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© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.1 Computer Networks and Internets, 5e By Douglas E. Comer Lecture PowerPoints By Lami Kaya, LKaya@ieee.org
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.2 Chapter 29 Multimedia and IP Telephony (VoIP)
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.3 Topics Covered 29.1 Introduction 29.2 Real-Time Data Transmission and Best-Effort Delivery 29.3 Delayed Playback and Jitter Buffers 29.4 Real-Time Transport Protocol (RTP) 29.5 RTP Encapsulation 29.6 IP Telephony 29.7 Signaling and VoIP Signaling Standards 29.8 Components of an IP Telephone System 29.9 Summary of Protocols and Layering 29.10 H.323 Characteristics 29.11 H.323 Layering 29.12 SIP Characteristics and Methods 29.13 An Example SIP Session 29.14 Telephone Number Mapping and Routing
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.4 29.1 Introduction This chapter –continues the discussion by examining the transfer of multimedia over the Internet –examines how multimedia can be sent over a best-effort communication mechanism –describes a general-purpose protocol for real-time traffic –considers the transmission of voice telephone calls in detail
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.5 29.2 Real-Time Data Transmission and Best-Effort Delivery Multimedia to refer to data that contains audio or video Real-time multimedia refers to multimedia data that must be reproduced at exactly the same rate that it was captured How can Internet be used for transfer of real-time multimedia? –recall that the Internet offers best-effort delivery service packets can be lost, delayed, or delivered out of order If multimedia data is sent across the Internet without special treatment, the resulting output may be unacceptable –Early systems solved the problem by creating communication networks specifically designed to handle audio or video The analog telephone network uses an isochronous network to provide high- quality reproduction of audio Analog cable television systems are designed to deliver multiple channels of broadcast video with no interruptions or loss
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.6 29.2 Real-Time Data Transmission and Best Effort Delivery The Internet uses additional protocol support –Instead of requiring the underlying networks to handle real-time transmission The most significant problem to be handled is jitter For example, consider a live webcast If a protocol uses timeout-and-retransmission to resend the packet, the retransmitted packet will arrive too late to be useful –the receiver will have played the video and audio from successive packets –it makes no sense to insert a snippet of the webcast that was missed earlier
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.7 29.3 Delayed Playback and Jitter Buffers To overcome jitter and achieve smooth playback of real-time data, some techniques are employed, such as Timestamps –A sender provides a timestamp for each piece of data –A receiver uses the timestamps to handle out-of-order packets and to display the data in the correct time sequence Jitter Buffer –To accommodate jitter (i.e., small variances in delay), a receiver buffers data and delays playback
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.8 29.3 Delayed Playback and Jitter Buffers In jitter buffer mechanism –a receiver maintains a list of data items –and uses timestamps to order the list Before playback –a receiver delays for d time units means the data being played is d time units behind the data that is arriving If a given packet is delayed less than d, –the contents of the packet will be placed in the buffer before it is needed for playback –items are inserted into a jitter buffer with some variation in rate –the playback process extracts data from a jitter buffer at a fixed rate Figure 29.1 illustrates the organization of a real-time playback system
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.9 29.3 Delayed Playback and Jitter Buffers
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.10 29.4 Real-Time Transport Protocol (RTP) IP suite contains the RTP –used to transmit real-time data across the Internet Term Transport is used because RTP sits above the transport layer RTP does not ensure timely delivery of data Instead, it provides three items in each packet that permit a receiver to implement a jitter buffer: –A sequence number that allows a receiver to place incoming packets in the correct order and to detect missing packets –A timestamp that allows a receiver to play the data in the packet at the correct time in the multimedia stream –A series of source identifiers that allow a receiver to know the source(s) of the data
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.11 29.4 Real-Time Transport Protocol (RTP) Figure 29.2 (below) illustrates an RTP packet header: –See how the sequence number, timestamp, and source identifier fields appear in an RTP packet header
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.12 29.5 RTP Encapsulation RTP uses UDP for message transport –Thus, each RTP message is encapsulated in a UDP datagram for transmission over the Internet –The resulting messages can be sent via broadcast or multicast Figure 29.3 (below) illustrates the three levels of encapsulation that are used when an RTP message is transferred over a single network
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.13 29.6 IP Telephony Companies around the world are replacing traditional telephone switches with IP routers The motivation is economic: –routers cost much less than traditional telephone switches Sending both data and voice in IP datagrams lowers cost –because the underlying network infrastructure is shared The basic idea behind IP telephony is straightforward: –continuously sample audio –convert each sample to digital form –send the resulting digitized stream across an IP network in packets –and convert the stream back to analog for playback However, many details complicate the task
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.14 29.6 IP Telephony (VoIP) However, many details complicate the task –A sender cannot wait to fill a large packet because doing so delays transmission by several seconds –The system must handle call setup when a caller dials, the system must translate the phone number to an IP address, and locate the specified party –When a call begins the called party must accept and answer the call –When a call ends the two parties must agree on how to terminate communication The most significant complications: –IP telephony strives to be backward compatible with Public Switched Telephone Network (PSTN) or some call Plain Old Telephone System (POTS) –Also, integration with Private Branch Exchange (PBX)
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.15 29.7 Signaling and VoIP Signaling Standards Two groups have created standards for VoIP: –International Telecommunications Union (ITU) –Internet Engineering Task Force (IETF) Both groups agree on the basics for encoding and transfer: –Audio is encoded using Pulse Code Modulation (PCM) –RTP is used to transfer the digitized audio The main complexity of VoIP –lies in call setup and call management The process of establishing and terminating a call is known as signaling and includes –mapping a phone number to a location –finding a route to the called party –handling other details such as call forwarding
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.16 29.7 Signaling and VoIP Signaling Standards Mechanism used in the traditional telephone system to handle call management is known as Signaling System 7 (SS7) To be compatible with existing telephones –new protocols must be able to interact with SS7 –this is to place outgoing calls and to accept incoming calls Set of signaling protocols were proposed for use with VoIP –IETF proposed Session Initiation Protocol (SIP) Media Gateway Control Protocol (MGCP) –ITU proposed a large, comprehensive set of protocols under the umbrella of H.323 –The two groups jointly proposed Megaco (H.248)
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.17 29.8 Components of an IP Telephone System Figure 29.4 (below) lists main components of an IP telephone system
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.18 29.8 Components of an IP Telephone System Figure 29.5 (below) illustrates how they are used to interconnect networks
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.19 29.8 Components of an IP Telephone System An IP telephone –connects to a network, uses IP for all communication –offers a traditional telephone interface that allows a user to place or receive telephone calls A Media Gateway Controller (Gatekeeper or Soft Switch) –provides overall control and coordination between a pair of IP telephones allowing a caller to locate a callee or access services such as call forwarding A Media Gateway –provides translation of audio as a call passes across the boundary between an IP network and the PSTN A Signaling Gateway –also spans the boundary between a pair of disparate networks –translation of signaling operations (either side to initiate a call)
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.20 29.8 Components of an IP Telephone System The concepts and terminology defined above present a straightforward and somewhat simplified view of VoIP –that was derives from work in the IETF and ITU on the Megaco and Media Gateway Control Protocol (MGCP) Practical implementations of VoIP service are more complex The next sections give examples: –29.8.1 SIP Terminology and Concepts –29.8.2 H.323 Terminology and Concepts –29.8.3 ISC Terminology and Concepts
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.21 29.8 Components of an IP Telephone System 29.8.1 SIP Terminology and Concepts The Session Initiation Protocol (SIP) defines set of elements for the signaling system User Agent (device that makes or terminates phone calls) Location Server –manages a database of information about each user (such as a set of IP addresses, subscribed services, and the user's preferences) Support Servers (proxy, redirect, registrar) –Proxy Server can forward requests from user agents to another location –Redirect Server handle tasks such as call forwarding and 800-number connections –Registrar Server to receive registration requests and update the database that location servers consult
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.22 29.8 Components of an IP Telephone System 29.8.2 H.323 Terminology and Concepts The H.323 defines alternative terminology and additional concepts, focuses on PSTN interaction It is extremely broad and covers many details H.323 can be summarized as follows: –Terminal provides the IP telephone function, which may also include facilities for video and data transmission –Gatekeeper H.323 gatekeeper provides location and signaling functions coordinates the operation of the gateway to provide connection to the PSTN – Gateway H.323 uses a single gateway to interconnect the IP telephone system with the PSTN the gateway handles both signaling and media translation – Multipoint Control Unit (MCU) An MCU provides services such as multipoint conferencing
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.23 29.8 Components of an IP Telephone System 29.8.3 ISC Terminology and Concepts Vendors formed International SoftSwitch Consortium (ISC) –to create a uniform, comprehensive functional model that incorporates all models of IP telephony into a single framework ISC defined a list of functions that suffices for all situations: –Media Gateway Controller Function (MGC-F) maintains state information in endpoints; it provides call logic and call control –Call Agent Function (CA-F) The CA-F is a subset of the MGC-F that maintains call state –InterWorking Function (IW-F) is a subset of the MGC-F that handles signaling between heterogeneous networks such as SS7 and SIP –Routing Function and Accounting Function (R-F/A-F) R-F handles routing of calls for the MGC-F A-F collects information used for accounting and billing
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.24 29.8 Components of an IP Telephone System 29.8.3 ISC Terminology and Concepts –Signaling Gateway Function (SG-F) handles signaling between an IP network and the PSTN – Access Gateway Signaling Function (AGS-F) handles signaling between an IP network and a circuit-switched access network such as the PSTN – Application Server Function (AS-F) The AS-F handles a set of application services such as voicemail – Service Control Function (SC-F) It is called when an AS-F must control (i.e., change) the logic of a service – Media Gateway Function (MG-F) handles translation of digitized audio between two forms may also include detection of events such as whether a phone is off-hook may include recognition of Dual Tone Multi-Frequency (DTMF) signals the audio signaling standard that is known as Touch Tone encoding –Media Server Function (MS-F) manipulates a media packet stream on behalf of an AS-F application
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.25 29.9 Summary of Protocols and Layering Multiple groups have proposed protocols for VoIP –competing protocols exist at most layers of the protocol stack Figure 29.6 (below) lists some of the proposed protocols –along with their position in the Internet 5-layer reference model
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.26 29.10 H.323 Characteristics H.323 standard consists of a set of protocols –that work together to handle all aspects of telephone communication The highlights of H.323 are: –Handles all aspects of a digital telephone call –Includes signaling to set up and manage the call –Allows the transmission of video and data while a call is in progress –Sends binary messages that are defined by ASN.1 –Messages are encoded using Basic Encoding Rules (BER) –Incorporates protocols for security –Uses a special hardware unit known as a Multipoint Control Unit to support conference calls –Defines servers to handle tasks such as address resolution, authentication, authorization, accounting, call forwarding, etc.
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.27 29.11 H.323 Layering H.323 protocols use both TCP and UDP for transport –audio can travel over UDP –while a data transfer proceeds over TCP Figure 29.7 (below) illustrates the basic layering in the H.323 standard
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.28 29.12 SIP Characteristics and Methods Highlights of IETF's Session Initiation Protocol (SIP) are: Operates at the application layer Encompasses all aspects of signaling –including location of a called party, notification and setup, determination of availability and termination Provides services such as call forwarding Relies on multicast for conference calls Allows the two sides to negotiate capabilities –and choose the media and parameters to be used A SIP-URI contains a user's name and a domain name –at which the user can be found
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.29 29.12 SIP Characteristics and Methods SIP defines some message types and extensions –Message types are known as SIP method Figure 29.8 (below) lists the basic SIP methods
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.30 29.13 An Example SIP Session An example of the messages sent during a SIP session will clarify some of the details –and help explain the general idea behind most IP telephony Figure 29.9 lists a sequence of messages sent –A user agent, A, contacts a DNS server then communicates with a proxy server, which invokes a location server –Once the call is established, the two VoIP communicate directly –Finally, SIP is used to terminate the call Typically, a user agent is configured with the IP address of one or more DNS and one or more proxy servers Each proxy server is configured with the address of one or more location servers –If a given server is unavailable SIP can find an alternate quickly
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.31 Figure 29.9 An example of the messages exchanged by SIP to manage a telephone call
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.32 29.14 Telephone Number Mapping and Routing How should IP telephone users be named and located? –The PSTN follows ITU standard E.164 for telephone numbers –The SIP uses IP addresses The problem of locating users is complicated –because multiple types of networks may be involved Designers define two sub-problems: –locate a user in the integrated network –and find an efficient route to the user The IETF has proposed protocols that correspond to the mappings needed for the two sub-problems: – ENUM converts a telephone number to a URI – TRIP finds a user in an integrated network
© 2009 Pearson Education Inc., Upper Saddle River, NJ. All rights reserved.33 29.14 Telephone Number Mapping and Routing ENUM (short for E.164 NUMbers ) –solves the problem of converting an E.164 tel number into a URI –ENUM uses the DNS to store the mapping An ENUM mapping can be 1-to-1 or 1-to-many Telephone Routing over IP (TRIP) –Solves the problem of finding a user in an integrated network –A location server or other network element can use TRIP to advertise routes Thus, two location servers use TRIP to inform each other about external routes that they each know –TRIP divides the world into a set of IP Telephone Administrative Domains (ITADs)
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