoIP) technology has many advantages over the traditional Public Switched Telephone Networks. In

More information

TECHNICAL CHALLENGES OF VoIP BYPASS

TECHNICAL CHALLENGES OF VoIP BYPASS TECHNICAL CHALLENGES OF VoIP BYPASS Presented by Monica Cultrera VP Software Development Bitek International Inc 23 rd TELELCOMMUNICATION CONFERENCE Agenda 1. Defining VoIP What is VoIP? How to establish

More information

Mobile VoIP: Managing, scheduling and refining voice packets to and from mobile phones

Mobile VoIP: Managing, scheduling and refining voice packets to and from mobile phones Mobile VoIP: Managing, scheduling and refining voice packets to and from mobile phones MOHAMMAD ABDUS SALAM Student ID: 01201023 TAPAN BISWAS Student ID: 01201003 \ Department of Computer Science and Engineering

More information

(Refer Slide Time: 6:17)

(Refer Slide Time: 6:17) Digital Video and Picture Communication Prof. S. Sengupta Department of Electronics and Communication Engineering Indian Institute of Technology, Kharagpur Lecture - 39 Video Conferencing: SIP Protocol

More information

VoIP Bandwidth Considerations - design decisions

VoIP Bandwidth Considerations - design decisions VoIP Bandwidth Considerations - design decisions When calculating the bandwidth requirements for a VoIP implementation the two main protocols are: a signalling protocol such as SIP, H.323, SCCP, IAX or

More information

Voice over IP: Issues and Challenges

Voice over IP: Issues and Challenges Voice over IP: Issues and Challenges IP is now at Columbus, OH 43210 Jain@cse.ohio-State.Edu Washington University in Saint Louis Jain@cse.wustl.edu http://www.cse.ohio-state.edu/~jain/ http://www.cse.wustl.edu/~jain/

More information

IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week

IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week Course Title: No. of Hours: IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week 1 Course Duration: 3 Months (12weeks) No. Of Hours: 7 Hrs./Day- 5 days/week.

More information

NTP VoIP Platform: A SIP VoIP Platform and Its Services

NTP VoIP Platform: A SIP VoIP Platform and Its Services NTP VoIP Platform: A SIP VoIP Platform and Its Services Speaker: Dr. Chai-Hien Gan National Chiao Tung University, Taiwan Email: chgan@csie.nctu.edu.tw Date: 2006/05/02 1 Outline Introduction NTP VoIP

More information

Network administrators must be aware that delay exists, and then design their network to bring end-to-end delay within acceptable limits.

Network administrators must be aware that delay exists, and then design their network to bring end-to-end delay within acceptable limits. Delay Need for a Delay Budget The end-to-end delay in a VoIP network is known as the delay budget. Network administrators must design a network to operate within an acceptable delay budget. This topic

More information

Operation Manual Voice Overview (Voice Volume) Table of Contents

Operation Manual Voice Overview (Voice Volume) Table of Contents Operation Manual Voice Over (Voice Volume) Table of Contents Table of Contents Chapter 1 Voice Over... 1-1 1.1 Introduction to VoIP... 1-1 1.1.1 VoIP System... 1-1 1.1.2 Basic VoIP Call Flow... 1-2 1.1.3

More information

ETM System SIP Trunk Support Technical Discussion

ETM System SIP Trunk Support Technical Discussion ETM System SIP Trunk Support Technical Discussion Release 6.0 A product brief from SecureLogix Corporation Rev C SIP Trunk Support in the ETM System v6.0 Introduction Today s voice networks are rife with

More information

Introduction to VoIP Technology

Introduction to VoIP Technology Lesson 1 Abstract Introduction to VoIP Technology 2012. 01. 06. This first lesson of contains the basic knowledge about the terms and processes concerning the Voice over IP technology. The main goal of

More information

Glossary of Telco Terms

Glossary of Telco Terms Glossary of Telco Terms Access Generally refers to the connection between your business and the public phone network, or between your business and another dedicated location. A large portion of your business

More information

Communication Systems SIP

Communication Systems SIP Communication Systems SIP Computer Science Organization I. Data and voice communication in IP networks II. Security issues in networking III. Digital telephony networks and voice over IP 2 Part 3 Digital,

More information

internet technologies and standards

internet technologies and standards Institute of Telecommunications Warsaw University of Technology 2015 internet technologies and standards Piotr Gajowniczek Andrzej Bąk Michał Jarociński multimedia in the Internet Voice-over-IP multimedia

More information

Multimedia Communications Voice over IP

Multimedia Communications Voice over IP Multimedia Communications Voice over IP Anandi Giridharan Electrical Communication Engineering, Indian Institute of Science, Bangalore 560012, India Voice over IP (Real time protocols) Internet Telephony

More information

VoIP with SIP. Session Initiation Protocol RFC-3261/RFC-2543. Tasuka@Tailyn.com.tw

VoIP with SIP. Session Initiation Protocol RFC-3261/RFC-2543. Tasuka@Tailyn.com.tw VoIP with SIP Session Initiation Protocol RFC-3261/RFC-2543 Tasuka@Tailyn.com.tw 1 Legacy Telephone 2 Legacy Telephone 2 Legacy Telephone 2 Legacy Telephone 2 Legacy Telephone 2 Legacy Telephone 2 Legacy

More information

Internet Working 15th lecture (last but one) Chair of Communication Systems Department of Applied Sciences University of Freiburg 2005

Internet Working 15th lecture (last but one) Chair of Communication Systems Department of Applied Sciences University of Freiburg 2005 15th lecture (last but one) Chair of Communication Systems Department of Applied Sciences University of Freiburg 2005 1 43 administrational stuff Next Thursday preliminary discussion of network seminars

More information

Application Notes. Contents. Overview. Introduction. Echo in Voice over IP Systems VoIP Performance Management

Application Notes. Contents. Overview. Introduction. Echo in Voice over IP Systems VoIP Performance Management Application Notes Title Series Echo in Voice over IP Systems VoIP Performance Management Date January 2006 Overview This application note describes why echo occurs, what effects it has on voice quality,

More information

Broadband Networks. Prof. Dr. Abhay Karandikar. Electrical Engineering Department. Indian Institute of Technology, Bombay. Lecture - 29.

Broadband Networks. Prof. Dr. Abhay Karandikar. Electrical Engineering Department. Indian Institute of Technology, Bombay. Lecture - 29. Broadband Networks Prof. Dr. Abhay Karandikar Electrical Engineering Department Indian Institute of Technology, Bombay Lecture - 29 Voice over IP So, today we will discuss about voice over IP and internet

More information

Overview ENUM ENUM. VoIP Introduction (2/2) VoIP Introduction (1/2)

Overview ENUM ENUM. VoIP Introduction (2/2) VoIP Introduction (1/2) Overview Voice-over over-ip (VoIP) ENUM VoIP Introduction Basic PSTN Concepts and SS7 Old Private Telephony Solutions Internet Telephony and Services VoIP-PSTN Interoperability IP PBX Network Convergence

More information

A Comparative Study of Signalling Protocols Used In VoIP

A Comparative Study of Signalling Protocols Used In VoIP A Comparative Study of Signalling Protocols Used In VoIP Suman Lasrado *1, Noel Gonsalves *2 Asst. Prof, Dept. of MCA, AIMIT, St. Aloysius College (Autonomous), Mangalore, Karnataka, India Student, Dept.

More information

Intel NetStructure Host Media Processing Software Release 1.0 for the Windows * Operating System

Intel NetStructure Host Media Processing Software Release 1.0 for the Windows * Operating System Datasheet Intel NetStructure Host Media Processing Software Release 1.0 for the Windows * Operating System Media Processing Software That Can Be Used To Build Cost-Effective IP Media Servers Features Benefits

More information

Software Engineering 4C03 VoIP: The Next Telecommunication Frontier

Software Engineering 4C03 VoIP: The Next Telecommunication Frontier Software Engineering 4C03 VoIP: The Next Telecommunication Frontier Rudy Muslim 0057347 McMaster University Computing and Software Department Hamilton, Ontario Canada Introduction Voice over Internet Protocol

More information

SIP (Session Initiation Protocol) Technical Overview. Presentation by: Kevin M. Johnson VP Engineering & Ops

SIP (Session Initiation Protocol) Technical Overview. Presentation by: Kevin M. Johnson VP Engineering & Ops SIP (Session Initiation Protocol) Technical Overview Presentation by: Kevin M. Johnson VP Engineering & Ops Page 1 Who are we? Page 2 Who are we? Workforce Automation Software Developer Page 3 Who are

More information

Voice over IP. Presentation Outline. Objectives

Voice over IP. Presentation Outline. Objectives Voice over IP Professor Richard Harris Presentation Outline Brief overview of VoIP and applications Challenges of VoIP IP Support for Voice Protocols used for VoIP (current views) RTP RTCP RSVP H.323 Semester

More information

Voice over IP. Raj Jain. The Ohio State University Columbus, OH 43210 Jain@cse.ohio-State.Edu http://www.cse.ohio-state.edu/~jain/ Raj Jain

Voice over IP. Raj Jain. The Ohio State University Columbus, OH 43210 Jain@cse.ohio-State.Edu http://www.cse.ohio-state.edu/~jain/ Raj Jain Voice over IP IP The Ohio State University Columbus, OH 43210 Jain@cse.ohio-State.Edu http://www.cse.ohio-state.edu/~jain/ 1 Overview Sample Products and Services 13 Technical Issues 4 Other Issues H.323

More information

Analog vs. Digital Transmission

Analog vs. Digital Transmission Analog vs. Digital Transmission Compare at two levels: 1. Data continuous (audio) vs. discrete (text) 2. Signaling continuously varying electromagnetic wave vs. sequence of voltage pulses. Also Transmission

More information

Digital Speech Coding

Digital Speech Coding Digital Speech Processing David Tipper Associate Professor Graduate Program of Telecommunications and Networking University of Pittsburgh Telcom 2720 Slides 7 http://www.sis.pitt.edu/~dtipper/tipper.html

More information

Clearing the Way for VoIP

Clearing the Way for VoIP Gen2 Ventures White Paper Clearing the Way for VoIP An Alternative to Expensive WAN Upgrades Executive Overview Enterprises have traditionally maintained separate networks for their voice and data traffic.

More information

Voice over IP (VoIP) for Telephony. Advantages of VoIP Migration for SMBs BLACK BOX. 724-746-5500 blackbox.com

Voice over IP (VoIP) for Telephony. Advantages of VoIP Migration for SMBs BLACK BOX. 724-746-5500 blackbox.com Voice over IP (VoIP) for Telephony Advantages of VoIP Migration for SMBs BLACK BOX Hybrid PBX VoIP Gateways SIP Phones Headsets 724-746-5500 blackbox.com Table of Contents Introduction...3 About Voice

More information

ACD: Average Call Duration is the average duration of the calls routed bya a VoIP provider. It is a quality parameter given by the VoIP providers.

ACD: Average Call Duration is the average duration of the calls routed bya a VoIP provider. It is a quality parameter given by the VoIP providers. ACD: Average Call Duration is the average duration of the calls routed bya a VoIP provider. It is a quality parameter given by the VoIP providers. API: An application programming interface (API) is a source

More information

Telephony Fundamentals

Telephony Fundamentals + Telephony Fundamentals Basic Telephony general terms Central Office (CO) - the telephone facility where telephone users lines are joined together to switching equipment that connects telephone users

More information

Internet Services & Protocols Multimedia Applications, Voice over IP

Internet Services & Protocols Multimedia Applications, Voice over IP Department of Computer Science Institute for System Architecture, Chair for Computer Networks Internet Services & Protocols Multimedia Applications, Voice over IP Dipl.-Inform. Stephan Groß Room: GRU314

More information

Gateways and Their Roles

Gateways and Their Roles Gateways and Their Roles Understanding Gateways This topic describes the role of voice gateways and their application when connecting VoIP to traditional PSTN and telephony equipment. Analog vs. Digital

More information

Chapter 10 Session Initiation Protocol. Prof. Yuh-Shyan Chen Department of Computer Science and Information Engineering National Taipei University

Chapter 10 Session Initiation Protocol. Prof. Yuh-Shyan Chen Department of Computer Science and Information Engineering National Taipei University Chapter 10 Session Initiation Protocol Prof. Yuh-Shyan Chen Department of Computer Science and Information Engineering National Taipei University Outline 12.1 An Overview of SIP 12.2 SIP-based GPRS Push

More information

- Basic Voice over IP -

- Basic Voice over IP - 1 Voice over IP (VoIP) - Basic Voice over IP - Voice over IP (VoIP) is a digital form of transport for voice transmissions, replacing analog phone systems. The benefits of VoIP are considerable: Better

More information

Software-Powered VoIP

Software-Powered VoIP Software-Powered VoIP Ali Rohani Anthony Murphy Scott Stubberfield Unified Communications Architecture Core Scenarios UC endpoints QOE Monitoring Archiving CDR AOL Public IM Clouds Yahoo Remote Users MSN

More information

VOICE OVER IP AND NETWORK CONVERGENCE

VOICE OVER IP AND NETWORK CONVERGENCE POZNAN UNIVE RSITY OF TE CHNOLOGY ACADE MIC JOURNALS No 80 Electrical Engineering 2014 Assaid O. SHAROUN* VOICE OVER IP AND NETWORK CONVERGENCE As the IP network was primarily designed to carry data, it

More information

Application Notes. Introduction. Contents. Managing IP Centrex & Hosted PBX Services. Series. VoIP Performance Management. Overview.

Application Notes. Introduction. Contents. Managing IP Centrex & Hosted PBX Services. Series. VoIP Performance Management. Overview. Title Series Managing IP Centrex & Hosted PBX Services Date July 2004 VoIP Performance Management Contents Introduction... 1 Quality Management & IP Centrex Service... 2 The New VoIP Performance Management

More information

QoS issues in Voice over IP

QoS issues in Voice over IP COMP9333 Advance Computer Networks Mini Conference QoS issues in Voice over IP Student ID: 3058224 Student ID: 3043237 Student ID: 3036281 Student ID: 3025715 QoS issues in Voice over IP Abstract: This

More information

Internet Services & Protocols Multimedia Applications, Voice over IP

Internet Services & Protocols Multimedia Applications, Voice over IP Department of Computer Science Institute for System Architecture, Chair for Computer Networks Internet Services & Protocols Multimedia Applications, Voice over IP Dr.-Ing. Stephan Groß Room: INF 3099 E-Mail:

More information