Voice over IP Presentation on Voice over IP Telecommunication and Computer Networks Presenter: Subash Chandra Pakhrin (072MSI616) MSC in Infromation and Communication Engineering Pulchowk Campus 1

Contents  Introduction  What is VoIP  How VoIP works  Major Components  VoIP Architecture  Signaling Protocols  Challenges  Security Issues 2

Introduction  Voice over Internet Protocol (VoIP) is a technology that enables one to make and receive phone calls through the internet instead of using the traditional PSTN (Public Switched Telephone Network) lines. 3

What is VOIP  VoIP is a packetisation and transport of classic public switched telephone system audio voice over an IP network.  It allows 2-way voice transmission over broadband connection.  It is also called IP telephony, internet telephony, voice over broadband, broadband telephony.  It was developed in February of 1995 by a small company in Israel called VocalTec. 4

PSTN vs. Internet PSTN Internet 1. Voice network use circuit 1. Data network use packet switching switching 2. Dedicated path between 2. No dedicated path between calling and called party Sender and receiver 3. Bandwidth reserved in 3. It acquires and releases advance bandwidth, as needed. 4. Cost is based on distance 4. Cost is based on time and and timebandwidth 5

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VoIP – How does it work ?  Continuously sample audio  Converts each sample to digital form  Send digitized stream across Internet in packet  Converts the stream back to analog for playback Analog to Digital Conversion and Packetizer Voice (Source) Internet Depacketizer and Digital to Analog Conversion Voice (Destination) VoIP Basic Principle 8

VoIP – How does it work ? 1.Compression – voice is compressed typically with the codecs such as PCM, ADPCM, ACELP, etc 2.Encapsulation – the digitized voice is wrapped in an IP packet 3.Routing – the voice packet is routed through the IP network to its final destination. 9

Components  VoIP Codecs  VoIP Gateway  VoIP Protocols 10

VoIP Codecs  Codecs (COder DECoder) are used to convert between analog voice signal and digitally encoded version. Coding Algorithm Voice Bit Rate (kbits/s) Type of Codec G bit PCM64 Waveform Coder G.726 ADPCM32 G.728 LD-CELP16 Model based Vocoding / Vocoders G.729a CS-ACELP8 G MPMLQ6.3 G ACELP5.3  Trade-off between various attributes of speech coders such as bit rate, algorithm’s processing delay, complexity and quality and depending upon the applications, bandwidth available, one can have the choice of a speech coders in a particular VoIP context. 11

What Kind of Transport Protocol Used?  UDP (User Datagram Protocol), But Why not TCP  TCP – Reliable Transport Mechanism  UDP – Unreliable Transport Mechanism.  In real time communication like voice retransmission of packet is not possible.  UDP has no control over the order in which packets arrive at the destination or how long it takes them to get there.  Real-time Transport Protocol (RTP) solves the problem enabling the receiver to put the packets back into the correct order and not wait too long for packets that have either lost their way or are taking too long to arrive. 12

VoIP Network Model LayerName of LayerProtocol 7Application VoIP data, H.323, SIP or MGCP 6Presentation 5SessionRTP 4TransportUDP 3NetworkIP 2LinkFrame (Ethernet, ATM..) 1PhysicalMedium 13

VoIP Data Units Fig. VoIP: Speech payload nested in a packet, with headers added by different protocols (example for VoIP in an Ethernet-based LAN) 14

VoIP – Architecture 15

Signaling Protocols  Main complexity of VoIP : Call setup and call Management  The process of establishing and terminating a call is called Signaling.  In PSTN, signaling protocol is SS7(Signaling System 7)  In VoIP, most implemented signaling protocols are: 1.H.323 by ITU-T 2.SIP (Session Initiation Protocol) by IETF (RFC 2443) 3.MGCP (Media Gateway Controller Protocol) by Cisco  VoIP signaling protocols should be able to interact with SS7. 16

Signaling Protocols H. 323  Most widely used protocol  Provides specifications for real-time, interactive videoconferencing, data sharing and audio applications (VoIP)  Logical components: Terminals, Gateways, Gatekeepers and Multipoint control units (MCU)  Terminals: IP phone  Gatekeeper: provides location and signaling functions; coordinates operation of Gateways.  Gateways: used to interconnect IP telephone system with PSTN, handling both signaling and media translation.  MCU : provides services as multipoint conferencing. 17

Signaling Protocols / SIP  Session Initiation Protocol. Developed by IETF.  Three main elements that comprise a signaling system: 1. User Agent: IP phone or applications. 2. Location servers: stores information about user’s location or IP address. 3. Support servers: a. Proxy Server: forward requests from user agent to another location. b. Registrar Server: receives user’s registration request and updates the database that location server consults. c. Redirect Server: provides an alternate called party’s location for the user agent to contact. 18

Schematic View of Different Server in SIP 19

VoIP SIP 20

SIP Messages Messages are used for communication between the client and the SIP server. These messages are: 21

Comparison of H.323 with SIP H.323SIP Complex ProtocolComparatively Simpler Binary representation for its messagesTextual representation Requires full backward compatibilityDoesn’t require full backward compatibility Not very modularVery modular Not very scalableHighly scalable Complex signalingSimple Signaling Large share of marketBacked by IETF Hundreds of elementsOnly 37 headers Loop detection is difficultLoop detection is comparatively easy 22

VoIP Gateway  Also known as Media Gateway  This provides conversion between the audio signals carried on telephone circuits and data packets carried over the Internet or over other packet networks.  Perform the conversion between Time-division multiplexing (TDM) voice to a media streaming protocol, such as the Real-time Transport Protocol (RTP), as well as signaling protocol used in the VoIP system. 23

Delay Jitter / How to Address It  Delay Smaller than 150ms are not perceived by human.  150 – 400 ms delay acceptable.  Delay Jitter : Due to random queuing delays.  Delay Jitter : Different packet will experience different delays. If all packets are experiencing a same 150 ms delay than that’s not a problem.  Delay Jitter actually degrades the speech quality.  How to address delay Jitter in VoIP? Answer: Use the play-out buffer in the Receiver (Rx). Play-out Buffer (In Receiver):  In this buffer arriving packets are stored in Rx and these packets are played out at an appropriate time. 24

Fixed Play-out Buffer Algorithm 25

Security Risks  As VoIP uses the Internet, for example, it is vulnerable to the same type as security risks  Hacking  Denial of Service  Eavesdropping  Most VoIP services do not support encryption  Further more : Lost or Delayed packets cause drop – out in voice (Addressed by Fixed Play-out buffer algorithm in receiver) 26

VoIP Applications 27

References: Freeman, Roger L. "Voice-Over IP." Telecommunication System Engineering. New York: Wiley, 1996 Voice over IP – Wikipedia : YouTube Video by Prof. Karandikar, IIT Bombay VoIP Overview IP Telephony (VoIP): Chapter 1: Working with VoIP

Thank You ! 29