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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