Network+ Guide to Networks 6 th Edition Chapter 12 Voice and Video Over IP.

Network+ Guide to Networks6th Edition

Chapter 12Voice and Video Over IP

Objectives

• Use terminology specific to converged networks• Explain VoIP (Voice over IP) services, PBXs, and

their user interfaces• Explain video-over-IP services and their user

interfaces• Describe VoIP and video-over-IP signaling and

transport protocols, including SIP, H.323, and RTP• Understand QoS (quality of service) assurance

methods critical to converged networks, including RSVP and DiffServ

Network+ Guide to Networks, 6th Edition 2

Terminology

• IP telephony (VoIP)– Any network carrying voice signals using TCP/IP

• Public or private– Runs over any packet-switched network

• Virtually any data connection type can carry VoIP signals, including:– T-carriers, ISDN, DSL, broadband cable, satellite

connections, WiFi, WiMAX, HSPA+, LTE, cellular telephone networks


Terminology (cont’d.)

• Internet telephony– VoIP calls carried over Internet



• Nondata applications on converged networks– IPTV (IP television)– Videoconferencing

• Multiple participants communicate and collaborate via audiovisual means

– Streaming video• Compressed video delivered in continuous stream

– Webcasts• Streaming videos supplied via the Web



• Multicasting– One node transmits same content to every client in

group• Video over IP Services:

– IPTV, videoconferencing, streaming video, & IP multicasting

• Unified communications (unified messaging) service– Several communication forms available from single

user interface• Users can access the Web, send & receive faxes, email

messages, instant messages, or telephone calls, and participate in videoconference calls—all from one console


VoIP Applications and Interfaces

• Reasons for implementing VoIP– Lower voice call cost (toll bypass)– New, enhanced features and applications– Centralize voice and data network management

• Voice & data can be combined on a network in several configurations:– Use traditional telephone (sends, receives analog signals)– Use telephone specially designed for TCP/IP transmission– Use a computer with microphone, speaker, VoIP client

software– Mixture of these types


Analog Telephones

• Traditional telephone used for VoIP– Signals converted to digital form

• Codec (coder/decoder)– Method of compressing, encoding, analog signals into

bits


9

Analog Telephones, cont.

• Analog VoIP clients may connect to IP networks in one of four ways:

• First analog-to-digital conversion method:• ATA (analog telephone adapter)

– Card within computer workstation– Or externally attached device– Telephone line connects to RJ-11 adapter port– Converts analog voice signals to IP packets

Network+ Guide to Networks, 6th Edition


Figure 12-1 ATA (analog telephone adapter)

Courtesy of Grandstream Networks, Inc.

Analog Telephones (cont’d.)

• Second analog-to-digital conversion method:– Connecting an analog telephone line to VoIP-enabled

device (switch, router, gateway)– Convert analog voice signals into packets and then

issuing the packets to a data network—and vice versa



Figure 12-2 VoIP router

Photo of SmartNode™4520 Analog VoIP router from Patton Electronics , Co.


• Third analog-to-digital conversion method:– Digital PBX (a.k.a IP-PBX)

• PBX (private branch exchange): telephone switch connecting calls within private organization

– IP-PBX• Is a private switch• Accepts, interprets both analog & digital voice signals• Connects with traditional PSTN lines, data networks• Transmits, receives IP-based voice signals to and from

other network connectivity devices• Most are packaged with sophisticated software used to

configure and maintain an organization’s phone systemNetwork+ Guide to Networks, 6th Edition 13


Figure 12-3 IP-PBX

Courtesy of Epygi Technologies, Ltd.


• Hosted PBX– Exists on the Internet– Separate provider for call management services– May also be called virtual PBXs

• Trademark of VirtualPBX company

• Advantage of Hosted PBXs– No installation or maintenance of hardware and

software for call completion and management• VoIP Trunk

– ONSIP example of a VoIP trunk provider



• Fourth analog-to-digital conversion method:– Traditional telephone connects to analog PBX

• Then connects to voice-data gateway– Gateway connects traditional telephone circuits with

TCP/IP network• Internet or private WAN

– The Gateway• Digitizes incoming analog voice signal• Compresses data• Assembles data into packets• Issues packets to packet-switched network



Figure 12-4 Integrating VoIP networks and analog telephones

Courtesy of Course Technology/Cengage Learning

IP Telephones

• IP telephones (IP phones)– Transmit, receive only digital signals– Voice immediately digitized, issued to network in

packet form– Requires unique IP address– Looks like traditional touch-tone phone– Connects to RJ-45 wall jack– Connection may pass through connectivity device

before reaching IP-PBX



Figure 12-5 Accessing a VoIP network from IP phones


IP Telephones (cont’d.)

• IP telephone features– Speed-dialing– Call hold– Transfer, forward– Conference calling– Voice-mail access– Speakers, microphones, LCD screen– Mobile and wired styles– Some can act as Web browsers– Easily moved from office to office


IP Telephones (cont’d.)

• Conventional analog telephone– Obtains current from local loop– Current used for signaling (ring, dial tone)

• IP telephones– Need electric current– Not directly connected to local loop– Most use separate power supply

• Susceptible to power outages• Requires assured backup power sources

– Some use PoE (power over Ethernet)



Figure 12-6 An IP phone

Courtesy of Grandstream Networks, Inc.

Softphones

• Computer programmed to act like IP telephone– Provide same calling functions– Connect to network; deliver services differently

• Prerequisites– Computer minimum hardware requirements– IP telephony client installed– Digital telephone switch communication– Full-duplex sound card– Microphone, speakers

• Softphone example: Skype is a popular Internet proprietary telephony software and is a type of softphone


Softphones (cont’d.)

• Graphical interface– Presented after user starts softphone client software– Customizable

• Versatile connectivity– VoIP solution for traveling employees and

telecommuters• Convenient, localized call management

– Call tracking• Date, time, duration, originating number, caller names

– Simplifies recordkeeping and billing



Figure 12-7 Softphone interface

Courtesy of CounterPath Corporation


Figure 12-8 Connecting softphones to a converged network


Video-over-IP Applications and Interfaces

• Cisco Systems estimate– By 2015, over two-thirds of Internet traffic will be

video traffic• Factors fueling growth

– Large quantity of video content available (Netflix & Hulu)

– Increasing number of devices accessing Internet– Decreasing cost of bandwidth, equipment

• Video-over-IP services categories– Streaming video, IPTV, videoconferencing


Streaming Video

• Is a video-over-IP application– Basic computer appropriate audiovisual hardware and

software to view the encoded video• One method is Video-on-demand:

– Files stored on video streaming server• Popular• Viewer chooses video when convenient: Web browser

• Another method is Streaming video:– Video issued live

• Directly: source to user


Streaming Video (cont’d.)

• Drawbacks of live stream– Content may not be edited before distribution– Viewers must connect with stream when issued

• Video-on-demand benefits– Content viewed at user’s convenience– Viewers control viewing experience

• Pause, rewind, fast-forward capabilities



Figure 12-9 Video-on-demand and live streaming video


Streaming Video (cont’d.)

• Most streaming video– Takes place over public networks


IPTV (IP Television)

• Telecommunications carrier and cable company networks– Offer high-bandwidth Internet connections to deliver

digital television signals using IPTV• Elements of delivering digital video to consumers

– Telco accepts video content at a head end– Telco’s CO (central office) servers provide

management services– Video channel assigned to multicast group– When an IPTV user changes the channel they are

opting out of one IP multicast group and opting into another



Figure 12-10 A telecommunications carrier’s IPTV network


Videoconferencing

• Videoconferencing– Full-duplex connections

• Participants send and receive audiovisual signals– Real time– Benefits

• Cost savings, convenience• Replace face-to-face business meetings• Allow collaboration


Videoconferencing (cont’d.)

• Videoconferencing uses– Telemedicine– Tele-education– Judicial proceedings– Surveillance

• Hardware, software requirements– Means to generate, send, receive audiovisual signals

• Computer workstation with cameras, microphones, software

• Video terminal or video phone



Figure 12-12 Videophone

Courtesy of Grandstream Networks

Videoconferencing (cont’d.)

• Video bridge– Manages multiple audiovisual sessions so that

participants can see and hear each other– May exist as a piece of Hardware or software, in the

form of a conference server• Internet-accessible video bridging services can be

leased– Well suited for occasional videoconference use

• Some organizations might maintain their own conference servers


Signaling Protocols

• Signaling protocols:– Set up and manage sessions between clients

• Early VoIP: used proprietary signaling protocols• Today: standardized signaling protocols are usually

used which are discussed next• SS7 (Signaling System 7): typically handles call

signaling on the PSTN– May come up when trying to interconnect the PSTN

with VoIP


H.323• Describes architecture & a group of protocols for:

– Establishing and managing packet-switched network multimedia sessions

• Supports voice and video-over-IP services• Terms:

– H.323 terminal (IP phone)– H.323 gateway (connects to other signaling protocols)– H.323 gatekeeper (authenticates terminals, manages

bandwidth, and oversees call routing, accounting, and billing– MCU (multipoint control unit) provides support for multiple

H.323 terminals– H.323 zone (collection of H.323 terminals, gateways, and

MCUs are managed by a single H.323 gatekeeper)



Figure 12-13 An H.323 zone


H.323 (cont’d.)

• H.225 and H.245 signaling protocols– Specified in H.323 standard– Operate at Session layer of OSI

• H.225 handles call or videoconference signaling– IP telephone requests a call setup via H.225

• H.245 ensures correct information type formatting—whether voice or video issued so that the H.323 terminal can interpret the data

• H.323 remains a popular signaling protocol on large voice and video networks

• In sum, H.323 is the call control signaling protocol– Used to setup, maintain, teardown, and redirect calls


SIP (Session Initiation Protocol)

• Performs similar functions as H.323• Application layer signaling and control protocol that

was modeled after HTTP– VoIP call is formatted like an HTTP request and rely

on URL style addresses• Limited capabilities compared to H.323

– SIP does not have features such as caller ID and instead it depends on other protocols and services to supply them


SIP (cont’d.)

• SIP network components (pg. 574):– User agent– User agent client– User agent server– Registrar server– Proxy server– Redirect server



Figure 12-14 An H.323 zone


SIP (cont’d.)

• Some VoIP vendors prefer SIP over H.323– Simplicity—easier to setup– Fewer instructions to control call– Consumes fewer processing resources than H.323

• SIP and H.323– Regulate call signaling and control for VoIP or video-

over-IP clients and servers– Do not account for communication between media

gateways• MGCP or MEGACO protocols are used for

communication between media gateways


MGCP (Media Gateway Control Protocol) and MEGACO (H.248)

• Media Gateway: A gateway capable of accepting connections from multiple devices, such as IP telephones, traditional telephones, IP fax machines, traditional fax machines, and translating analog signals into packets and vice versa– Accepts PSTN lines, converts signals into VoIP format, and

translates between signaling protocols – Gateways still need to exchange and translate signaling and

control information with each other so that voice and video packets are properly routed through the network

• They rely on the MGC (media gateway controller) device, which can manage multiple media gateways

Network+ Guide to Networks, 6th Edition46

MGCP and MEGACO (cont’d.)

• MGC (media gateway controller)– Used by media gateways to make sure that voice and

video packets are properly routed through the network– Enables multiple media gateways to communicate– Computer managing multiple media gateways

• Also called a softswitch (software performs the call switching)


MGC (media gateway controller)

• Example:– When a media gateway receives a call, rather than

attempting to determine how to handle the call, the gateway contacts the MGC with a message that says, “I received a signal. You figure out what to do with it next.”

– The MGC then determines which of the network’s media gateways should translate the information carried by the signal. It also figures out which physical media the call should be routed over

– Media gateways simply follow orders from the MGC



• MGC communicate with media gateways according to several protocols (MGCP or MEGACO)– MGCP (Media Gateway Control Protocol)

• Commonly used on networks that support a number of media gateways

• Can operate with both H.323 or SIP call signaling and control protocols

• Older protocol• Newer gateway control protocol is MEGACO



• MGC gateway communicate protocols (cont’d.)– MEGACO (H.248)

• Is a newer gateway control protocol• Performs same functions as MGCP• Different commands and processes• Operates with H.323 or SIP• Superior to MGCP



Figure 12-15 Use of an MGC (media gateway controller)Courtesy of Course Technology/Cengage Learning

Transport Protocols

• Session Protocols: only communicate information about a voice or video session

• Transport layer Protocols: are a set of protocols used to actually deliver the voice or video payload—the bits of encoded voice that together make up words spoken into an IP telephone


Transport Protocols, cont.

• UDP and TCP operate at the Transport layer– TCP: connection oriented and provides some

measure of delivery guarantees– UDP: connectionless

• The preferred protocol for telephone conversations and videoconferences—requires less overhead and therefore can transport packets more quickly

• Packet loss tolerable if additional protocols overcome UDP shortcomings

– Real-time Transport Protocol (RTP)– Real-time Transport Control Protocol (RTCP)


RTP (Real-time Transport Protocol)

• A Transport layer protocol used with voice and video transmission

• RTP operates on top of UDP and provides information about packet sequence to help receiving nodes detect delay and packet loss

• RTP also assigns packets a timestamp that corresponds to when the data in the packet was sampled from the voice or video stream– Timestamp helps the receiving node synchronize incoming

data• Has no mechanism to detect whether or not it is successful

– For that, it relies on RTCP


RTCP (Real-time Transport Control Protocol)

• A companion protocol to RTP that provides feedback on the quality of a call or videoconference to its participants– Packets transmitted periodically to all session endpoints– Recipients of RTP data use RTCP to issue information

about the number and percentage of packets lost and delay suffered between the sender and receiver

• Not mandatory on RTP networks and some network admins may prefer not to use it

• RTP and RTCP– Provide information about packet order, loss, delay– Cannot correct transmission flaws that is handled by QoS

protocols


QoS (Quality of Service) Assurance

• QoS: is a measure of how well a network service matches its expected performance

• Despite all the advantages of using VoIP and video-over-IP, it is more difficult to transmit these types of signals over a packet-switched network than it is to transmit data signals


Techniques to overcome QoS Challenges

• QoS techniques:– RSVP (Resource Reservation Protocol)– DiffServ (Differentitated Service)– MPLS (Multiprotocol Label Switching)

• Help to eliminate call and video delays


RSVP (Resource Reservation Protocol)

• Transport layer protocol• Before transmission

– Attempts to reserves network resources (bandwidth) for a transmission so the signal will hopefully arrive without suffering delays

• Creates path between sender & receiver– Provides sufficient bandwidth so the signal arrives without

delay• Sending node issues PATH statement via RSVP to

receiving node– Indicates required bandwidth and expected service level it

expectsNetwork+ Guide to Networks, 6th Edition 58

RSVP (cont’d.)

• Two service types:– Guaranteed service: assures the transmission will not suffer packet

losses and will experience minimal delay– Controlled-load service: provides the type of QoS a transmission

would experience if the network carried little traffic• Each router that the PATH message traverses marks the

transmission’s path by noting which router the PATH message came from

• After the destination node receives the PATH message, it responds with a Reservation Request (RESV) message– Follows same path in reverse– Routers between destination and sending nodes allocate the

requested bandwidth– Sending node transmits data


RSVP (cont’d.)

• Specifies and manages unidirectional transmission– For two users to participate in a VoIP or

videoconference, the resource reservation process takes place in both directions

• RSVP Emulates circuit-switched path– Which provides excellent QoS

• Drawback: high overhead– Acceptable on small networks– Larger networks use DiffServ


DiffServ (Differentiated Service)

• Differentiated Service (DiffServ)—another way to prioritize network traffic

• Places information in the DiffServ field in an IPv4 datagram. In IPv6 datagrams, it uses a similar field known as the Traffic Class field– This information indicates to the network routers how

the data stream should be forwarded• Because of its simplicity and relatively low

overhead, DiffServ is better suited to large, heavily trafficked networks than RSVP


DiffServ (cont’d.)

• Two forwarding types:– EF (Expedited Forwarding)

• Data stream assigned minimum departure rate• Circumvents delays

– AF (Assured Forwarding)• Data streams assigned different router resource levels• Prioritizes data handling, but does not guarantee that

on a busy network packets will arrive on time and in sequence

– A deeper discussion of these prioritization techniques is beyond the scope the textbook and class


MPLS (Multiprotocol Label Switching)

• Modifies data streams at Network layer• To indicate where data should be forwarded, MPLS replaces

the IP datagram header with a label at the first router a data stream encounters

• The MPLS label contains information about where the router should forward the packet next– Router’s in the data stream’s path revises label

• To indicate next hop to the receiving router• Routers only need to read the MPLS (shim)• Routers do not have a perform a route lookup to determine where

to forward the packet next

• With MPLS, data streams are more likely to arrive without delay• Networks can and often do combine multiple QoS techniques


Summary

• VoIP services carry voice signals over TCP/IP protocol

• Four ways to connect VoIP clients to IP networks• Streaming video can be delivered live or on-demand• IPTV television signals travel over packet-switched

connections• Video bridges manage communication for

videoconferences• H.323 standard

– Describes architecture and protocols for managing multimedia sessions on a packet-switched network


Network+ Guide to Networks 6 th Edition Chapter 12 Voice and Video Over IP.

Documents

network network

voip networks

data networks

ip slide

analog pbx

analog voice signals

voip voice

ip packets network