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Introduction to VoIP Part II

Introduction to VoIP Part II. Dr. Farid Farahmand CET479 Updated 5/18/2007. Overview. Basic concepts of VoIP and its motivating facts How to digitally decode voice prior to its transport How to transport voice between users After the session is established how to transport voice

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Introduction to VoIP Part II

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  1. Introduction to VoIPPart II Dr. Farid Farahmand CET479 Updated 5/18/2007

  2. Overview • Basic concepts of VoIP and its motivating facts • How to digitally decode voice prior to its transport • How to transport voice between users • After the session is established how to transport voice • How to setup and teardown voice sessions • How to create sessions • How signaling protocols work

  3. Speech Coding • Voice has to be digitally encoded/decoded • Streams of 1’s and 0’s • How voice is coded impacts the channel efficiency (BW utilization) • Various speech coding techniques are used • Bandwidth and voice quality are related • Yet the relation is not linear • For example: 16 Kbps voice transmission is not necessarily better than 32 Kbps • Objective of speech coding is to minimize BW and maintaining high quality of speech • High quality is measured by MOS metric (Mean-Option Score) • Other metric alternatives are available (PSQM)

  4. A Little about Speech More bits requires more BW but typically more quantization level • Speech is considered to be an analog signals • The objective is to reconstruct the speech digitally A signal can be reconstructed if the sampling rate is twice the max. input frequency

  5. A Little about Speech • Uniform quantization level can cause discrimination • Loud voices will have lower quantization error • A more effective approach is to us non-uniform quantization • Smaller levels  smaller quantization level • Larger levels  Less granularity More accuracy Less accuracy

  6. Speech-Coding Techniques • Choice of speech coding is critical to having high-quality voice • Two conflicting objectives • Reducing bandwidth • Maintaining the natural-sounding speech (toll quality)

  7. G. 711 Speech Coding • ITU Recommendation G . 711 Speech decoding • Typical human speech has a maximum frequency of about 4 KHz: Fmax = 4KHz • Based on Nyquist Theorem, analog signals must be sampled at twice their maximum frequency: Sampling rate =8000 sample/second = 2 x Fmax • Each sample is represented with 8 bits • BW requirement will be 64 Kbps for standard circuit switch based telephone • Toll-quality (MOS) is 4.3 = Excellenet • More efficient coding techniques • G.726  32 Bit rate (Kbps) toll-quality = 4.0 • G.728  16 Bit rate (Kbps) toll-quality = 3.9 • G.729  08 Bit rate (Kbps) toll-quality = 4.0 • VoIP uses more efficient coding techniques • The two ends negotiate on which coding technique to use

  8. Next: • Basic concepts of VoIP and its motivating facts • How to digitally decode voice prior to its transport • How to transport voice between users • After the session is established how to transport voice • How to setup and teardown voice sessions • How to create sessions • How signaling protocols work

  9. Transporting Voice Signals • Digitally codes voice can be encapsulated into IP packets • IP is just a routing protocol • IP routing is based on the destination address – packets with the same source/destination address can take different paths • Provides no quarantine of service • One way to transport the IP packet packets is using TCP • The transmission control protocol (TCP) • Ensuring that all packet are delivered in sequence • Providing transmission reliability • TCP provides port number in its header to distinguish between different applications (SMTP: Port 25 / Web: port 80 / Telnet: Port 23)

  10. TCP/IP Model (Click for more information)

  11. TCP/IP Headers

  12. Introduction to UDP • The User Defined Protocol performs a very simple function • Passing IP packets to the end user • Provides no guarantee of service and inherently unreliable • Has no concept of packet ordering • Yet, provides a quick one-shot transmission • Most common example is using UDP in DNS

  13. UDP

  14. Voice over UDP • UDP was not designed for transporting voice • Due to its quick transporting ability, it is suitable for voice • Basic shortcoming of UDP • No packet loss recovery mechanism • Voice communications can tolerate some loss • Efficient coding techniques can be design to recover some lost packets • Supporting QoS can reduce the probability of packet loss • No packet ordering scheme • Packets in the same session are unlikely to follow different paths  lower probability of out of ordering …we still like to resolve some of the shortcomings of UDP

  15. A Transport Protocol for Real-Time Application Protocol (RTP) • RTP is designed to support transporting real-time applications (voice, video, etc.) • RTP contains two protocols • RTP • RTP Control Protocol • Main functionalities • Detect packet out-of-sequencing • Report packet loss • Only provides information and takes no action!

  16. RTP Protocols • RTP resides on top of UDP • Includes packet sequence number • Provides timestamp (used for synchronization and calculating jitter and delay) • RTP Control Protocol (RTCP) • Considered as a companion to RTP / optional • Provides feedback about quality of the voice session • Number of lost RTP packets • Packet delays • Inter-arrival jitter • RTP and RTCP are often established as two separate sessions • Odd/Even port numbers between 1025-65,535

  17. Next: • Basic concepts of VoIP and its motivating facts • How to digitally decode voice prior to its transport • How to transport voice between users • After the session is established how to transport voice • How to setup and teardown voice sessions • How to create sessions • How signaling protocols work

  18. Call Setup and Teardown • The main question: • How to establish a voice session • How to teardown the session • Call setup and teardown is commonly used in traditional telephony • Signaling protocols are invoked before and during the call • Setup • Monitor/maintenance • Teardown • SS7 is the most common signaling example used in our telephone network • In case of VoIP most initial signaling protocols were proprietary • ITU-T (International Telecommunications Union Telecommunications Standardization Sector) recommended H.323 as the signaling protocol • Version 1: 1996 • Version 2: 1998 • Version 4: Today!

  19. H.323 Architecture • Basic components and scope • Terminal • Endpoints / end-user communication devices • Multipoint control unit (MCU) • An H.323 endpoint supporting multipoint conference • Gatekeeper • Optional entity • Controls a number of H.323 terminal, gateways and MCUs • Offers BW control services used to support QoS • Gateway • Establishes connection to other networks (etc. ISDN) • Provides translation services between H.323 and other types of networks • A set of terminals, MCUs, that a single gatekeeper controls is called a ZONE SCN = traditional switched circuit network (SCN)

  20. General Idea

  21. Overview of H.323 Protocols • The actual signaling messages between H.323 entities are specified by • H.225 RAS Signaling • H.223 Call Signaling • H.245 Control Signaling • H.225 has two parts • Call Signaling: • The setup and teardown signaling is very similar to ISDN layer 3 spec. (Q.931) • Can be carried over UDP or TCP / can be performed together – whichever is established first • RAS (registration, admission and Status) signaling • Used between endpoint and a gatekeeper • Always carried over UDP

  22. Overview of H.323 Protocols • H.245 is a control protocol used between two or more endpoints • Manages the media streams between H.323 session participants • Establishes logical channels between endpoints • The channel carries media streams between participants and include media type, bit rate, and so on

  23. References • http://www.analog.com/library/analogDialogue/archives/40-04/blackfin_voip.html • http://www.freesoft.org/CIE/RFC/1889/18.htm - RTCP

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