Week Eleven Agenda Attendance Announcements Mimic Simulator Lab Assignment 4-1-3, Basic Network troubleshooting due December 8, 2010. The lab session is scheduled for December 1, 2010. Review Week Ten Information Current Week Information Upcoming Assignments
Week Eleven Topics Review Week Ten Information 1.Interior Versus Exterior Routing Protocols 2.What is convergence? 3.Autonomous Systems 4.Definitions 5.Loop Free Path Current Week Information
Review Week Ten Static routing refers to manually configuring routes for traffic to use in a network. Dynamic routing is performed automatically when the network topology changes. These changes are made without administrator involvement.
Interior Versus Exterior Routing Protocols Routing protocols designed to work inside an autonomous system are categorized as interior gateway protocols (IGPs). IGPs are RIPv1 and RIPv2, OSPF, Integrated Intermediate System to Intermediate System (IS-IS), and EIGRP. Protocols that work between autonomous systems are classified as exterior gateway protocols (EGPs). EGPs are BGPv4is an acceptable version of BGP on the Internet. Protocols can be further categorized as either distance vector or link-state routing protocols, depending on their method of operation.
Interior Versus Exterior Routing Protocols An interior gateway protocol (IGP) is a routing protocol that is used within an autonomous system (AS). Distance-vector routing protocols each router does not possess information about the full network topology. It advertises its distances to other routers and receives similar advertisements from other routers. Using these routing advertisements each router populates its routing table. In the next advertisement cycle, a router advertises updated information from its routing table. This process continues until the routing tables of each router converge to stable values.
Interior Versus Exterior Routing Protocols Distance-vector routing protocols make routing decisions based on hop-by-hop. A distance vector router’s understanding of the network is based on its neighbors definition of the topology, which could be routing by RUMOR. Route flapping is caused by pathological conditions, hardware errors, software errors, configuration errors, intermittent errors in communications links, unreliable connections within the network which cause certain reach ability information to be repeatedly advertised and withdrawn.
Interior Versus Exterior Routing Protocols In Cisco networks, with distance vector routing protocols flapping routes can trigger routing updates with every state change. Cisco trigger updates are sent when these state changes occur. Traditionally, distance vector protocols do not send triggered updates.
Interior Versus Exterior Routing Protocols Link-state routing protocols, each node possesses information about the complete network topology. Each node then independently calculates the best next hop from it for every possible destination in the network using local information of the topology. The collection of best next hops forms the routing table for the node. This contrasts with distance-vector routing protocols, which work by having each node share its routing table with its neighbors. In a link-state protocol, the only information passed between the nodes is information used to construct the connectivity maps.
Interior Versus Exterior Routing Protocols Each router floods the network with information about itself and it’s state to other routers in the network. Each router has a map of the network Each router looks at itself as the center of the topology Compare this to a “you are here” map at the mall The map is the same, but the perspective depends on where you are at the time
Routing Protocols Interior routing protocols are designed for use in a network that is controlled by a single organization RIPv1 RIPv2, EIGRP, OSPF and IS-IS are all Interior Gateway Protocols
Link State Routing Protocol The link-state algorithm is also known as Dijkstra's algorithm or as the shortest path first (SPF) algorithm The link-state routing algorithm maintains a complex database of topology information The link-state routing algorithm maintains full knowledge of distant routers and how they interconnect. They have a complete picture of the network
Distant Vector Versus Link State Distant Vectors Routing Protocols Link State Routing Protocols RIP (v1 and v2)IGRP EIGRP (hybrid)IS – IS, OSPF
Exterior Gateway Routing Protocol An exterior routing protocol is designed for use between different networks that are under the control of different organizations An exterior routing routes traffic between autonomous systems These are typically used between ISPs or between a company and an ISP BGPv4is the Exterior Gateway Protocol used by all ISPs on the Internet
What is Convergence Definition: Convergence is the process required for all routers in an internetwork to update their routing tables and create a consistent view of the network, using the best possible paths. Convergence occurs when switches transition to either forwarding or blocking modes. No data is forwarded during this time. Before data can be forwarded again, all devices must be updated.
What is Convergence Convergence is important to make sure all devices have the same database, but it does cost you some time: it usually takes several seconds to go from blocking to forwarding mode, and it is not recommended to change the default STP timers.
What is Convergence Routers share information with each other, but must individually recalculate their own routing tables For individual routing tables to be accurate, all routers must have a common view of the network topology When all routers in a network agree on the topology they are considered to have converged.
Why is Quick Convergence Important? When routers are in the process of convergence, the network is susceptible to routing problems because some routers learn that a link is down while others incorrectly believe that the link is still up It is virtually impossible for all routers in a network to simultaneously detect a topology change.
Convergence Issues Factors affecting the convergence time include the following: Routing protocol used Distance of the router, or the number of hops from the point of change Number of routers in the network that use dynamic routing protocols Bandwidth and traffic load on communications links Load on the router Traffic patterns in relation to the topology change
What are Autonomous Systems? An Autonomous System (AS) is a group of routers that share similar routing policies, rules, and operate within a single administrative domain. An AS can be a collection of routers running a single IGP, or it can be a collection of routers running different protocols all belonging to one organization. In either case, the outside world views the entire Autonomous System as a single entity.
Autonomous System AS Numbers Each AS has an identifying number that is assigned by an Internet registry or a service provider. This number is between 1 and 65,535. AS numbers within the range of 64,512 through 65,535are reserved for private use. This is similar to RFC 1918 IP addresses. Because of the finite number of available AS numbers, an organization must present justification of its need before it will be assigned an AS number. An organization will usually be a part of the AS of their ISP
Each AS has its own set of rules and policies. The AS number uniquely distinguish it from other ASs around the world.
Definitions Metric is a numeric value used by routing protocols to help determine the best path to a destination. RIP uses the metric hop count number. The lower the numeric value, the closer the destination. OSPF uses the metric bandwidth. EIGRP uses bandwidth
Definitions Definition: Flat network is one large collision domain and one large broadcast domain. Flat routing protocol is when all routing information is spread through the entire network.
Definitions Hierarchical routing protocol are typically classless link-state protocols. This means that classless means that routing updates include subnet masks in their routing updates. Administrative distance is the measure used by Cisco routers to select the best path when there are two or more different routes to the same destination from two different routing protocols. Administrative distance defines the reliability of a routing protocol. Each routing protocol is prioritized in order of most to least reliable (believable) using an administrative distance value. A lower numerical value is preferred.
EIGRP Characteristics EIGRP is an advanced distance vector protocol that employs the best features of link-state routing.
OSPF Characteristics OSPF is the standardized protocol for routing IPv4. Since it’s initial development, OSPF has been revised to be implemented with the latest router protocols. Developed for large networks (50 routers or more) Must be a backbone area Routers that operate on boundaries between the backbone and non-backbone are called, Area Border Routers (ABR) OSPF is a link state protocol
OSPF Characteristics When the OSPF topology table is fully populated, the SPF algorithm calculates the shortest path to the destination. Triggered updates and metric calculation based on the cost of a specific link ensure quick selection of the shortest path to the destination.
OSPF Characteristics OSPF is link-state routing protocol OSPF has fast convergence OSPF supports VLSM and CIDR
OSPF Characteristics Cisco’s OSPF metric is based on bandwidth OSPF only sends out changes when they occur. RIP sends entire routing table every 30 seconds, IGRP every 90 seconds OSPF also uses the concept of areas to implement hierarchical routing A large internetwork can be broken up into multiple areas for management and route summarization
OSPFCharacteristics Two open-standard routing protocols to choose from: RIP, simple but very limited, or OSPF, robust but more sophisticated to implement. EIGRP is Cisco proprietary
When all routers are configured into a single area, the convention is to use area 0(zero) If OSPF has more than one area, it must have an area 0 Multi-area OSPF becomes more complicated to configure and understand OSPF Routing Domain Single Area OSPF uses only one area, usually Area 0
OSPF Characteristics 1. Flooding of link-state information The first thing that happens is that each node, router, on the network announces its own piece of link-state information to all other routers on the network. This includes who their neighboring routers are and the cost of the link between them. Example: “Hi, I’m Router A, and I can reach Router B via a T1 link and I can reach Router C via an Ethernet link.” Each router sends these announcements to all of the routers in the network.
4. Shortest Path First Tree This algorithm creates an SPF tree, with the router making itself the root of the tree and the other routers and links to those routers, the various branches. 5. Routing Table Using this information, the router creates a routing table.
Large OSPF Networks Large link-state table Each router maintains a LSDB for all links in the area The LSDB requires the use of memory Frequent SPF calculations A topology change in an area causes each router to re-run SPF to rebuild the SPF tree and the routing table. A flapping link will affect an entire area. SPF re-calculations are done only for changes within that area.
Issues with large OSPFNetworks Large routing table Typically, the larger the area the larger the routing table. A larger routing table requires more memory and takes more time to perform the route look-ups. Solution: Divide the network into multiple areas
OSPF Uses “Areas” Hierarchical routing enables you to separate large internetworks (autonomous systems) into smaller internetworks that are called areas. With this technique, routing still occurs between the areas (called inter- area routing), but many of the smaller internal routing operations, such as recalculating the database –re-running the SPF algorithm, are restricted within an area
OSPF Uses “Areas” Changes in one area are generally not propagated (spread) to another Route summarization is extensively used in multi- area OSPF
Internal: Routers with all their interfaces within the same area Backbone: Routers with at least one interface connected to area 0 ASBR:(Autonomous System Boundary Router): Routers that have at least one interface connected to an external internetwork (another autonomous system) ABR: (Area Border Router): Routers with interfaces attached to multiple areas.
IS - IS Characteristics IS-IS is an Open System Interconnection (OSI) routing protocol originally specified by International Organization for Standardization (ISO) IS-IS is a dynamic, link-state, intra-domain, interior gateway protocol (IGP) IS-IS was designed to operate in an OSI Connectionless Network Service (CLNS) environment It was not originally designed to work with the IP protocol
IS - IS Characteristics Extensions were added so that IS-IS can route IP packets IS-IS operates at Layer 3 (Network) of the OSI model IS-IS selects routes based upon a cost metric assigned to links in the IS-IS network A two-level hierarchy is used to support large routing domains A large domain can be administratively divided into areas
OSPF and IS – IS Similarities Classless Link-state databases an Dijkstra’s algorithm Hello packets to form and maintain adjacencies Use areas to form hierarchical topologies Support address summarization between areas Link-state representation, aging, and metrics Update, decision, and flooding processes Convergence capabilities Deployed on ISP backbones
IS – IS and the OSI Protocol Suite The OSI suite of protocols were never widely implemented at the Layers 3-7 because the TCP/IP Protocols at these layers became the de-facto standard.
OSI Terminology End system (ES) is any non-routing network node (host) Intermediate system (IS) is a router An area is a logical entity formed by a set of contiguous routers, hosts, and the data links that connect them Domain is a collection of connected areas under a common administrative authority(think AS) The areas are connected to form a backbone
IS – IS is Designed to be Hierarchical An OSI network is a hierarchy of these entities: Domain -any portion of an OSI network under a common administration Area –a part of a domain, broken up for easier management Backbone –areas connect to other areas through the backbone
IS – IS is Hierarchical There are four levels of routing: Level 0, routing between an ES and IS Level 1, routing between ISs in the same area Level 2, routing between different areas in the same domain Level 3, routing between separate domains
Why use IS – IS instead of OSPF? IS-IS is more scalable than OSPF because it uses smaller LSPs for advertisements Up to 1000 routers can reside in an IS-IS area versus several hundred for OSPF IS-IS is more efficient with its updates and requires less CPU power IS-IS has more timers that can be fine-tuned to speed up convergence
EIGRP Characteristics Cisco proprietary, released in 1994 EIGRP is an advanced distance-vector routing protocol that relies on features commonly associated with link-state protocols. (sometimes called a hybrid routing protocol) Supports VLSM and CIDR Uses multicasts for communication –not broadcasts Establishes adjacencies with its neighbor routers by using a Hello protocol Keeps all routes in a topology table Has speed and efficiency of routing updates like a link-state protocol
EIGRP Metric Calculation By default, EIGRP uses only these: Bandwidth (carrying capacity) Delay (end-to-end travel time) If these are the default: Bandwidth (default) Delay (default) When are these used? load Reliability These values are used when the administrator manually enters them
EIGRP Terminology EIGRP uses DUAL, the Diffusing Update Algorithm to calculate routes –not Bellman-Ford algorithm. The lowest cost path to a destination is called the feasible distance (FD) The cost of the route as advertised by the neighboring router, is called reported distance (RD) The best (primary) route to a destination is called the successor route (successor) The next best route, (backup), if there is one, is called the feasible successor (FS)
EIGRP Tables The following three tables are maintained by EIGRP: Neighbor table Topology table Routing table
BGP BGP is a path vector routing protocol. Defined in RFC 1772 BGP is a distance vector routing protocol, in that it relies on downstream neighbors to pass along routes from their routing table. BGP uses a list of AS numbers through which a packet must pass to reach a destination.
BGP Basics Exchange routing information between autonomous systems Guarantee the selection of a loop free path. BGP4 is the first version of BGP that supports CIDR and route aggregation. Common IGPs such as RIP, OSPF, and EIGRP use technical metrics. BGP does not use technical metrics. BGP makes routing decisions based on network policies, or rules (later) BGP does not show the details of topologies within each AS. BGP sees only a tree of autonomous systems.
BGP Basics BGP updates are carried using TCP on port 179. In contrast, RIP updates use UDP port 520 OSPF, IGRP, EIGRP does not use a Layer 4 protocol Because BGP requires TCP, IP connectivity must exist between BGP peers. TCP connections must also be negotiated between them before updates can be exchanged. Therefore, BGP inherits those reliable, connection- oriented properties from TCP.
Loop Free Path To guarantee loop free path selection, BGP constructs a graph of autonomous systems based on the information exchanged between BGP neighbors. BGP views the whole internetwork as a graph, or tree, of autonomous systems. The connection between any two systems forms a path. The collection of path information is expressed as a sequence of AS numbers called the AS Path. This sequence forms a route to reach a specific destination
BGP Operation When two routers establish a TCP-enabled BGP connection between each other, they are called neighbors or peers. Each router running BGP is called a BGP speaker.
Analog and Digital Signaling The human voice generates sound waves A telephone converts the sound waves into analog signals. However, analog transmission is not particularly efficient. The PSTN is a collection of interconnected voice-oriented public telephone networks, both commercial and government- owned. The PSTN today consists almost entirely of digital technology, except for the final link from the central (local) telephone office to the user. To obtain clear voice connections, the PSTN switches convert analog speech to a digital format and send it over the digital network.
Analog and Digital Signaling The human voice generates sound waves To obtain clear voice connections, the PSTN switches convert analog speech to a digital format and send it over the digital network. At the other end of the connection, the digital signal is converted back to analog and to the normal sound waves that the ear can hear. Digital signals don’t pick up the noise levels as analog signals, and doesn’t induce any additional noise when amplifiing signals. Digital signals hold their original form better than analog signals over greater distances, regeneration, coded, and decoded translations.
Analog and Digital Signaling The range for speech is from 400 to 4000 hertz (hz). Higher frequencies are filtered. Sampling is the method used on analog signals to formalize the digitizing process. A voltage level corresponds to the amplitude of the signal.
Analog and Digital Signaling Pulse Code Modulation (PCM) is a digital representation of an analog signal where the magnitude of the signal is sampled regularly at uniform intervals, then quantized to a series of symbols in a numeric (usually binary) code. The standard word size is 8 bits.
Analog and Digital Signaling There are several steps involved in converting an analog signal into PCM digital format, as shown in the figure
Companding Signal is compressed for more efficient transmission, and less noise Two common methods: The A-law standard is used in Europe, Mu-law is used in North America and Japan The methods are similar—but they are not compatible
Analog and Digital Signaling 1.Filter analog signal – remove frequencies > 4000 hertz 2.Sample – rate at least twice the highest frequency according to Nyquist Theorem. Samples the filtered input signal at a constant frequency using Pulse Amplitude Modulation (PAM). 3.Digitize – occurs prior to transmission over the telephone network (PCM process)
Analog and Digital Signaling 4. Quantization and coding – A process that converts each analog sample value into a discrete value to which a unique digital code word is assigned. 5. Companding – A process in which compression is followed by expansion; often used for noise reduction in equipment, in which case compression is applied before noise exposure and expansion after exposure. A process in which the dynamic range of a signal is reduced for recording purposes and then expanded to its original value for reproduction or playback.
Companding A signal is compressed for more efficient transmission, and less noise Two common methods: The A-law standard is used in Europe, Mu-law is used in North America and Japan The methods are similar—but they are not compatible
Public Switched Telephone Network (PSTN) Telephones connect to a CO (Central Office) through the local loop The local loop is an analog connection All analog signals are converted to digital at the CO Except for the local loop the entire phone system is a modern digital network
What is a Private Branch Exchange (PBX)? PBX is a private telephone network used within a company. The users of the PBX phone system share a number of outside lines for making external phone calls. A PBX connects the internal telephones within a business and also connects them to the public switched telephone network (PSTN).
PBX Features A PBX is a business telephone system that provides business features such as call hold, call transfer, call forward, follow-me, call park, conference calls, music on hold, call history, and voice mail. Most of these features are not available in traditional PSTN switches. A PBX switch often connects to the PSTN through one or more T1 digital circuits. A PBX supports end-to-end digital transmission, employs PCM switching technology, and supports both analog and digital proprietary telephones
Trunk Line Capacity In this diagram, 7 telephones connect to the CO in Neighborhood A and 6 connect to the CO in Neighborhood B Design question: How many simultaneous conversations should this trunk line carry?
Trunk Line Capacity The science of Traffic Engineering answers this question
What is Traffic Engineering? Voice traffic engineering is the science of selecting the correct number of lines and the proper types of service to accommodate users. Detailed capacity planning of all network resources should be considered to minimize degraded voice service in integrated networks. We can calculate the bandwidth required to support a number of voice calls with a given probability that the call will go through
Terminology Blocking probability Grade of Service (GoS) Erlang Centum Call Second (CCS) Busy hour Busy Hour Traffic (BHT) Call Detail Record (CDR)
Definitions The blocking probability value describes the calls that cannot be completed because insufficient lines have been provided. For example, a blocking probability value of 0.01 means that 1 percent of calls would be blocked. GoS is the probability that a voice gateway will block a call while attempting to allocate circuits during the busiest hour. GoS is written as a blocking factor, Pxx, where xx is the percentage of calls that are blocked for a traffic system. For example, traffic facilities that require P01 GoS define a 1 percent probability of callers being blocked.
Definitions One Erlang equals one full hour, or 3600 seconds, of telephone conversation The busy hour is the 60-minute period in a given 24- hour period during which the maximum total traffic load occurs. The busy hour is sometimes called the peak hour. The BHT, in Erlang’s or CCSs, is the number of hours of traffic transported across a trunk group during the busy hour (the busiest hour of operation). A CDR is a record containing information about recent system usage, such as the identities of sources (points of origin), the identities of destinations (endpoints), the duration of each call, etc
Trunk Capacity Calculation For example, one hour of conversation (one Erlang might be ten 6-minute calls or 15 4-minute calls. Receiving 100 calls, with an average length of 6 minutes, in one hour is equivalent to ten Erlangs For example, if you know from your call logger that 350 calls are made on a trunk group in the busiest hour and that the average call duration is 180 seconds, you can calculate the BHT as follows: BHT = Average call duration (seconds) * calls per hour/3600 BHT = 180 * 350/3600 BHT = 17.5 Erlangs
Capacity Information There are years of data on the number and duration of a phone conversation This historical data can be used to calculate the capacity or number of trunk lines needed in a telephone system Erlang Tables are used for this calculation
What is an Erlang Table? Erlang tables show the amount of traffic potential (the BHT) for specified numbers of circuits for given probabilities of receiving a busy signal (the GoS) The BHT calculation results are stated in Erlangs Erlang tables combine offered traffic (the BHT), number of circuits, and GoS in the following traffic models:
What is an Erlang Table? Erlang B: This is the most common traffic model, which is used to calculate how many lines are required if the traffic (in Erlangs) during the busiest hour is known. The model assumes that all blocked calls are cleared immediately. Extended Erlang B: This model is similar to ErlangB, but it takes into account the additional traffic load caused by blocked callers who immediately try to call again. The retry percentage can be specified. Erlang C: This model assumes that all blocked calls stay in the system until they can be handled. This model can be applied to the design of call center staffing arrangements in which calls that cannot be answered immediately enter a queue
What is an Erlang Table? Erlang C: This model assumes that all blocked calls stay in the system until they can be handled. This model can be applied to the design of call center staffing arrangements in which calls that cannot be answered immediately enter a queue
Trunk Capacity Calculation The network design is based on a star topology that connects each branch office directly to the main office. There are approximately 15 people per branch office. The bidirectional voice and fax call volume totals about 2.5 hours per person per day (in each branch office). Approximately 20 percent of the total call volume is between the headquarters and each branch office. The busy-hour loading factor is 17 percent. In other words, the BHT is 17% of the total traffic. One 64-kbps circuit supports one call. The acceptable GoS is P05
Trunk Capacity Calculation 2.5 hours call volume per user per day * 15 users = 37.5 hours daily call volume per office 37.5 hours * 17 percent (busy-hour load) = 6.375 hours of traffic in the busy hour 6.375 hours * 60 minutes per hour = 382.5 minutes of traffic per busy hour 382.5 minutes per busy hour * 1 Erlang/60 minutes per busy hour = 6.375 Erlangs 6.375 Erlangs* 20 percent of traffic to headquarters = 1.275 Erlangs volume proposed
Final Calculation To determine the appropriate number of trunks required to transport the traffic, the next step is to consult the Erlangtable, given the desired GoS This organization chose a P05 GoS. Using the 1.275 Erlangsand GoS= P05, as well as the ErlangB table: http://www.erlang.com/calculator/erlb/ http://www.erlang.com/calculator/erlb/ four circuits are required for communication between each branch office and the headquarters office
What do the terms FXS and FXO mean? FXS and FXO are the name of ports used by Analog phone lines (also known as POTS -Plain Old Telephone Service) or phones. FXS -Foreign eXchange Subscriber interface is the port that actually delivers the analog line to the subscriber. In other words it is the ‘plug on the wall’ that delivers a dial tone, battery current and ring voltage. FXO -Foreign eXchange Office interface is the port that receives the analog line. It is the plug on the phone or fax machine, or the plug(s) on your analog phone system. It delivers an on-hook/off-hook indication (loop closure). Since the FXO port is attached to a device, such as a fax or phone, the device is often called the ‘FXO device’. FXO and FXS are always paired, i.e similar to a male / female plug. Without a PBX, a phone is connected directly to the FXS port provided by a telephone company
Connecting a Traditional PBX to the PSTN If you have a PBX, then you connect the lines provided by the telephone company to the PBX and then the phones to the PBX. Therefore, the PBX must have both FXO ports (to connect to the FXS ports provided by the telephone company) and FXS ports (to connect the phone or fax devices to).
Telephone Signaling In a telephony system, a signaling mechanism is required for establishing and disconnecting telephone communications.
Three Types of Signaling Used To Make a Phone Call Supervision signaling: Typically characterized as on-hook, off- hook, and ringing, supervision signaling alerts the CO switch to the state of the telephone on each local loop. Supervision signaling is used, for example, to initiate a telephone call request on a line or trunk and to hold or release an established connection. Address signaling: Used to pass dialed digits (pulse or DTMF) to a PBX or PSTN switch. These dialed digits provide the switch with a connection path to another telephone or customer premises equipment. Informational signaling: Includes dial tone, busy tone, reorder tone, and tones indicating that a receiver is off-hook or that no such number exists, such as those used with call progress indicators
Analog Telephony Signaling Loop start: Loop start is the simplest and least intelligent signaling protocol, and the most common form of local-loop signaling. Only for residential use. Ground start: Also called reverse battery, ground start is a modification of loop start that provides positive recognition of connects and disconnects (off-hook and on-hook)., PBXs typically use this type of signaling. E&M: E&M is a common trunk signaling technique used between PBXs.
Digital Telephone Signaling CAS CCS DPNSS ISDN QSIG Digital Signaling –standards based protocol to allow different vendor’s PBXs to communicate SS7 Digital Signaling -used within the PSTN for signaling between PSTN switches
Why Integrate Voice and Data Networks? Integrating data, voice, and video in a network enables vendors to introduce new features The unified communications network model enables distributed call routing, control, and application functions based on industry standards Enterprises can mix and match equipment from multiple vendors and geographically deploy these systems wherever they are needed Only one network to maintain
VoIP or IP Telephony? Cisco distinguishes between the two Most technical discussions don’t VoIP –analog phones and/or analog PBXs are still used, but the analog signals are converted to IP packets with a Voice Enabled router IP Telephony –IP phones are used; the system is completely IP. Specialized call processing software replaces the PBX –this may be called an IP PBX
VoIP Connection To setup a VoIP communication we need the do the following: The ADC (Analog to Digital Converter) converts analog voice to digital signals (bits) The voice data is compressed to send the fewest number of bits while still retaining the original information (Codec) Voice packets are sent using a real-time protocol (typically RTP over UDP over IP) We need a signaling protocol to call users: ITU-T H323 or SIP At the receiver we have to disassemble packets, extract data, then convert them to analog voice signals and send them to sound card (or phone) All that must be done in a real time fashion cause we cannot waiting for too long for a vocal answer! (QOS)
VoIP Technology VoIP is an “Overlay” technology VoIP is applied on top of an IP Network If the IP network is not working properly VoIP will simply be one more thing that is broken Make sure the IP network is working correctly FIRST--then implement VoIP
H.323 Protocol H.323 is a standard for teleconferencing that was developed by the International Telecommunications Union (ITU). It supports full multimedia audio, video and data transmission between groups of two or more participants, and it is designed to support large networks. H.323 is still a very important protocol, but it has fallen out of use for consumer VoIP products due to the fact that it is difficult to make it work through firewalls that are designed to protect computers running many different applications. It is a system best suited to large organizations that possess the technical skills to overcome these problems. As a solution for a home or small office telephony system it is best avoided
Session Initiation Protocol (SIP) SIP (Session Initiation Protocol) is an Internet Engineering Task Force (IETF) standard signaling protocol for teleconferencing, telephony, presence and event notification and instant messaging. It provides a mechanism for setting up and managing connections, but not for transporting the audio or video data. It is probably now the most widely used protocol for managing Internet telephony
SIP Protocols SIP-Session Initiation Protocol MegacoH.248 -Gateway Control Protocol MGCP-Media Gateway Control Protocol MIMERVP over IP -Remote Voice Protocol Over IP Specification SAPv2-Session Announcement Protocol SDP-Session Description Protocol SGCP-Simple Gateway Control Protocol Skinny-Skinny Client Control Protocol (SCCP
SIP Protocols Sip is the major VoIP protocol in use today Very similar to http Sip uses port 5060 Sip has the same Status Codes as http Instead of a getas in http, Sip issues an INVITE when someone makes a call. The following are SIP responses: 1xx Informational (e.g. 100 Trying, 180 Ringing) 2xx Successful (e.g. 200 OK, 202 Accepted) 3xx Redirection (e.g. 302 Moved Temporarily) 4xx Request Failure (e.g. 404 Not Found, 482 Loop Detected) 5xx Server Failure (e.g. 501 Not Implemented) 6xx Global Failure (e.g. 603 Decline
SIP VoIP System User agents or phones register with a SIP Proxy. To initiate a session, the caller (or User Agent Client) sends a request with the SIP URL of the called party. If the client knows the location of the other party it can send the request directly to their IP address; if not, the client can send it to a locally configured SIP network server. The server will resolve the called user's location and send the request to them. During the course of locating a user, one SIP network server can proxy or redirect the call to additional servers until it arrives at one that definitely knows the IP address where the called user can be found. Once found, the request is sent to the user.
SIP VoIP System If phone A know the location of phone B, it can call phone B directly without going through the proxy server Sip uses email-style addresses to identify users
RTP RTP is the Real-time Transport Protocol RTP is used by H.323 and SIP for the actual transmission of the VoIP packets RTP uses UDP Additionally, RTCP (Real-time Control Protocol) provides this information: Packet Loss Jitter Delay Signal Level Call Quality Metrics Echo Return Loss
OSI Model ISO Model LayerProtocol or Standard PresentationApplications/CODECS SessionH.323 and SIP TransportRTP / UDP / TCP NetworkIP – Non QoS Data LinkATM, FR, PPP, Ethernet
Cisco’s Solution IP Telephony The main component of Cisco’s solution is the Cisco Unified Communications Manager: It is a server used for call control and signaling. It replaces a PBX. The IP phone itself performs voice-to-IP conversion, and voice-enabled routers are not required within the enterprise network. If connection to the PSTN is required, a voice- enabled router or other gateway must be added where calls are forwarded to the PSTN.
Definition of CODEC A codec is a device or computer program capable of encoding and/or decoding a digital data stream or signal. The word codec is a portmanteau of 'compressor-decompressor' or, more commonly, 'coder-decoder‘.
Voice Coding and Compression CODEC A DSP (Digital Signal Processor is a hardware component that converts the analog signal to digital format Codecs are software drivers that are used to encode the speech in a compact enough form that they can be sent in real time across a network using the bandwidth available Codecs are implemented within a DSP VoIP software or hardware may give you the option to specify the codecs you prefer to use This allows you to make a choice between voice qualityand network bandwidth usage, which might be necessary if you want to allow multiple simultaneous calls to be held using an ordinary broadband connection
Coding and Compression Algorithm The different codecs provide a certain quality of speech Advances in technology have greatly improved the quality of compressed voice and have resulted in a variety of coding and compression algorithms PCM: The toll quality voice expected from the PSTN. PCM runs at 64 kbps and provides no compression, and therefore no opportunity for bandwidth savings The other algorithms use compression to save bandwidth Voice quality is affected
Which CODEC is most affective? G.729 is the recommended voice codec for most WAN networks (that do not do multiple encodings) because of its relatively low bandwidth requirements and high mean opinion score (MOS) (ITU-T P.800)
Reducing the Amount of Voice Traffic The codecs chosen are a trade-off between bandwidth and voice quality Two techniques used to reduce voice traffic: cRTP
Every IP packet consists of a header and the payload (data, voice) Although the payload of a voice packet is small (20 bytes when G.729 is used), the header is 40 bytes cRTP compresses the header to 2 or 4 bytes Use on slow WAN links, but it is CPU intensive
VAD Voice Activity Detection On average, about 35 percent of calls are silence In traditional voice networks, all voice calls use a fixed bandwidth of 64 kbps regardless of how much of the conversation is speech and how much is silence When VoIP is used, this silence is packetized along with the conversation. VAD suppresses packets of silence, so instead of sending IP packets of silence, only IP packets of conversation are sent Therefore, gateways can interleave data traffic with actual voice conversation traffic, resulting in more effective use of the network bandwidth
QoS for Voice Classify Packets Mark Packets Marked packets can be prioritized in the scheme of queuing LLQ –Cisco’s Low Latency Queuing is the recommended method for VoIP networks
CAC –Call Admission Control CAC protects voice traffic from being negatively affected by other voice traffic by keeping excess voice traffic off the network. If a WAN link is fully utilized with voice traffic then adding more voice calls will degrade all the calls CAC checks if the link is maximized and won’t allow new calls to go through until bandwidth is available Callers will get a busy signal or “all circuits busy message”
LFI Link fragmentation and interleaving ensures that small voice packets don’t get stuck behind a large data packet The data packets are fragmented into smaller packets The voice packets can slip in between them
Upcoming Deadlines Assignment 10-1, Concept Questions 7 due November 17, 2010. Assignment 11-1, Concept Questions 8 due November 24, 2010.