[Integrated VOIP System]

Author's Acknowledgments
We would like to thank ALLAH for helping us in our work and giving us patience till we finished. Then of course we would like to thank:

Dr. AHMED EL-SHAZLY
Who was more than a supervisor, he was like a father and spent alot of his time in helping us and and giving us advices from his long experience. Finally we thank every one who helped in this project.

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[Integrated VOIP System]

Prepared By
ADNAN ADEL AMIN MOHAMMAD INAS ADEL YOSEF ELTIJANI OSMAN MOHAMMAD HANAFY ALI AHMED REFAT TWFIQ MOHAMMAD ADEL MORSY MOHAMMAD ABD EL-MAKSOUD MOHAMMAD IBRAHIM MEDHAT IBRAHIM AMGAD EHAB SCANDAR

3

.............................................................................................2 Synchronization sources........................2.......................................................24 2........4 Real-Time Voice in a Best-Effort IP Internetwork.2................................................................................... Delay........................................……………………......24 2...........................................................5 RTP Protocol Definition.......................22 2..............................................49 4 ....5 SIP Addressing..............................27 2.........................................8 Reliability and Availability.5......................2 Message Encapsulation..................................................................................2........33 2...............................................1 RTCP Packet Types..............................1 Brief history................36 2........11 1...1 Sessions...........................6 Call Setup Models...[Integrated VOIP System] TABLE OF CONTENTS Chapter1: Voip Overview………………………………....27 2..............................................1 Introduction.....................................................38 2......16 1...2..1 Overview...........28 2..........................4 Status Codes........21 2..............................................44 3................2...................42 3..................................................................................31 2..................2 Components of SIP...............................2.......................3 RTP Protocol Architecture (Protocol layering).................................3 SIP Messages.........2 RTP functions..........32 2......7 Reordering of Voice Packets..................4 RTCP Packet Structure.................................................24 2....................1 Fixed RTP Header Format.............................................................................................5 Packet Loss..............................................18 Chapter2: Major Voip Protocols……………………………………………………..................................................................................................4............................................40 3.2..38 Chapter3: Sip Protocol……………………………………………………………………………......2 RTP and RTCP.......................................46 3....................................................................................7 1....................................................................................................2.....14 1............31 2...............4.41 3...6 Consistent Throughput..............................................……..................................................................................1 Overview............25 2.........................4 RTP Concepts and Terms.....................................................................................5..................3 Reducing Header Overhead with CRTP....................................................................…….......2..................3 IP Networking Overview..........47 3.......9 1.......5 CRTP Packet Components............................................8 1............................4......................................4..............................8 1..………..........................................2 Extended RTP Header Format............................................................................2......2 Advantages and disadvantages of VOIP..........................28 2.................................................................3 Mixers...........................................................................................6 When to Use RTP Header Compression.24 2..........................................25 2.... and Jitter.............10 1.4....................................

................6 IP PBX.................................76 5..........12 Switching Unit..............................................10............76 5................................1 Ring Detector..............8 Block diagram of the overall system................................……53 4...........................................76 5.................10...80 5...............11...4......................................................................................................10..........3 Hook Switch...........................4 PBX..............9 Analog PBX....58 4.6 Melody player Circuit Schematic.................................9 Ethernet frames……………………………………………………………………...80 5.......................4..................................4 Hosted PBX systems...................................65 4..................................................................81 5.......................1 Slic Circuit Schematic...................79 5.....................................4....................77 5.8 Ethernet standards…………………………………………………………………...........................2 Ring detector Circuit Schematic.........................61 4...................2 The OSI Model................................................................70 4..........6.....80 5.................81 5...........................81 5...........82 5 ..................................................................................................................................................................................3 The TCP/IP Protocol Suite........................................76 5........................................2 Objectives.....................................4..............81 5.........................5 IP Addresses.................78 5................................80 5..........................................................................................................3 Terminals.....79 5............82 5...........................77 5..................................................................54 4......................................[Integrated VOIP System] Chapter4: Networking Basics……………………………………………………..................................10 Switches………………………………………………………………………….1 Overview...................1 User Datagram Protocol (UDP)..........................2 Transmission Control Protocol (TCP)......................4 Hook switch Circuit Schematic....................................................65 4...................78 5.................................10...........................................................................................................................10 COU(central office unit)………….........6......................4.............69 4.....................................5 Mobile PBX....................2 PBX System Components.....4 The Internet Protocol (IP)...................................................1 Overview.............................................54 4..............................................................................1 Tone generator Circuit Schematic.................13............................................................10................................................................................................................82 5.....................76 5....................76 5..........................................................72 4...............................................................7 Server......................................................................78 5.........................................82 5.................1 Introduction to Networking...................................................................................................................................7 Ethernet……………………………………………………………………………...........70 4...........66 4.4.....................................................................................6 Transport Layer protocols.5 Melody Player…………………………………………………………….....................................80 5...................11 SLIC (subscriber line interface card ) ..................63 4.......................73 Chapter5:Voip System Requirements…………………………………………………75 5..78 5........13 Tone Generator............11 Routers……………………………………………………………………………........................................................10............3 PBX functions..............5 Service....................................................6 Voip Gateways.........................................................

.........................................................83 5..[Integrated VOIP System] 5.........................15........83 5...................95 6......................1 Power supply circuit schematic........................................................................................84 5.......84 5...................2 Trixbox installation...........................................................5 Softphone Configuration......15 DTMF Receiver (decoder) ........85 5.......2 Dtmf decoder Circuit Schematic..........................86 Chapter6:System Configuration…………………………………………………….....................................88 6.....1 How does it work? .............................................................................................98 APPENDIX…………………………………………………………………….16..............................................................................3 Trixbox Configuration......19 Why using trixbox..............1 Overview..............................................................................88 6.................20 Software PBX Vs hardware PBX.............................85 5......................................................................................................................................................................….........................92 6..................................................................................................85 5...................16 Power Supply.....4 IP phone configuration.....................................14 Ring Generator............................................83 5.................................87 6.....................17 Software PBX.....................…..................................15...................103 6 .............................................................................84 5.........................18 Different Software Used as SoftPBX.........

[Integrated VOIP System] About the book: This book is written for anyone who wants to reduce or eliminate the toll charges by applying the voip system as an upgrading level for the computer network. 7 . And not only descriptions and dreams but also detailed practical procedure for applying the system .

[Integrated VOIP System] Chapter 1 Voip Overview 8 .

allowing call monitoring. However. However this PC-to-PC internet telephony only worked if both parties were using the Internet Phone software. the implementation of additional services such as voicemail. there are a number of additional benefits. Equipped with voice-processing cards. it is innately scalable. Voice over IP. primarily the cost savings that can be made by avoiding the use of the traditional PSTN. The forum also worked on the standardisation of directory services and the use of touch-tone standards for accessing voicemail. these gateway servers enable users to communicate via standard telephones over great distances without using the ‗long distance‘ telephone network. is simply a means of making tel ephone calls over a data network instead of over the traditional analogue public switched telephone network (PSTN). Inherent in the term is the management of the protocol. including lower telecom charges (cheaper calls) flexibility reduced infrastructure Integrated services and greater user features. speakers. The software encoded and compressed the voice signal. 9 . and a reduction in the physical cabling required for new installations. poor quality. and a range of other features to be implemented. Vocalec and 3Com) to promote and develop the use of the International Telecommunications Union‘s (ITU) H. The voice data can be easily processed using a standard PC.323 protocol. In addition to the financial savings. In general. the digital nature of VOIP allows easy administration. The ability to transfer voice over the internet. or Voice over Internet Protocol to use the full title. The major advantage of VOIP is cost saving (although do not expect immediate cost savings due to the upgrading of existing infrastructure and systems implementation).[Integrated VOIP System] 1. and frequent delays. Internet telephony has made a number of important advances since 1995. and modem.2 Advantages and disadvantages of VOIP VOIP offers a number of potential advantages and benefits to its users:     reduced operating cost. There are many advantages to this method of telephony. microphone. For example.1 Brief history The term VOIP was first used by the VOIP Forum (a group of major companies including Cisco. In addition to cost savings. gateway servers are emerging to act as an interface between the internet and the PSTN. 1. Many software developers now offer PC telephony software but. rather than the PSTN was first made possible in February 1995 when Vocaltec released its Internet Phone software. allowing the quick and easy addition of new terminals and connections. this means that the voice information is encoded into discrete digital packets and then transferred across an IP-based network. Users also had to contend with the lack of any kind of directory service. more importantly. due to the distributed nature of the internet the Quality of Service (QoS) can often suffer. converting it into IP packets then transmitted over the internet. and there are still a number of technical issues affecting the widespread adoption of VOIP. voicemail. no formalised ringing protocols meant that it was necessary for both parties to pre-arrange the time of the call in order to be available to make the final connections themselves. Inherent in this technology were a number of severe limitations. The term VOIP describes the use of the Internet Protocol (IP) to transfer speech between two or more sites. This software was designed to run on a standard personal computer (PC) equipped with a sound card.

(Firewalls need to be H. IP can be used to communicate across any set of interconnected networks and is equally suited to both LAN and WAN communication. although the connectionless network remains unaware of the virtual circuit (VC). IP is a connectionless network protocol. The concept of session establishment exists between end systems. and a large number of products on the market implement both technologies in a single package. 10 . A side benefit of VOIP is that many of the steps to full implementation are the same as the steps required to implement video conferencing and similar technologies. reliable and offer high quality of service Compatibility with existing firewalls and security devices may cause problems. it can transport IP packets over deterministic and nondeterministic Layer 2 protocols. IP resides at the network layer of the Open System Interconnection (OSI) protocol stack.323 and possibly SIP compliant depending on the VOIP solution. such as Frame Relay or ATM.) 1. VOIP systems can also be configured using standard networking tools such as SNMP (simple network management protocol). there are drawbacks:   the network has to be fast. However.[Integrated VOIP System] A single network topology can be put in place.3 IP Networking Overview This topic provides an overview of IP networking and some of the inherent challenges when conveying voice over an IP network. Connectionless networks generally do not participate in signaling. Whilst VOIP is not strictly speaking a subset of video conferencing there are many similarities. Ther efore. requiring only network cabling to be installed in a building rather than both data and traditional telephone cabling.

and loss. causing delay and delay variation at the receiving end. UDP is the connectionless transport layer protocol used for VoIP. and packet ordering. IP traffic transmits on a FIFO basis. These problems include jitter. The user must have a well-engineered network. they arrive with varying delays and out of sequence. end to end. a voice packet is treated as just another data packet. Data networks were not designed to carry voice traffic. These problems must be addressed with QoS mechanisms. when running delay-sensitive applications such as VoIP. jitter. allowing large file transfers to take advantage of the efficiency that is associated with larger packet sizes. The figure shows how packets may be received out of sequence or become completely lost at the receiving end. FIFO queuing affects the way that voice packets transmit. The traditional telephony network was originally designed to carry voice.[Integrated VOIP System] IP information is transferred in a sequence of datagrams. As they traverse the IP network. As the packets arrive at the destination router on the right. Fine-tuning the network to adequately support VoIP involves a series of protocols and features geared toward QoS. while other packets traverse a different path. Traditionally. A message is sent as a series of datagrams that are reassembled into the completed message at the receiving location. In the absence of any special QoS parameters. all packets must be received in sequence i mmediately and without interpacket variable delay. Example: IP Networking Due to the very nature of IP networking. voice traffic is real-time traffic that requires a certain quality of service (QoS). UDP is a simple protocol that exchanges datagrams without acknowledgments or guaranteed delivery. voice packets sent across IP will be subject to certain transmission problems. Different packet types vary in size.4 Real-Time Voice in a Best-Effort IP Internetwork This topic lists problems associated with implementation of real-time voice traffic in a best-effort IP internetwork. Because a voice conversation that is transported in IP can be considered a continuous audio file. delay. Example: Real-Time Voice Delivery Issues 11 . packets sent from the originating router on the left are in sequence and sent with predictable transmission intervals. Although data traffic is best-effort traffic and can with stand some amount of delay. the routing protocol may send some of the packets through one path. 1. UDP requires that other protocols handle error processing and retransmission. In the figure. The design of circuit switched calls provides a guaranteed path and a delay threshold between source and destination The IP network was originally designed to carry data.

and minimal packet loss. Delay. and Jitter This topic discusses the causes of packet loss. and packet loss to the speech signal. echo. If a VoIP network is to provide the same quality that users have come to expect from traditional telephony services.5 Packet Loss. The far-end voice-enabled router or gateway has to re-sort the packets and adjust the interpacket interval for a proper-sounding voice play out. Over the long term. Configuring voice in a data network environment requires network services with low delay. Because each of these routes may have different delay characteristics. end-to-end delay. VoIP is susceptible to these network behaviors. jitter. and the associated jitter. minimal jitter. This condition is called jitter. 1. the arrival rate of the packets may vary. delay.[Integrated VOIP System] In the IP network shown in the figure. then the network must ensure that the delay in transmitting a voice packet across the network. In traditional telephony networks. voice has a guaranteed delay across the network by strict bandwidth a ssociation with each voice stream. Network transmission adds corruptive effects like noise. does not exceed specific thresholds. and jitter delay in an IP internetwork. delay. Another effect of multiple routes is that voice packets can arrive out of order. as follows: 12 . voice packets that enter the network at a constant rate can reach the intended destination by a number of routes. packet loss. and jitter will all affect voice quality. which can degrade the voice application.

or if there is too much variable delay in the network. the greater the serialization delay. it can be considered variable because a larger data packet can enter the egress queue at any time before a voice packet. if the network is congested. and voice packetization time for various VoIP codecs. the delay incurred by the voice packet is its own serialization delay. End-to-end delay consists of the following two components:  Fixed network delay: You should examine fixed network delay during the initial design of the VoIP network. Codec algorithms can correct small amounts of loss. Research at Cisco Systems has shown that there is a negligible difference in voice quality scores using networks built with 200-ms delay budgets. but too much loss can cause voice clipping and skips. plus the serialization delay of the data packet in front of it. The International Telecommunication Union (ITU) standard G. Although this ratio is known. The chief cause of packet loss is network congestion.114 states that a oneway delay budget of 150 ms is acceptable for high.[Integrated VOIP System] Packet loss: The IP network may drop voice packets if the network quality is poor. Serialization delay is a constant function of link speed and packet size. Delay: End-to-end delay is the time that it takes the sending endpoint to send the packet to the receiving endpoint. voice encoding delay. The larger the packet and the slower the link clocking speed. 13   . If the voice packet must wait for the data packet to serialize. Examples of fixed network delay include propagation delay of signals between the sending and receiving endpoints.quality voice. Variable network delay: Congested egress queues and serialization delays on network interfaces can cause variable packet delays.

[Integrated VOIP System] 14 .

[Integrated VOIP System] Jitter: Jitter is the variation between the expected arrival of a packet and when it is actually r eceived. The data stream may be affected by error conditions in the network.. and jitter can be heard as follows: The calling party says..Good m. . This instantaneous buffer use can lead to a difference in delay times between packets in the same voice stream.. how.6 Consistent Throughput This topic describes the methods that you can use to ensure consistent delivery and throughput of voice packets in an IP internetwork. how are you? With jitter.Good morning.. .. delay. the called party hears. the called party hears. The amount of data that is placed in the pipe at the originating end is not necessarily the same amount of data that comes out at the destination. how are you? With end-to-end delay. because network congestion can be encountered at any time within a network. To compensate for these delay variations between voice packets in a conversation. however.ning. and Jitter Problems The effect of end-to-end packet loss. the called party hears. .Good morning. . w are you? 1. Buffers can fill instantaneously. Throughput is the actual amount of useful data that is transmitted from a source to a destination.morning.are you? With packet loss. Example: Packet Loss. VoIP endpoints use jitter buffers to turn the delay variations into a constant value so that voice can be played out smoothly. for 15 .Good. Delay.

Low Latency Queuing (LLQ) is one of the newest Cisco queuing mechanisms. Cisco routers offer several different queuing mechanisms that can be implemented based on traffic requirements. Compressed RTP (CRTP) reduces the headers to 2 bytes in most cases. voice is carried in Real-Time Transport Protocol (RTP). These techniques provide preferential treatment under congestion situations for premium (priority) class traffic. leaving the packet unusable. This constitutes 40 bytes of RTP/UDP/IP header. Weighted random early detection (WRED) is one of the QoS congestion avoidance mechanisms used in Cisco IOS software. At the same time. Cisco IOS software uses a number of techniques to reliably deliver real-time voice traffic across the modern data network. 16   . using twice the amount of bandwidth for that packet. thus saving considerable bandwidth and providing for better throughput. include the following:  Queuing: The act of holding packets so that they can be handled with a specific priority when lea ving the router interface. which is carried in User Datagram Protocol (UDP). voice had guaranteed bandwidth associated with each voice stream. potentially forcing a retransmit. prioritize traffic. Packets may also be dropped during times of congestion. Queuing enables routers and switches to handle bursts of traffic. The aim is to anticipate and avoid congestion at common network and internetwork bottlenecks before it becomes a problem. Header compression: In the IP environment.[Integrated VOIP System] example. In the traditional telephony network. which is then put inside an IP packet. This header size is large when compared to the typical voice payload of 20 bytes. Congestion avoidance: Congestion avoidance techniques monitor network traffic loads. bits may be corrupted in transit. such as voice. These techniques. these techniques maximize network throughput and capacity use and minimize packet loss and delay. which all work together to ensure consistent delivery and throughput of voice packets. and allocate bandwidth. measure network congestion.

Because voice rides in UDP/IP packets. both voice and data packets can be carried together on low-speed links without causing excessive delay to the real-time voice traffic. real-time voice packets are queued until the large data packet is transmitted. voice traffic on the network is assured priority and its delivery is more consistent. Unlike routing protocols.[Integrated VOIP System]  Resource Reservation Protocol: Resource Reservation Protocol (RSVP) is a transport layer protocol that enables a network to provide differentiated levels of service to specific flows of data.7 Reordering of Voice Packets This topic describes how RTP ensures consistent delivery order of voice packets in an IP internetwork. In this way. Voice calls in the IP environment can request RSVP service to provide guaranteed bandwidth for a voice call in a congested environment. for example.  There are many QoS tools that can be used to ensure consistent throughput. This delay is unacceptable for voice traffic. Because the path is circuit-switched. Because IP provides connectionless transport with the possibility of multiple paths between sites. 17 . Fragmentation: Fragmentation defines the maximum size for a data packet and is used in the voice environment to prevent excessive serialization delays. the path between the source and destination is reserved for the duration of the call. Hosts use RSVP to request a QoS level from the network on behalf of an application data stream. Data flows consist of discrete sessions between specific source and destination machines. voice samples are carried in an orderly manner through the use of timedivision multiplexing (TDM). a 1500-byte packet takes 187 ms to leave the router over a 64-kbps link. there is no automatic reordering of packets. Routers use RSVP to deliver QoS requests to other routers along the paths of the data stream. RSVP is designed to manage flows of data rather than make decisions for each individual datagram. If a best-effort data packet of 1500 bytes is sent. 1. In traditional telephony networks. they can be interleaved with real-time (voice) packets. After an RSVP reservation is made. When these mechanisms are employed. However. All of the voice samples stay in order as they are transmitted across the wire. weighted fair queuing (WFQ) is the mechanism that actually delivers the queue space at each device. if best-effort data packets are fragmented into smaller pieces. voice packets cannot arrive out of order at the destination. Serialization delay is the time that it takes to actually place the bits onto an interface.

such as interactive voice and video. 18 . the services provided by RTP include payload-type identification. and delivery monitoring. time stamping. According to RFC 1889. sequence numbering.[Integrated VOIP System] RTP provides end-to-end delivery services for data that require real-time support.

8 Reliability and Availability 19 .[Integrated VOIP System] Example: Reordering Voice Packets In the figure. RTP reorders the voice packets through the use of sequence numbers before playing them out to the user. 1. The table illustrates the various stages of packet reordering by RTP.

This topic describes methods that you can use to improve reliability and availability in data networks. Now the network devices must have protected power to continue to function and provide power to the end devices. Many data networks cannot make the same claim. or switches for IP Phones. This corresponds to 5. Efforts to ensure reliability may include choosing hardware and software with a low mean time between failure. provide an uninterruptible power supply (UPS) to these devices in addition to providing network availability. This delay is una cceptable for telephony users. and taking steps to correct problems in design of the network as it grows. the 20 . it may not come back up for minutes or even hours.[Integrated VOIP System] The traditional telephony network strives to provide 99. This is why telephone companies can claim a high availability rate. In a fully redundant network. When a user wants to make a call. To provide telephony users the same or close to the same level of service as they experience with traditional telephony.999 percent uptime to the user. Efforts to ensure availability may include installing proactive network management to predict failures before they happen. switches have multiple redundant connections to other switches. the reliability and availability of the data network takes on new importance. for example. Administrators must. Reliability is a measure of how resilient a network can be. the telephone company can route the call in different ways. or installing redundant hardware and links. If either a link or a switch becomes unavailable.25 minutes per year of downtime. now find that their connectivity is terminated. In traditional telephony. Previously. Local users with network equipment. gateways. the network should be accessible to that user at any time a call is required. therefore. depending on the type of connection the user had. Availability is a measure of how accessible the network is to the users. such as voice enabled routers. When the data network goes down. they received their power directly from the telephone company central office (CO) or through a UPS that was connected to their key switch or PBX in the event of a power outage. High availability encompasses many areas of the network. Network reliability comes from incorporating redundancy into the network design.

even through different providers  Power supplies and UPSs 21 .[Integrated VOIP System] following components need to be duplicated:  Servers and call managers  Access layer devices. such as WAN links and public switched telephone network (PSTN)  gateways. such as multilayer switches  Interconnections. such as LAN switches  Distribution layer devices. such as routers or multilayer switches  Core layer devices.

[Integrated VOIP System] Chapter 2 Major VoIP Protocols 22 .

proxy. Specified in RFC 2705. MGCP defines a protocol to control VoIP gateways connected to external call-control devices. referred to as call agents. In essence. SIP and its partner protocols. such as H. provide announcements and information about multicast sessions to users on a network.[Integrated VOIP System] 2. and similar header and response 23 . SIP: A detailed protocol that specifies the commands and responses to set up and tear down calls. video. Session Announcement Protocol (SAP) and Session Description Protocol (SDP).323: An ITU standard protocol for interactive conferencing. SIP defines end-toend call signaling between devices. The H.323 is an umbrella of standards that defines all aspects of synchronized voice.323 defines end-to-end call signaling. The major VoIP protocols include the following:  H. SIP is a text-based protocol that borrows many elements of HTTP. MGCP provides the signaling capability for less expensive edge devices. any time an event such as off hook occurs at the voice port of a gateway. It also details features such as security. such as gateways.1 Overview This topic defines the major VoIP protocols and matches them with the seven layers of the OSI model. that may not contain a full voice-signaling stack. and data transmission. using the same transaction request and response model. The ITU standard protocol was originally designed for multimedia in a connectionless environment. and transport (TCP or UDP) services. such as a LAN. H. The call agent then signals that device to provide a service. MGCP: An emerging standard for PSTN gateway control or thin device control.323. the voice port reports that event to the call agent. such as dial-tone signaling.

the user must make the connectionless network appear more connection-oriented. The user can configure which codec to use or a codec is negotiated according to what is available. RTP provides sequence numbers and time stamps for the orderly processing of voice packets. It also adopts a modified form of the URL-addressing scheme used within e-mail that is based on Simple Mail Transfer Protocol (SMTP). so human speech is the application. RTCP is used for QoS reporting. Example: VoIP and the OSI Model Successfully integrating connection-oriented voice traffic in a connectionless-oriented IP network requires enhancements to the signaling stack. Applications such as Cisco IP Soft phone and Cisco Call Manager provide the interface for users to originate voice at their PCs or laptops and convert and compress it before passing it to the network. Every RTP flow has a corresponding RTCP flow that reports statistics on the call. 24 . RTP carries the voice payload across the network. In some ways.[Integrated VOIP System] codes. RTCP: Provides out-of-band control information for an RTP flow. RTP: An Internet Engineering Task Force (IETF) standard media-streaming protocol. Codecs define how the voice is compressed. a standard telephone becomes the interface to users. If a gateway is used.

over unicast and multicast network services.2 RTP and RTCP RTP (version 2) is a real-time transport protocol that provides end-to-end delivery services to support applications transmitting real-time audio.  Guarantee delivery or prevent out-of-order delivery. The variables in VoIP are the signaling methods used. Loss detection for quality estimation. but relies on lower-layer services to do so.2 RTP functions Segmentation/reassembly done by UDP (or similar). Other transport protocols besides UDP can carry RTP as well. 2. the packets must be reordered and resynchronized before playing them out to the user. Voice traffic still flows directly from endpoint to endpoint.2. RTP typically runs on top of UDP to utilize its multiplexing and checksum services. Quality-of-service feedback and rate adaptation. recovery. MGCP uses a call agent to control signaling on behalf of the endpoint devices.2. RTP provides sequencing functionality. Source identification. Since UDP does not provide services such as sequence numbers or time stamps.2. Intra-media synchronization: remove delay jitter through playout buffer. 2. 2. RTP provides end-to-end delivery services.[Integrated VOIP System] One of the constants in VoIP implementation is that voice uses RTP inside of UDP to carry the payload across the network. such as gateways. The central control device participates in the call setup only. Intra-media synchronization: drifting sampling clocks. RTP does not:  Provide any mechanism to guarantee quality-of-service. Resequencing (if needed). along with a profile for carrying audio and video over RTP in RFC 1890. H. but it does not provide all of the functionality that is typically provided by a transport protocol. 2.3 RTP Protocol Architecture (Protocol layering) 25 . In fact. Because IP voice packets can reach the destination out of order and unsynchronized. MGCP defines a method to separate the signaling function from the voice call function. Inter-media synchronization (lip sync between audio and video).1 Introduction RTP is defined in IETF RFC 1889.323 and SIP define end-to-end call signaling methods.

datagram protocol Traditional transport services such as:     Addressing.2. Quality-of-service.4 RTP Concepts and Terms 2. and Delivery semantics Are all provided by a lower level protocol 2. If multiple media types are communicated by the group. the transmission of each medium constitutes a session.1 Sessions An RTP session is the sending and receiving of RTP data by a group of participants. Segmentation/reassembly.4. For each participant a session is a (pair of) transport addresses used by a participant to communicate with the group. 26 .2.[Integrated VOIP System] RTP is an application-level.

[Integrated VOIP System] 27 .

4. Timestamp. Resulting stream is multicast to a new group address. Sequence number.2 Synchronization sources Each source of RTP packets is called a synchronization source which identified by a unique. 2. 28 .4. randomly chosen 32-bit ID (the SSRC).[Integrated VOIP System] 2. Media data unit(s).2.3 Mixers An RTP mixer is an intermediate system that receives & combines packets of one or more RTP sessions into a new packet.2. A basic RTP message consists of Synchronization source identifier of sender. A mixer will typically have to define synchronization relationships between streams. special effects may be performed. A host generating multiple streams within a single RTP session must use a different SSRC per stream. Streams may be transcoded.

As an example. In addition. thus allowing multiple data and compression types. 2. a payload format might specify what type of audio or video encoding is carried in the RTP packet.5 RTP Protocol Definition 2. Sources that are mixed together become contributing sources (CSRC).2.2.5.1 Fixed RTP Header Format The RTP header provides the timing information necessary to synchronize and display audio and video data and to determine whether packets have been lost or have arrived out of order.[Integrated VOIP System] Mixers are synchronization sources. Encoded data can be compressed before delivery. RTP is tailored to a specific application via auxiliary profile and payload format specifications. the header specifies the payload type. 29 .

Marker (M): 1 bit For voice packets. (The value 1 is used by the first draft version of RTP and the value 0 is used by the protocol initially implemented in the "vat" audio tool). the packet contains one or more additional padding octets at the end which are not part of the payload. The fields have the following meaning: Version (V): 2 bits This field identifies the version of RTP. the fixed header is followed by exactly one header extension. CSRC count (CS cnt): 4 bits The CSRC count contains the number of CSRC identifiers that follow the fixed header. Padding may be needed by some encryption algorithms with fixed block sizes or for carrying several RTP packets in a lower-layer protocol data unit. while listeners generally are not sensitive to slight variations in the durations of a pause. The version defined by this specification is two (2). the marker bit indicates the beginning of a talk spurt.[Integrated VOIP System] The first twelve octets are present in every RTP packet. Beginning of talk spurts are good opportunities to adjust the play out delay at the receiver to compensate for differences between the sender and receiver clock rates as well as changes in the network delay jitter. The last octet of the padding contains a count of how many padding octets should be ignored. Extension (X): 1 bit: If the extension bit is set. Padding (P): 1 bit If the padding bit is set. 30 . Packets during a talk spurt need to be played out continuously.

and may be used by the receiver to detect packet loss and to restore packet sequence. Applications without silence suppression set the bit to zero. SSRC: 32 bits 31 . the first packet of a talk spurt (first packet after a silence period) is distinguished by setting the marker bit in the RTP data header. Additional payload type codes may be defined dynamically through non-RTP means. The sampling instant must be derived from a clock that increments monotonically and linearly in time to allow synchronization and jitter calculations. because the packets may flow through a translator that does. Sequence number: 16 bits The sequence number increments by one for each RTP data packet sent. The initial value of the sequence number is random (unpredictable) to make known-plaintext attacks on encryption more difficult. An RTP sender emits a single RTP payload type at any given time. even if the source itself does not encrypt. Timestamp: 32 bits The timestamp reflects the sampling instant of the first octet in the RTP data packet. Payload type (PT): 7 bits This field identifies the format of the RTP payload and determines its interpretation by the application. A profile specifies a default static mapping of payload type codes to payload formats. this field is not intended for multiplexing separate media streams. The clock frequency is dependent on the format of data carried as payload and is specified statically in the profile or payload format specification that defines the format.[Integrated VOIP System] Silence Suppression For applications which send no packets during silence.

only 15 may be identified. for audio packets the SSRC identifiers of all sources that were mixed together to create packets are listed. all RTP implementations must be prepared to detect and resolve collisions.2 Extended RTP Header Format CSRC list: 0 to 15 items. with the intent that no two synchronization sources within the same RTP session will have the same SSRC identifier. Although the probability of multiple sources choosing the same identifier is low. Feedback of information to the application can be used to control performance and for diagnostic purposes.2. nu m- 32 . The number of identifiers is given by the CC field. CSRC identifiers are inserted by mixers. For example. Each participant in an RTP session periodically transmits RTCP control packets to all other participants. 32 bits each The CSRC list identifies the contributing sources for the payload contained in this packet. using the SSRC identifiers of contributing sources.3 Reducing Header Overhead with CRTP RTCP is the control protocol that works in conjunction with RTP. Each RTCP packet contains sender and/or receiver reports that report statistics useful to the application. If there are more than 15 contributing sources.[Integrated VOIP System] The SSRC field identifies the synchronization source. 2. These statistics include number of packets sent. 2. RTCP performs the following four functions. Provide information to application: The primary function is to provide information to an application regarding the quality of data distribution.5. allowing correct talker indication at the receiver. This identifier is chosen randomly.

[Integrated VOIP System]
ber of packets lost, inter-arrival jitter, etc. This reception quality feedback will be useful for the sender, receivers, and third-party monitors. For example, the sender may modify its transmissions based on the feedback; receivers can determine whether problems are local, regional or global; network managers may use information in the RTCP packets to evaluate the performance of their networks for multicast distribution. Identify RTP source: RTCP carries a transport-level identifier for an RTP source, called the canonical name (CNAME). This CNAME is used to keep track of the participants in an RTP session. Receivers use the CNAME to associate multiple data streams from a given participant in a set of related RTP sessions, e.g., to synchronize audio and video. Control RTCP transmission interval: To prevent control traffic from overwhelming network resources and to allow RTP to scale up to a large number of session participants, control traffic is limited to at most 5 percent of the overall session traffic. This limit is enforced by adjusting the rate at which RTCP packets are transmitted as a function of the number of participants. Since each participant sends control packets to everyone else, each can keep track of the total number of participants and use this number to calculate the rate at which to send RTCP packets. Convey minimal session control information: As an optional function, RTCP can be used as a convenient method for conveying a minimal amount of information to all session participants. For example, RTCP might carry a personal name to identify a participant on the user‘s display. This function might be useful in loosely controlled sessions where participants informally enter and leave the session. 2.4 RTCP Packet Structure Each RTCP packet begins with a fixed part similar to that of RTP data packets, followed by structured elements that may be of variable length according to the packet type but always end on a 32-bit boundary. Each individual RTCP packet in the compound packet may be processed independently with no requirements upon the order or combination of packets. However, in order to perform the functions of the protocol, the following constraints are imposed. Reception statistics (in SR or RR) should be sent as often as bandwidth constraints will allow to maximize the resolution of the statistics, therefore each periodically transmitted compound RTCP packet should include a report packet. New receivers need to receive the CNAME for a source as soon as possible to identify the source and to begin associating media for purposes such as lip-sync, so each compound RTCP packet should also include the SDES CNAME.

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[Integrated VOIP System]

2.4.1 RTCP Packet Types RFC 1889 defines several RTCP packet types to carry a variety of control information:

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RTCP Sender Reports (SR)

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[Integrated VOIP System] RTCP Receive Reports (RR) 36 .

2 Message Encapsulation 37 .4.[Integrated VOIP System] 2.

You can use CRTP headers on a link-by-link basis to save bandwidth. Given the number of protocols that are necessary to transport voice over an IP network. with compressed audio payloads between 20 and 50 bytes. as well as many other fields in all three headers.[Integrated VOIP System] This topic describes how IP voice headers are compressed using CRTP. Static fields include source and destination IP address. the packet header can be large. CRTP works on the premise that most of the fields in the IP/UDP/RTP header do not change. the CRTP process is illustrated in the following table: 38 . or that the change is predictable. RTP header compression is especially beneficial when the RTP payload size is small. For those fields where the change is predictable. for example. Using CRTP compresses the IP/UDP/RTP header from 40 bytes to 2 bytes without UDP checksums and from 40 bytes to 4 bytes with UDP checksums. In addition. source and destination UDP port numbers.

5 CRTP Packet Components In a packet voice environment when speech samples are framed every 20 ms.[Integrated VOIP System] 2. RTP header compression reduces the header to 2 bytes. In the figure. The compressed header is one tenth of the payload size. Without CRTP. a payload of 20 bytes is generated.6 When to Use RTP Header Compression You must configure CRTP on a specific serial interface or sub-interface if you have any of these conditions: Narrowband links Slow links (less than 2 Mbps) Need to conserve bandwidth on a WAN interface Compression works on a link-by-link basis and must be enabled for each link that fits these requirements. You must enable compression on both sides of the link for proper results. Enabling compression on both 39 . the total packet size includes the following components: IP header (20 bytes) UDP header (8 bytes) RTP header (12 bytes) Payload (20 bytes) The header is twice the size of the payload: IP/UDP/RTP (20 + 8 + 12 = 40 bytes) versus payload (20 bytes). 2. the header consumes a large portion of bandwidth. When generating packets every 20 ms on a slow link.

you can change the number of header compression connections with the ip rtp compression-connections number command. D epending on the traffic on the interface. However. the software supports a total of 16 RTP header compression connections on an interface. use the frame-relay ip rtp header-compression command. use the ip rtp header-compression command.[Integrated VOIP System] ends of a low-bandwidth serial link can greatly reduce the network overhead if there is a significant volume of RTP traffic on that slow link. By default. Example: Applying CRTP If you want the router to compress RTP packets. Note Do not use CRTP if you have high-speed interfaces or links faster than 2 Mbps. this command provides a passive mode setting in instances where you want the router to compress RTP packets only if it has received compressed RTP on that interface. When applying to a Frame Relay interface. The ip rtp header-compression command defaults to active mode when it is configured. 40 .

[Integrated VOIP System] Chapter 3 Sip Protocol 41 .

and multimedia distribution.1 Overview SIP provides another framework for establishing and maintaining VoIP calls. SIP multimedia sessions include Internet telephone calls. DNS for service location. SIP operates on the principle of session invitations. URLs for addressing. including the SAP defined in RFC 2974. which incorporates a session description according to the Session Description Protocol (SDP) defined in RFC 2327. maintenance. unicast. and Telephony Routing over IP (TRIP) for call routing. for example. and termination of multimedia sessions with one or more participants. SIP uses other IETF protocols to define other aspects of VoIP and multimedia sessions. Session communications may be based on multicast. 42 . SIP initiates sessions or invites participants into established sessions. or both. This topic describes SIP and its standards. multimedia conferences. SIP supports personal mobility and other Intelligent Network (IN) telephony subscriber services through name mapping and redirection services. Personal mobility allows a potential participant in a session to be identified by a unique personal number or name. Descriptions of these sessions are advertised by any one of several means. Through invitations.[Integrated VOIP System] 3. SIP is a signaling and control protocol for the establishment.

which renders obsolete RFC 2543 (March 1999).2 Components of SIP SIP is modeled on the interworking of user agents (UAs) and network servers. This topic describes the functional and physical components of a UA. SIP is described in IETF RFC 3261 (June 2002). 3. A UA consists of two functional components: 43 . Calling-card services. The peers in a session are called UAs. 800 services.[Integrated VOIP System] IN provides carriers with the ability to rapidly deploy new user services on platforms that are external to the switching fabric. Access to the external platforms is by way of an independent vendor and standard user interface. the primary motivation behind the protocol is to create an environment that supports nextgeneration communication models that use the Internet and Internet applications. SIP is a peer-to-peer protocol. Multimedia sessions are established and terminated by the following services:  User location services: Locate an end system  User capabilities services: Select the media type and parameters  User availability services: Determine the availability and desire for a party to participate  Call setup services: Establish a session relationship between parties and manage call progress  Call handling services: Transfer and terminate calls Although the IETF has made great progress in defining extensions that allow SIP to work with legacy voice networks. and local number portability are just three of these services.

Proxy servers can provide functions such as authentication. From an architectural standpoint. Whether the endpoint functions as a UAC or a UAS depends on the UA that initiated the request: the initiating UA uses a UAC and the terminating UA uses a UAS. network access control. Location server: An abstraction of a service providing address resolution services to SIP proxy or redirect servers. reliable request transmissions. Redirect server: Provides a UA with information about the next server that the UA should contact. SIP servers: SIP servers include the following types:  Proxy server: Intermediate component that receives SIP requests from a client. Gateway: Acts as a UAS or UAC and provides call control support. rwhois. Lightweight Directory Access Protocol (LDAP). A location server embodies mechanisms to resolve addresses. a SIP UA can function as a UAC or a UAS during a session. This function includes translation between transmission formats and between communications procedures. then forwards the requests on behalf of the client to the next SIP server in the network. authorization. Software telephones and Cisco Systems SIP IP Phones initiate SIP requests and respond to requests. Gateways provide many services. the physical components of a SIP network are grouped into the following two categories: User agents: SIP user agents include the following devices:   IP telephone: Acts as a UAS or UAC on a session-by-session basis. most often a location server. but not both in the same session. the registrar server is partly responsible for populating a database associated with the location server. A registrar server can be modeled as one subcomponent of a location server.    Example: SIP Applications 44 . These mechanisms can include a database of registrations or access to commonly used resolution tools such as finger.[Integrated VOIP System]   User agent client (UAC): A client application that initiates a SIP request. the most common being a translation function between SIP user agents and other terminal types. or operating-systemdependent mechanisms. User agent server (UAS): A server application that contacts the user when a SIP invitation is received and then returns a response on behalf of the user to the invitation originator. routing. The UA redirects the invitation to the server identified by the redirect server. Registrar servers are often located near or even collocated with other network servers. Registrar server: Requests from UACs for registration of their current location. A gateway translates between audio and video signals and performs call setup and clearing on both the IP side and the SCN side. Typically. and security. The server can be another network server or a UA. The next server can be another proxy server or a UAS.

and a message body Response from a server to a client: Consists of a status line. Cisco is enabling the advance of new communications services with a complete SIP-enabled portfolio. call control and signaling. and carriers that have deployed SIP in their networks include interexchange carriers such as WorldCom and Genuity.323. and they are offering attractive new communications services to their customers.3 SIP Messages Communication between SIP components uses a request and response message model. MGCP. header lines. Only Cisco is dedicated to providing\ ubiquitous and seamless protocol interoperability in its packet-telephony solutions. packet voice gateways. and a message body 45 . SIP communication involves the following two messages:   Request from a client to a server: Consists of a request line. SIP is gaining momentum in every market. IP Phones. including H. alternate carriers such as Vonage. Internet telephony service providers such as delta three. This topic describes the types. which can coexist in the same customer network. These products are available today. 3. and advanced service providers such as Microsoft all Cisco customers. and firewalls. Cisco solutions support a variety of call control and standard protocols. including proxy servers.[Integrated VOIP System] Leaders in the communications industry are developing new products and services that rely on SIP. header lines. Microsoft recently added support for SIP clients in core product offerings Windows XP and Windows Messenger a step that will proliferate SIP clients on personal computers worldwide. and structure of these messages. and SIP. use. telephony application service providers and communications service providers such as Talking Nets and Tell Me.

method. and a response header. an entity header.1.[Integrated VOIP System] All SIP messages are text-based and modeled on RFC 822. BYE: A client or server originates the BYE method to initiate call termination. The first two appear on both message types. ACK: A client originates the ACK method to indicate that the client has received a response to its earlier invitation. to indicate the action to be taken by the responding component (usually a server). a request header. Hypertext Transfer Protocol . In the request line. and RFC 2068. HTTP/1. SIP defines four types of headers: a general header. CANCEL is not used to terminate active sessions. CANCEL: A client or server originates the CANCEL method to interrupt any requests currently in progress. The following request methods indicate the action that the responding component should take:     INVITE: A client originates the INVITE method to indicate that the server is invited to participate in a session. 46 . respectively. Standard for ARPA Internet Text Messages. The latter two are specific to request and response. SIP uses a . An invitation includes a description of the session parameters.

A status code reflects the outcome of the request. a redirect server responds with . REGISTER: A UA uses the REGISTER method to provide information to a network server. Registrations have a finite life and must be renewed periodically.[Integrated VOIP System]  OPTIONS: A client uses the OPTIONS method to solicit capabilities information from a server. Indicates that the request is still being processed. failure.  SIP response messages are sent in response to a request and indicate the outcome of request interpretation and execution. 47 . 3xx. to advise the client to redirect its invitation.Informational: Provisional response. Responses take one of three basic positions: success. 2xx. 3. for example. This method is used to confirm cached information about a UA or to check the ability of a UA to accept an incoming call. This prevents the use of stale information when a UA moves.moved.Redirection: Indicates that the requestor requires further action.Successful: Indicates that the requested action is complete and successful. or provisional.4 Status Codes The following response messages indicate the status of a request:    1xx.

3. a host description. the final recipient. address registration. the current destination and final recipient URLs are the same.Global failure: Fatal response. 5xx. Indicates that the request cannot be fulfilled by any server. The host description may be a domain name or an IP address. and address resolution. This topic describes address formats. and any contact party. Indicates that the request is valid but the server failed to complete it. 6xx. and a port number is associated with the host description. Example: SIP Addressing Variants 48 . An address in SIP is defined in the syntax for a URL with sip: or sips: (for secure SIP connections) as the URL type. However. and optional parameters to qualify the a ddress more precisely. SIP URLs are used in SIP messages to identify the originator. the current destination and the final recipient are different if a proxy or redirect server is used.5 SIP Addressing To obtain the IP address of a SIP UAS or a network server. An address consists of an optional user ID.Client error: Fatal response.Server error: Fatal response. the current destination. a UAC performs address resolution of a user identifier.[Integrated VOIP System]    4xx. A password is associated with the user ID. When two UAs communicate directly with each other. Indicates that the client request is flawed or impossible to complete.

For a network server to assist. it must recognize the endpoints in the network. In the example sip:14085551234@gateway. the password changed is defined for the user. A SIP address is acquired in several ways: by interacting with a user.1. the user ID is taken literally as a numeric string.com. by caching information from an earlier session. 49 .1 is an example of a numeric user ID.1.[Integrated VOIP System] The figure provides examples of SIP addresses. or by interacting with a network server. user=phone. This knowledge is abstracted to reside in a location server and is dynamically acquired by its registrar server. In the same example. The 14085559876 in the URL sip:14085559876@10. the user=phone parameter is required to indicate that the user part of the address is a telephone number. Without the user=phone parameter.

or it leaves that responsibility to a network server. 3. The figure illustrates a SIP proxy server resolving the address by using the services of a location server. the client solicits the assistance of a network server. A network server uses any of the tools available to a UA or interacts through a nonstandard interface with a location server. a UA uses a variety of internal mechanisms such as a local host table. an endpoint registers its user addresses with a registrar server. To resolve an address. or LDAP. In situations in which the client is unable to establish a direct relationship. and u sing a redirect server.[Integrated VOIP System] To contribute to this dynamic knowledge. rwhois. finger.6 Call Setup Models If a UAC recognizes the destination UAS. the client communicates directly with the server. 50 . This topic illustrates three interworking models for call setup: direct. using a proxy server. The figure illustrates a REGISTER mode request to a registrar server. DNS lookup.

Direct setup is the fastest and most efficient of the call setup procedures. 51 . The originating UAC issues an ACK. if the UA must keep information on a large number of destinations. the UAC may initiate direct (UAC-to. management of the data can become prohibitive. It relies on cached information or internal mechanisms to resolve addresses. or has the capacity to resolve it by some internal mechanism.[Integrated VOIP System] When a UA recognizes the address of a terminating endpoint from cached information. which can become outdated if the destination is mobile. the UAC and UAS have all the information that is required to establish RTP sessions between them. direct setup has some disadvantages. The message i ncludes an endpoint description of the UAC and SDP.UAS) call setup procedures. In addition. it responds positively to the originator UAC. At this point. This makes the direct method nonscalable. However. If the UAS of the recipient determines that the call parameters are acceptable. Direct call setup proceeds as follows: The originating UAC sends an invitation (INVITE) to the UAS of the recipient.

52 . The proxy server. If the UAS of the recipient determines that the call parameters are acceptable. AU noitanitsed gnihsilbatse fo elbapacni si AU eht . the call setup procedure is as follows: The originating UAC sends an invitation (INVITE) to the proxy server. The proxy server forwards the ACK to the recipient UAS. The UAC and UAS now have all the information required to establish RTP sessions. ytilibapac noitulos er sserdda etad -ot -pu dna cimanyd erom a gnidivorp dna putes eht htiw etacinummoc llits nac tey . AU noitanitsed eht fo setanidrooc eht nrael ot deen ton seod ti taht gnigassem erom seriuqer revres yxorp a gnisu taht era dohtem siht fo segatnavdasid ehT . if required. llac fo tnemeganam dna lortnoc gnizilartnec yb dohtem tcerid eht fo seussi eht ot sdnopser revres yxorp A si AU eht ot tifeneb ehT . consults the location server to determine the path to the recipient and its IP address. The proxy server sends the invitation to the UAS of the recipient. The proxy server responds to the originating UAC. The proxy server intercepts and forwards an invitation to the destination UA on behalf of the originator. sliaf revres yxorp eht fI . The originating UAC issues an ACK.[Integrated VOIP System] The proxy server procedure is transparent to a UA. snoisses nwo sti When a proxy server is used. revres yxorp eht no ycnedneped a setaerc d na . it responds positively to the proxy server.

the redirect server reports back to a UA with the destination coordinates that the UA should try next.moved. the call setup procedure is as follows: 1. However. The originating UAC issues an acknowledgment. 3. 4. 6. consults the location server to determine the path to the recipient and its IP address. 2. 5. When a redirect server is used. The redirect server. it responds positively to the UAC. The originating UAC sends an invitation (INVITE) to the redirect server.[Integrated VOIP System] A redirect server is programmed to discover a path to the destination. The redirect server returns a . If the UAS of the recipient determines that the call parameters are acceptable. if required. response to the originating UAC with the IP address obtained from the location server. A redirect server offers many of the advantages of the proxy server. The UA has a heavier workload because it must initiate the subsequent invitation. The UAC and UAS now have all the information required to establish RTP sessions between them. 7. the number of messages involved in redirection is fewer than with the proxy server procedure. Instead of forwarding the invitation to the destination. 53 . The originating UAC sends an invitation to the remote UAS. The originating UAC acknowledges the redirection.

[Integrated VOIP System] Chapter 4 Networking basics 54 .

it merely displayed data on the monitor and processed keystrokes to send back to the mainframe. This terminal. it became necessary to link these networks together across floors. 55 . located in a remote location. he or she would sit at a terminal that was connected to the mainframe by some type of cabling. PCs could connect to the mainframe in place of the dumb terminals. For this reason. The terminal performed very little work on its own. In fact. A WAN is a means of connecting LANs together across a distance boundary. or WAN. Perhaps the most well-known network is the Internet. Typical WAN connectivity was accomplished through phone lines. the paradigm began to shift. As PCs began to work in conjunction with the mainframe. It therefore makes sense to use the divide-and conquer strategy.1 Introduction to Networking In the early days of computing. Local Area Networks. was the gateway to the processing power of the mainframe. These computers were large and centrally located. LANs were implemented in a business using technologies such as Ethernet and Token Ring to connect computers together using Network Interface Cards. The OSI model is made up of seven functional layers. thereby slowing productivity. However. hence. Today. LANs.2 The OSI Model A network protocol is an agreed-upon format for transmitting data between two devices. Although processing was performed on the mainframe. data communication systems are so complex that it is very difficult to use a single protocol to handle all transmission tasks and problems. and various combinations of the two. and the increasing importance of the home and office computer was realized. i. taught as the foundation for this knowledge. there were mainframe computers.e. or LANs. but more importantly. computers throughout the world are connected through WANs. with each layer providing a distinct functionality. became the term used to describe the way in which computers were connected together to share data. LAN connectivity became a new industry market. these terminals were often called dumb terminals. cities. When the personal computer (PC) became a reality in the late 1980s. The PC revolution began. the average user did not walk up to it and start an application. more and more users were connecting to the mainframe computer through the terminals. new technology was required to efficiently connect them. the International Standard Organization (ISO) created the Open System Interconnect (OSI) reference model.[Integrated VOIP System] 4. The means of connecting all these networks together to achieve a desired goal is called internetworking. The mainframe computers were continually being enhanced and upgraded to keep up with the processing demand. As time progressed. Technology started producing smarter terminals to decrease the load on the mainframe. and even countries. or NICs. 4. buildings. usually in a very cold and climate-controlled environment. to split a problem into several smaller problems and resolve each of them individually. they could process data on their own. This increased the load on the mainframe. Table 1 summarizes the functionalities of the OSI layers. As more and more LANs became operational. ISO is an international organization composed of national standards bodies from over 75 countries. Instead. In 1983. and new businesses worldwide started operations. the introduction of the Wide Area Network.

This allows developers the freedom to choose the best method they can design. Notice that the bottom layer is identified as the first layer. Each layer thinks it is talking directly to the same layer on the remote computer (see Figure 1–2) through a virtual link. Doing so creates a better understanding of the network data flow. we must clarify one thing. but possibly the most common is Each layer is separated.[Integrated VOIP System] How Does a Model Work? Before defining how a model works. The OSI model defines what each layer should do—it does not tell you how to do it. Many acrostics can be used to remember the order. each layer can only communicate with the layers above and below it. or encapsulated. The OSI model is divided into seven layers. Furthermore. It is also needed to pass an exam. the layer doesn‘t know that 56 . This means that each layer can function on its own. In fact. Figure 1 lists the name and order of each layer. It is important to remember the order of the layers in the OSI model. from each other layer.

[Integrated VOIP System] any other layers even exist. 57 . flows down the layers. Finally. across the wire to the receiving computer. For example. and then back up the layers to the Application layer (see Figure 4). the flow of data starts at the Application layer of the sending computer. notice in Figure 1–3 that the Transport layer can communicate only with the Network and the Session layers.

[Integrated VOIP System] 58 .

e.e. Layer 3) in OSI. how does the data get to its destination correctly? 4. a question may arise: Since there may be more than one application using more than one communication partner using more than one protocol. From the above figure.[Integrated VOIP System] Each layer has a specific function for which it is responsible. Layers 1) in OSI.  The Network Interface layer in TCP/IP corresponds to the Data Link layer (i. Although the layers start at the bottom. It is named after the two most important protocols in the suite: the Transmission Control Protocol (TCP) and the Internet Protocol (IP). Presentation and Session layers (i.3 The TCP/IP Protocol Suite The TCP/IP protocol suite. Layer 4) in OSI. relying on lower layer protocols to translate data into forms that are transmitted physically over the network. is the set of communications protocols that implements the protocol stack on which the Internet and most commercial networks run.  The Transport layer in TCP/IP corresponds to the Transport layer (i. Layers 7.e.  The Internet layer in TCP/IP corresponds to the Network layer (i. Data Encapsulation Using the OSI Model As we read through the description of the OSI layers.  The Hardware layer in TCP/IP corresponds to the Physical layer (i. also referred to as the Internet protocol suite. 59 . Layers 2) in OSI. Upper layers are logically closer to the user and deal with more abstract data.e.e. The TCP/IP protocol suite—like the OSI reference model—is defined as a set of layers. 6 and 5 respectively) in OSI. we will examine the layers starting at the top. it is clear that:  The Application layer in TCP/IP roughly corresponds to the Application.

Some well known examples of application level entities within the TCP/IP domain are: • FTP/Telnet/SSH • HTTP/Secure HTTP (SHTTP) • POP3/SMTP • SNMP Transport Layer The transport layer of the TCP/IP model maps fairly closely to the transport layer of the OSI model. Also. Application Layer The application layer of the TCP/IP model corresponds to the application layer of the OSI reference model. what they mean is a protocol that functions at the Network layer in OSI. So when they talk about a ―Layer 3 protocol‖. Two commonly used transport layer entities are TCP and User Datagram Protocol (UDP) Internet Layer 60 .[Integrated VOIP System] Some people combine the Network Interface and Hardware layers when talking about TCP/IP. be aware that networking people tend to use terms like Layer 2 and Layer 3 instead of Data Link layer and Network layer respectively. and therefore TCP/IP has only four layers in that case.

PPP over Ethernet (PPPoE) • ATM/Frame Relay 61 . The network access layer contains two sub layers. The primary component of the Internet layer is the Internet Protocol (IP).Many of the TCP/IP routing protocols are also classified as part of the Internet layer. The physical sub layer aligns with the physical layer of the OSI model. Network Access Layer The lowest layer of the TCP/IP protocol stack is the network access layer.[Integrated VOIP System] The Internet layer of the TCP/IP model maps to the network layer of the OSI model. and is sometimes referred to by that name. Consequently. the Internet layer is sometimes referred to as the network layer. Examples of the network access layer include: • Ethernet • Wireless Fidelity (Wi-FI)/WiMAX • PPP. Note: Some references divide the TCP/IP model into 5 layers. The MAC sub layer aligns closely with the data link layer of the OSI model. the media access control (MAC) sub layer and the physical sub layer. with the MAC and physical layers occupying the lowest two layers.

These addresses are used by intermediate routers to select a path through the network for the packet. into smaller packets. Packet timeouts. IP supports traffic prioritization by allowing packets to be labeled with an abstract type of service. IP packets may be split. Fragmentation. preventing packets from running in circles forever and flooding a network. IHL: 4 bits Internet Header Length is the length of the internet header in 32 bit words.[Integrated VOIP System] 4. Also. and thus points to the beginning of the data. This course describes version 4. Each IP packet contains a Time to Live (TTL) field. trace the route a packet takes (record route). This means each packet must contain complete addressing information. only the IP header. which is decremented every time a router handles the packet. allowing a packet's sender to set requirements on the path it takes through the network (source routing). Options. the packet is discarded. 62 . IP fragments and reassembles packets transparently.4 The Internet Protocol (IP) IP is a datagram-oriented protocol. IP makes no attempt to determine if packets reach their destination or to take corrective action if they do not. and label packets with security features. IP provides several optional features. Type of Service. Note that the minimum value for a correct header is 5. Version: 4 bits The Version field indicates the format of the internet header. If TTL reaches zero. This permits a large packet to travel across a network which can only handle smaller packets. IP headers contain 32-bit addresses which identify the sending and receiving hosts. or fragmented. treating each packet independently. The form of the packets is called an IP datagram as shown in the following figure. Nor does IP checksum the contents of a packet.     IP defines the format of the packets and how to handle them when sending or receiving. IP provides several services:  Addressing.

Options: variable 63 . The fragment offset is measured in units of 8 octets (64 bits). The values for various protocols are specified in "Assigned Numbers". but since every module that processes a datagram must decrease the TTL by at least one even if it process the datagram in less than a second Protocol: 8 bits This field indicates the next level protocol used in the data portion of the internet datagram. These parameters are to be used to guide the selection of the actual service parameters when transmitting a datagram through a particular network. Such long datagrams are impractical for most hosts and networks. Time to Live: 8 bits This field indicates the maximum time the datagram is allowed to remain in the internet system. this is recomputed and verified at each point that the internet header is processed. The first fragment has offset zero. Header Checksum: 16 bits A checksum on the header only. Total Length: 16 bits Total Length is the length of the datagram. Identification: 16 bits An identifying value assigned by the sender to aid in assembling the fragments of a datagram. This field allows the length of a datagram to be up to 65.. including internet header and data. The time is measured in units of seconds. Destination Address: 32 bits The destination address. then the datagram must be destroyed. All hosts must be prepared to accept datagrams of up to 576 octets (whether they arrive whole or in fragments). Flags: 3 bits Various Control Flags Fragment Offset: 13 bits This field indicates where in the datagram this fragment belongs. Source Address: 32 bits The source address. This field is modified in internet header processing.[Integrated VOIP System] Type of Service: 8 bits The Type of Service provides an indication of the abstract parameters of the quality of service desired.535 octets. If this field contains the value zero. time to live).g. measured in octets. Since some header fields change (e. It is recommended that hosts only send datagrams larger than 576 octets if they have assurance that the destination is prepared to accept the larger datagrams.

not their implementation. What is optional is their transmission in any particular datagram. Each computers Ethernet address is unique. 127.6. For sites connected to the Internet (a global computer network of universities. 4. it first translates the destination IP address into a physical address in order to send packets to other computers on the network (this is called address resolution). The construction of an IP address is divided into three classes. 080BF0AFDC09). which takes different forms on different networks. where each byte represents a value between 0 and 255. or accessing machines remotely. and are independent of any particular hardware or network component. This logical numbering is important in sending information to other users at other networks. It consists of a 4 byte (32-bit) numeric value which identifies the network number and the device number on the network. and are independent of any particular hardware or network component.g. e. For ETHERNET networks.11 When a computer wants to exchange data with another computer using TCP/IP. In some environments the security option may be required in all datagrams. and corresponds to the a ddress of the physical network card installed in the computer. Internet addresses are logical addresses.46.5 IP Addresses Each networked computer is assigned a physical address. Class A Addressing       first byte specifies the network portion remaining bytes specify the host portion the highest order bit of the network byte is always 0 network values of 0 and 127 are reserved there are 126 class A networks there are more than 16 million host values for each class A network 64 .g. companies and US defense sites). The IP logical numbering is comprised of a network number and a local number. the physical address is a 6 byte numeric (or 12 digit hexadecimal) value (e. Each node or computer using TCP/IP within the organization MUST HAVE a unique host part of the IP address. Internet addresses are logical addresses. which a machine identifies itself as. Which class is used by an organization depends upon the maximum number of work stations that is required by that organization. The TCP/IP protocol implements a logical network numbering.[Integrated VOIP System] The options may appear or not in datagrams. The 4 byte IP a ddress is represented in dotted decimal notation. stored in configuration files. the network portion is assigned by applying to a company responsible for maintaining the Internet Domain Names. databases. They must be implemented by all IP modules (host and gateways)..

[Integrated VOIP System]

Class B Addressing
    

the first two bytes specify the network portion the last two bytes specify the host portion the highest order bits 6 and 7 of the network portion are 10 there are more than 16 thousand class B networks there are 65 thousand nodes in each class B network

Class C Addressing
    

the first three bytes specify the network portion the last byte specifies the host portion the highest order bits 5, 6 and 7 of the network portion are 110 there are more than 2 million class C networks there are 254 nodes in each class C network

Reserved IP Addresses The following IP addresses are reserved for special purposes, and must NOT be assigned to any host.
  

Network Addresses : The host portion is set to all zero's (129.47.0.0) Broadcast Address : The host portion is set to all one's (129.47.255.255) Loop back Addresses : 127.0.0.0 and 127.0.0.1

Private IP Addresses

65

[Integrated VOIP System]
Network hosts that do not need to have their addresses visible on the public Internet can be a ssigned private IP address. There are four address ranges that are reserved for private networking use only. These addresses can not be used to route data outside of a private network. The following are the four IP address ranges reserved for private networks:     • 10.0.0.0—10.255.255.255 • 172.16.0.0—172.31.255.255 • 192.168.0.0—192.168.255.255 • 169.254.0.0—169.254.255.255

4.6 Transport Layer protocols Within the TCP/IP protocol suite, the two most common transport layer entities are the UDP and the TCP. 4.6.1 User Datagram Protocol (UDP) The User Datagram Protocol is very simple. The PDU used by UDP is called a datagram. Datagrams are considered unreliable, in that there is no guarantee datagrams will be received in the correct order, if at all. If reliability of the information transmitted is needed, UDP should not be used. While UDP is unreliable, the lack of error checking and correction make UDP fast and efficient for many less data intensive or time sensitive applications, such as the Domain Name Service (DNS), the Simple Network Management Protocol (SNMP), and the Dynamic Host Configuration Protocol (DHCP) and the Routing Information Protocol (RIP). UDP is also well suited for streaming video. Basic Protocol Operation The UPD protocol is simple in operation. When invoked by the application layer, the UDP protocol performs the following operations: 1. Encapsulates the user data into UDP datagrams 2. Passes the datagram to the IP layer for transmission At the opposite end, the UDP datagram is passed up to UDP from the IP layer. UDP then removes the user data from the datagram and presents it upward to the application layer. Ports A port is a number that identifies the application using the UDP service. It can be thought of as an address for applications. For example, the application level protocols used for e-mail, POP3 and SMTP, are assigned standard port numbers. The port number is used by the UDP client on the receiving end to know what application to pass user data to. The UDP Packet Structure The UDP packet structure is illustrated in Figure. It consists of 5 fields, some of which are optional:

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[Integrated VOIP System]
     Source Port—the sending application. This is an optional field. Destination Port—the target application at the receiving end. Length—the length of the entire packet. Checksum—Optional field used to perform basic error correction on the packet. Data—the user data to be transmitted.

4.6.2 Transmission Control Protocol (TCP) In the TCP/IP protocol suite, TCP is the intermediate layer between IP below it, and an application above it. Using TCP, applications on networked hosts can establish reliable connections to one another. The pr otocol guarantees in-order delivery of data from the sender to the receiver. Basic Protocol Operation The Transmission Control Protocol is connection-oriented, meaning user data is not exchanged between TCP peers until a connection is established between the two end points. This connection exists for the duration of the data transmission. TCP connections have three phases: 1. Connection establishment 2. Data transfer 3. Connection termination Connection Establishment To establish a connection, TCP uses a 3-way handshake. Before a client attempts to connect with a server, the server must first bind to a port to open it up for connections. This is called a passive open. Once the passive open is established, a client may initiate an active open. The server then sends an acknowledgement to the client. At this point, both the client and server have received an acknowledgement of the connection. Data Transfer A few key features set TCP apart from UDP:  Error-free data transfer  Ordered-data transfer  Retransmission of lost packets  Discarding of duplicate packets 67

Retransmission of Lost Packets When transmitting large amounts of data. These units are encapsulated in the TCP packet that is passed to the IP protocol. with each side of the connection terminating independently. so TCP breaks up the segments into smaller units of data. In order to guarantee reliable transfer of data. Discarding Duplicate Packets The TCP client retransmits packets that it determines to be lost. When an end point wishes to stop its half of the connection. until packets begin to become unacknowledged during the time out period. if the checksum does not match the contents of the packet. The goal is for TCP to be able to send data to the receiving end at the fastest rate possible. This may result in the receiving end receiving two or more copies of the same packet. Because the sending side does not receive an acknowledgement of the discarded packet. 68 . TCP requires an acknowledgement of each packet it sends. at most. the TCP module uses the sequence numbers in the packet to reconstruct the user data in the correct order. a four-way handshake. When a significant number of packets have to be retransmitted. TCP will incrementally increase the transmission speed. Connection Termination The connection termination phase uses. If an acknowledgement is not received within a specified time period. The TCP module at the receive side may eventually receive packets that were considered to be lost after the sending side has retransmitted the data. At the receiving end. It does this by calculating a 16-bit checksum over the TCP packet (header and data).[Integrated VOIP System]  Congestion throttling Error Free Data Transfer Error-free data transfer is guaranteed by TCP. it is not unusual for some information to get lost along the way. it is discarded. The other end acknowledges the flag. TCP slows down the rate at which it sends data to the other end. If all packets are received well before the timer expires. it sets a timer. The timer determines how long the sender should wait for a packet to be acknowledged before retransmitting it. which becomes part of the TCP packet. Ordered-Data Transfer Streams of data called segments are used by TCP peers to speak to each other. At the receiving end. A typical connection termination includes this two-phase handshake from both ends of the connection. it is retransmitted. The receiving end TCP module uses the unique sequence numbers in the packet to determine if data duplication has occurred and discards any packets it determines to be duplicates. it transmits a special packet with a flag indicating it is finished. Each unit of data is assigned a sequence number. This acknowledgement is sent by the TCP module at the receiving host. it will be retransmitted. without overwhelming it. The segments can be quite large. Congestion Throttling The final property of TCP is congestion throttling or flow control. When TCP first begins transmitting data to the far end.

RST—Reset the connection. From a logical standpoint.URG—Urgent pointer field is significant.      69 .PSH—Push function.  Options—Additional header fields (called options) may follow the urgent pointer.SYN—Synchronize sequence numbers. TCP Packet Structure A TCP packet consists of two sections.  Data—the contents of this field are the user data being transmitted between two application level entities. then this 16-bit field is an offset from the sequence number indicating the last urgent data byte. should be set to zero. . The end points of the connection between TCP peers are called sockets. In reality. it is also the offset from the start of the TCP packet to the data portion.  Reserved—Reserved for future use. . . Sequence number—Used for assembling segmented data in the proper order at the receiving end. of which 10 are required: Source port—identifies the sending application. A socket is identified by a combination of the source host address and port together with the destination host address and port. reading and writing packets to a socket is how TCP interfaces with the IP layer below it.  Flags (also known as control bits)—contains 6 1-bit flags: .FIN—No more data from sender.ACK—Acknowledgement field is significant. . Acknowledgement number—the sequence number the sender (the receiving end) expects next.  Window—the number of bytes the sender is willing to receive starting from the acknowledgement field value. . Arriving T CP data packets are identified as belonging to a specific TCP connection by its socket. A virtual connection is first created then maintained through the duration of data transfer. TCP peers communicate directly with each other over the socket connection. All fields may not be used in every transmi ssion. header and data. Data offset—the size of the TCP header.[Integrated VOIP System] TCP Sockets Transmission Control Protocol is connection-oriented.  Urgent pointer—if the URG flag is set. A flag field is used to indicate the type of transmission the packet represents and how the packet should be interpreted.  Checksum—used for error-checking of the header and data. The header consists of 11 fields. Destination port—identifies the destination application.

particularly if it operates above the physical sub layer. The remaining portion is assigned by the manufacturer. network attached storage devices. CSMA/CD makes the following assumptions: 70 . This includes computers. and so forth. A scheme known as Carrier Sense Multiple Access with Collision Detection (CSMA/CD) was developed to govern the way the computers on the network shared the channel. A portion of the MAC address is a manufacturer identifier. as it is known. Ethernet peers communicate by exchanging frames. it defines: • A common addressing format • A means of accessing and sharing the network media • Several wiring and signaling standards Ethernet at the MAC Sub layer At the MAC sub layer.7 Ethernet Ethernet is a family of network access layer frame-based computer networking technologies. any device directly connected to an Ethernet LAN must have a MAC address. assigned by the IEE E registration authority.[Integrated VOIP System] 4. network printers. Ethernet frames are transmitted in the network using globally unique 48-bit physical layer addresses. As a network access layer standard. Generally speaking. Accessing the Physical Medium Originally. which encapsulate the Internet layer datagram. Ethernet used a single shared coax attached to every device on a network. typically comes programmed into an Ethernet device by the manufacturer. routers. The MAC address.

Retry the main procedure. 2. the Digital Equi pment Corporation (DEC). abort the transmission. this committee published the portion of the standard pertaining to Ethernet (based on the DIX standard)—IEEE 802. and Xerox (DIX) released the DIX Ethernet standard. The Ethernet device starts transmission. In 1980. and IEEE 802. 4. it waits for a period called the inter-frame gap. Continue transmission until minimum packet time is reached (jam signal) to ensure that all receivers detect the collision. the name had caught on. the Institute of Electrical and Electronics Engineers (IEEE) commissioned a committee to develop open network standards. 4. Even though the IEEE title does not mention Ethernet. requiring a retransmission of the collided frames.3 Carrier Sense Multiple Access with Collision Detection Access Method and Physical Layer Specifications. 2. Frame transmission ends successfully. both frames can coexist at the physical level. An Ethernet device will be able to sense if another device is using the medium by detecting a carrier signal. at the sub network level. Collision Detected Procedure 1. 4. and trailer. Modern Ethernet networks still use CSMA/CD to govern access to the transmission medium. it uses the following algorithm.9 Ethernet Frames Ethernet frames consist of 3 portions: header. 4. Two Ethernet devices may inadvertently use the medium at the same time. Calculate and wait a random back-off period. While the DIX frame is the most commonly deployed. 5.[Integrated VOIP System] 1. That same year.3 was and is referred to as the Ethernet standard. causing frame collisions. 3. There are very subtle differences between the DIX Ethernet frame format and the 802. The Ethernet device listens to see if any other device is using the channel. If the medium is not idle. Ethernet has a frame ready for transmission. Did a frame collision occur? If so. When one computer on an Ethernet LAN is required to transmit information. 2. An Ethernet device will be able to detect these frame collisions. invoke the collision procedure. Main Procedure 1. payload. 3. Intel. 71 . If the maximum number of transmission attempts has been reached.8 Ethernet Standards Ethernet was originally developed by the Xerox corporation in the 1970s. In 1985.3 frame format.

 Frame Check Sequence (FCS)—Detects transmission errors and provides quality of service at receiving end. Gigabit Ethernet can be transmitted over twisted pair cables or optical fiber. Data—Payload contained in a field between 46 bytes to just over 1500 bytes in length.3ae) is similar to that of lower speed Ethernet networks. It maintains the IEEE 802. and flow control methods. Ethernet Versions Ethernet has evolved significantly since the original thick-coax 10 Mb/s network of the 1980s. Destination Address—Destination MAC address.3u is commonly referred to as Fast Ethernet or 100Base-T. Pad—0 bits added to the data field if there are fewer than 46 bytes of data in that field.3 Ethernet frame size and format which preserves layer 3 and higher protocols. Additionally.3 only). Gigabit Ethernet Gigabit Ethernet works much the same way as 10 Mb/s and100 Mb/s Ethernet. Payload Logical Link Control (LLC)—governs the assembly of data at the data link (Layer 2) level. full duplex. Fast Ethernet IEEE 802. Type—indicates the protocol sending the frame (DIX only). (10Mb/s Ethernet). Because Fast Ethernet offers a choice of 10 or 100 Mb/s bandwidth. and it supports Simple Network Management Protocol (SNMP) tools. it takes advantage of CSMA/CD when in half-duplex mode.[Integrated VOIP System] Header           Preamble—7-bit sequence used for synchronizing receiver to transmitter. Jumbo frames are between 64 and 9215 bytes. Fast Ethernet can be cabled over either twisted-pair or fiber optic cables. 72 . 10 Gigabit Ethernet The operation of 10 Gigabit Ethernet (IEE 802. Start Frame Delimiter—8-bit sequence (10101011). It uses the same frame format. Several different versions of the Ethernet standard are in use today. Gigabit Ethernet takes advantage of jumbo frames to reduce the frame rate to the end host. Length—Indicates the length of data field (number of LLC data bytes) (IEEE 802. Source Address—Source MAC address. Trailer  Cyclical Redundancy Check (CRC)—Detects DIXonly transmission errors. only faster. the standard also allows for equipment that can auto negotiate the two speeds.

Unlike hubs.10 Switches Switches operate at Layer 2 (Data Link Layer) of the OSI model.[Integrated VOIP System] However. The MAC address table on the switch in Figure 2 may look like this: 73 . Additionally. eliminating the need for CSMA/CD. 10 Gigabit Ethernet only operates over point-to-point links in full-duplex mode. the switch record the source MAC address and port number in a table called the MAC address table. Figure 2 shows three PCs connected to a switch. it uses only fiber optical cable at the physical layer. and forwarding the frame to the appropriate switch port. network switches conserve network bandwidth and offer generally better performance than hubs. determining the source and destination MAC addresses of that frame. 4. By delivering data frames only to the connected device that it was intended for. switches are capable of i nspecting the data frames as they are received. When the PCs transmit data frames.

Therefore the fewer the frames a host needs to check. Compared to hubs. However ―manageable switches‖ or ―managed switches‖ do understand IP addresses but only for management purposes. When a new host is plugged into the switch. for PC B). 4. it looks up the MAC address table and finds that the host with the MAC address bb-bb-bb-bb-bb-bb (i. If a frame is sent to this new host before its MAC address and port number are recorded in the MAC address table. instead of copying the frame to all ports. Switches in general do not understand IP addresses because they are Layer 2 devices. and therefore forwards the frame only to port 3.[Integrated VOIP System] When the switch receives a frame with the destination MAC address bb-bb-bb-bb-bb-bb (i. however. The end result is that the hosts have better performance because they receive and therefore need to check fewer frames on average.e.11 Routers Routers operate at Layer 3 (Network Layer) of the OSI model.e. The router in Figure 3 has three network interfaces: two serial and one Ethernet. the switch will copy the frame to all ports except the port through which the frame entered the switch. When the new host replies. hosts connected to switches receive less unnecessary frames. its MAC address is not in the MAC address table until it transmits the first frame. A router must examine each packet‘s Layer 3 header before making a routing decision.  Serial Interface 1 is connected to the Internet  Serial Interface 2 is connected to a trusted partner‘s network. PC B) is located on port 3.  Ethernet Interface 1 is connected to the company‘s LAN. Figure 3 shows an example in which a router connects a co mpany‘s LAN to the Internet and a trusted partner‘s network. its MAC address and port nu mber will be recorded in the MAC address table. The routing table on the router would look like this: 74 . and subsequent frames for the new host will only be copied to its switch port. Remember a host on a network needs to check every single frame to see if the frame is for it. the less busy it is.

0 network. Similarly when the router receives a packet for the 10. 75 .0 and 10.0.0. it would route the packet to the trusted partner‘s network via Serial Interface 2.168.0.0 network.0.[Integrated VOIP System] When the router receives a packet for the 192.0.0. Layer 3 routing is generally performed by microprocessor-based engines. This can become a bottleneck due to a latency of packet examination and processing.0. When the router receives a packet for a network other than 192. it would route the packet to the company‘s LAN via Ethernet Interface 1. which require CPU cycles to examine each packet‘s Layer 3 header.168. it would route the packet to the Internet via Serial Interface 1.

[Integrated VOIP System] Chapter 5 Voip System Requirements 76 .

electronic private automatic branch exchange. PBXs are also referred to as:  PABX . whereas PABXs are modern systems that do so automatically. Technically. how to answer and the protocols for using it. some kind of display to show nu mbers dialed and caller ID.2 Objectives Upon completing this part. In the business environment most potential users even understand the common mechanisms for using most of the advanced features: voicemail. you will gain the power to construct and build a voip network that meets your needs and requirements. 5. conferencing. When it comes down to it. Any phone has to conform to these standards as well as provide any additional functionality that a VoIP system may provide. a PBX system uses a live operator to route calls. transferring calls. a speakerphone and mute capability and volume controls.4 PBX: PBX is nothing except a telephone exchange or telephone switch which is a system of electronic components that connects telephone calls.4.4.3 Terminals : By every measure the easiest of the components to understand and to evaluate. a phone is one of the most well understood pieces of technology in the world.  EPABX . etc. 5.1 overview This part describes the requirements and arranges that should be done to implement a voip practical system.[Integrated VOIP System] 5. System Components: 5.1 Overview: PBXs make connections among the internal telephones of a private organization usually a particular business or office and also connect them to the public switched telephone network (PSTN) via trunk lines. the primary advantage of PBXs was cost savings on internal phone calls: handling the circuit switching locally reduced charges for local phone service. 5. and to move from the theoretic part to the real part. Initially. hold. 5. how to dial. a handset to talk to and a way to plug it in. menu systems.2 PBX System Components: A PBX will often include: 77 . many VoIP systems make most of their functionality available through in-call menus that mean that a phone on the desk really just needs the standard 12 buttons on a dial pad. In fact. These should be regarded as a bare minimum for the desktop phone in any new phone system. But most phones offer a range of additional buttons to control options. Everyone knows the basics.private automatic branch exchange.

channeling voice signals between the users)  Disconnecting those connections as per the user's requirement  Providing information for accounting purposes (e.4. appeared in a hosted service before they became available in hardware PBX equipment Today. mapping a dialed number to a physical phone. or to contract with companies that provide less functionality for simple needs. metering calls) In addition to these basic functions.4.  Music on hold. 5.  Call forwarding on busy or absence. Stations or telephone sets. using equipment located in the premises of the telephone company's exchange. some PBX functions.  Welcome Message. Cabinets. In fact. it is possible to get hosted PBX service that includes far more features than were available from the first systems of this class. ensuring the phone isn't already busy)  Maintaining such connections as long as the users require them (i.  Call waiting. closets. sometimes called lines. power cards and related devices that facilitate PBX operation. vaults and other housings.  Speed Dialing.[Integrated VOIP System]          The PBX‘s internal switching network. Console or switchboard allows the operator to control incoming calls. Outside Telco trunks that deliver signals to (and carry them from) the PBX. This means the customer organization doesn't need to buy or install PBX equipment and the telephone company can use the same switching equipment to service multiple PBX hosting accounts.g. such as follow-me calling.4 Hosted PBX systems: A hosted PBX system delivers PBX functionality as a service. Logic cards.e. Microcontroller or microcomputer for arbitrary data processing. Uninterruptible Power Supply (UPS) consisting of sensors. switching and control cards.  Voice mail. The first hosted PBX service was very feature-rich compared to most premise-based systems of the time. with different manufacturers :  Auto dialing.3 PBX functions: Functionally. available over the Public Switched Telephone Network (PSTN) and/or the internet Hosted PBXs are typically provided by the telephone company. Interconnecting wiring. PBXs offer many other calling features and capabilities.  Conference call. power switches and batteries. users contract for PBX services from a hosted PBX service pr ovider. control and logic. 5. 78 .  Automatic ring back.g. the PBX performs four main call processing duties:  Establishing connections (circuits) between the telephone sets of two users (e. Instead of buying PBX equipment.

6 Voip Gateways: These devices connect VoIP networks to the PSTN system. A PBX (Private Branch Exchange) is the system which controls calls within your office. It has an RJ-11 port (for a regular phone line) into which users can plug an ordinary telephone handset. or other external controllers. Mobile Phone. performing the call-routing activities of the traditional PBX or key system as a software system.5 Service: The final piece of the puzzle is service. An IP PBX phone system is sometimes called a VoIP phone system and uses the same protocol an office computer network runs on. with functionality like voicemail. to control PBX phone functions and to manage communications without having to call into the system first. or IAX. 79 . keys and other input devices. The ATA converts the analog signals of the phone to digital data so it can go over the Internet. using a standard VoIP protocol such as SIP. PDA. The virtual version is also called a "Soft PBX". 5. PSTN Gateway. the services it provides and the support it gives you are critical to your new phone system. Mobile PBX systems are different from other hosted PBX systems that simply forward data or calls to mobile phones by allowing the mobile phone itself. No matter what you do. or it may be a third party. or can carry out its functions virtually. And good example for voip gateways. Gateways perform compression and decompression of voice transmissions. Microsoft Exchange.323. It also has an Ethernet port by which users can connect it to the local network. Another name for an ATA is VoIP router. auto attendants (virtual receptionist) and conference rooms.[Integrated VOIP System] 5. PSTN PCI Cards and SIP trunk providers. through the use of buttons. and can incorporate features such as network management and accounting. 5. It may also be the same company that runs your PBX or sold you your PBX. And that somebody is your service provider.4. VoIP gateways may interface with gatekeepers. as well as call routing. An ATA allows users to use a regular analog telephone to make VoIP calls.6 IP-PBX: An IP PBX handles voice signals under Internet protocol. smart phones and PDA phones by provisioning them as extensions. H. hunt groups. An IP-PBX can exist as physical hardware. 5. somebody has to provide broadband Internet access and termination of telephone calls into and out of the PSTN. bringing benefits for computer telephony integration (CTI). allowing calls between VoIP phones and PSTN Or mobile phones. The ATA communicates with the VoIP server.5 Mobile PBX: A mobile PBX is a hosted PBX service that extends fixed-line PBX functionality to mobile devices such as cellular handsets. but whichever it is.4. soft switches. The internet protocol (IP) transmits data between the PBX and phone or any other device such as a soft phone. Mobile PBX services also can include fixed-line phones.

8 System overall block diagram 80 . In a SIP/SDP . the server is a call agent.in a system based on H.323 .[Integrated VOIP System] 5. the server is a SIP server.the server is known as gatekeeper . 5. In a system based on MGCP or MEGACO .7 Server: The server provides management and administrative functions to support the routing of calls across the network .

[Integrated VOIP System] 5. The ring circuit must provide electrical isolation between telephone line and modem electronics.  Melody player and holding circuit. Over the years that PBX systems have been used.10. one resistor to limit the current passing through optoisolator LED and one reverse connected diode in parallel with optoisolator LED to prevent negative voltages from damaging the LED. This is the basic ring detection circuit.10. These pulses are counted by Timer1 in the micro controller. two bells have 37 pulses. Power Supply. Tone Generator. The system hold the (Outer) call after two bells. The optoisolator output can be easily connected to digital electronics. 5. This ring isolation is usually done using one optoisolator circuit. Ring Generator. DTMF Receiver.10.9 Analog PBX: Hardware analog PBX components:        COU.3 Hook Switch: 81 .1 Ring Detector: This stage converts the incoming bell signals into pulses. The cou includes many basic circuits for the telephone exchange system which are:  Ring detector circuit  Hook switch (dial pulse) circuit.2 Ring detector Circuit Schematic: 5. 5.10 COU(central office unit): Public telephone lines connected to traditional PBX systems are commonly referred to as "CO-lines". 5. Switching Unit. SLIC. but the optoisolator input side needs more electronics: one capacitor for not letting DC to pass through optoisolator.

10..10.then the melody is transmitted to holding circuit through C10.10. 5. 5.5 Melody Player : This circuit is used to record a special short msg or melody to be played if the hold circuit is to be used . 82 .11 SLIC (subscriber line interface card ): The slic is the interface unit between the PBX unit and the extensions.[Integrated VOIP System] The hook switch part is the circuit that provides a dial pulse in the telephone when the speaker is off-hook and the user is making or receiving a call and opens the circuit between the co-line and telephone when the speaker is on hook. this unit is required to detect every status from an usual telephone set for the Controlling circuit ( telephone exchange ) also it provides the ringing control to the extensions.4 Hook switch Circuit Schematic: 5.6 Melody player Circuit Schematic: 5.

5. It controls many functions such as:  Intercom calls. In our case the switching unit is computized and integrated at the microcontroller chip . One for dial and other for busy.12 Switching Unit: the function of the switching unit is to organize the calls handling between subscribers whatever the calling party was from extension to extension(intercom) or from extension to external call and vice versa.  call forwarding  Msg or music in hold.[Integrated VOIP System] 5. The microcontroller is the basic and only controller for all the exchange operations at that may occur in the PBX.13 Tone Generator: Tone generator is responsible for producing the different required tones at different cases such as dial tone which means that the telephone is ready to make a call and the busy tone which means that the called party is not available for making a connection now.1 Slic Circuit Schematic: 5. this stage is based on the CD4093 NAND chip where gate A oscillates at high frequency and gate B oscillates at a different low frequency then a third gate say C ―AND‖ the output of A or B to produce different tones .13. 5.  Call transfer.1 Tone generator Circuit Schematic: 83 .11.

14 Ring Generator: Ring generator is the part of the system that is used to provide ringing for any telephone instrument.15 DTMF Receiver (decoder): DTMF acronym stands for dual tone multi frequency. 5.[Integrated VOIP System] 5.15. a connection is made that generates two tones at the same time. A "Row" tone and a "Column" tone. 5.1 How does it work? When you press the buttons on the keypad. The circuit rings the phone in a completely realistic manner until someone answers. 84 . These two tones identify the key you pressed to any equipment you are controlling. The DTMF decoder recognizes sequences of DTMF tones from a standard touch tone phone. The problem is the ringer needs a high AC voltage to make a satisfactory ring which is done by the bell voltage transformer which Boost up the signal to the desired output signal depending on the transformer voltage.

16 Power Supply : Power supply unit is a unit that provides different supply voltages for all different electronics in the system where there is the microcontroller which needs 5v DC supply .also 12v DC required for all analog electronics and relays. the tone 697 is the same for both digits. but it take two tones to make a digit.15. Pressing the digit 2 will generate the tones 1336 Hz and 697 Hz.1 Power supply circuit schematic: 85 . And the 55v AC for the ringing voltage. 5. Sure.16. 5.2 Dtmf decoder Circuit Schematic: 5. you generate the tones 1209 Hz and 697 Hz.[Integrated VOIP System] When you press the digit 1 on the keypad.

G.19 Why using trixbox: Key Features:          . G.it defers only in the system requirements such as operating system (windows .726 and iLBC Includes most of Protocols such as H.. Axon. performing the callrouting activities of the traditional PBX or key system as a software system.. VOIP used protocol and some extra not basic pbx features such as voice mail . Call Transfer. SIP..Easy phone system management! Reduce long distance and inter office call costs No more expensive proprietary system phones .711 . they are almost the same .323.. or can carry out its functions virtually. Support for many Call features such as Call Recording.Use standard SIP phones Eliminate the phone wiring and make moving offices easier 86 . The virtual version is also called a "Soft PBX".. 5. call recording .[Integrated VOIP System] 5. Brekeke. No proprietary expansion modules needed! Web based configuration & status indication .Unlimited extensions and phone lines.) .etc. Trixbox.18 Different Software Used as SoftPBX: There are alot of companies that provides software pbx . MGCP. Linux. Pbxnsip.17 Software PBX: An IP-PBX can exist as physical hardware. Call Waiting and Caller ID Contains many codecs supported such as ADPCM.. This software are:      3CX. and IAX (Inter-Asterisk Exchange) Purchase cost dramatically lower than a traditional hardware PBX Scaleable . 5.

Guaranteed QoS So now you have the measures that make you able to decide which one to use dependant on your needs. Proprietary software.interfaces. To know which one to use we need to know the properties of each one: Hardware PBX:     Software PBX:      General purpose platform.20 Software PBX Vs hardware PBX: So now we have two choices for the PBX either a software or hardware .phones.. Not Guaranteed QoS Proprietary platform: Nortel.. Standards-based. Proprietary software.. Avaya. scalable hardware.[Integrated VOIP System] 5. Generally more expensive. 87 . Lower cost.

[Integrated VOIP System] Chapter 6 System Configuration 88 .

 Press the Enter key to start the installation. you will be asked to select the type of keyboard that you will be using. If necessary change the BIOS settings to enable this. the following screen will be displayed.[Integrated VOIP System] 6. Use your cursor keys to navigate the keyboard types and use the Tab key to move the focus to the OK or BACK option. After a few seconds.1 Overview This section describes the basic configuration of the system components that have to be configured to make a properly built voip system. it is not really necessary. Just press [Enter] to start the installation.  After initial system detection.  Ensure that your PC will boot from the CD. 89 .  Burn the ISO image that you have just downloaded to a blank CD. Select the appropriate keyboard.2 Trixbox installation After you have completed the download.  Boot your Trixbox PC with the CD in the CD Drive and press enter.You press F2 to see the various options. 6. However.Selected US and move the focus to OK and hit [Enter].

 Use your cursor arrow keys to navigate to the appropriate time zone. you will be asked to select the Time Zone you are in. 90 .[Integrated VOIP System]  After system hardware probing is completed. Use the TAB key to move the focus to OK and hit [Enter] to accept .  Next you will be asked to enter your root password (remember this password).

  After Linux is loaded the CD will eject.  From this point it will take about 30-45 minutes for the installation to be completed and ready for the configuration stage. 91 .  During this stage. All you do is. you will be presented with the following screen. Linux and the required files are being installed.[Integrated VOIP System]  After entering and confirming your password. When it reboots. installation will now commence by first formatting the Hard Disk/s. wait for it to finish. Take the CD out and wait for the system to reboot. you will see screens similar to the following.

This process will take a while because it is building Asterisk. 92 . it will reboot itself.  When Asterisk build is complete. Once rebooted. Trixbox© is ready for you to continue configuring and make changes to the system default. it will continue and you will see lots of lines of codes.[Integrated VOIP System]  After a moment.

g. select the type of trunk e. IAX2. Therefore you should select Generic SIP Device from the device drop-down list then click ―Submit‖. I used the extension number as the secret password) 93 .0.[Integrated VOIP System] 6.156) to configure Trixbox.3 Trixbox Configuration Now. using your browser. The important ones are:  User Extension: 100 (that‘s the extension number I gave for reception)  Secret: 100 (for simplicity. You will notice a few fields that you will need to populate. you can connect to http://ipaddress/ (e. Create Extensions To create extensions. SIP.0. The illustration below is where you create the extension. ZAP or Custom. is done from the Create Extension menu illustrated below: We need to create a few SIP extensions. http:10.g.

[Integrated VOIP System] I left the rest of the fields at their default values. 94 . Submit when done.

95 .[Integrated VOIP System] Click on the red bar on the top of the screen after every time you have submitted a new extension.

[Integrated VOIP System] 6. 96 .6:9999 then select ―Enter‖ on your keyboard  You will be presented with a log in dialog box. Select ―Submit‖ at the bottom of the screen to save this setting. Enter the ― Registrar Server Domain Name/IP address‖ as 10.  for example enter is as: http://10.0.0.0.  Leave this box blank and select OK to proceed to the DPH-120s web configuration interface.4 IP phone configuration  Open your web Browser  Enter your IP address directed to port 9999.156 Change the ―Authentication expire time‖ from 3600 >> 240.0.     Select SIP SETTINGS on the left menu.

registration will take around 10-30 seconds 97 .  At this point the DPH-120s will register and show the go talk VOIP username on the display screen of  the phone.  If all settings are correctly entered.[Integrated VOIP System]  Select ―SIP ACCOUNT SETTINGS‖ from the menu on the left  The Default Account should be set as Account 1  Enter your log in details with your go talk VoIP account details  Similarly enter your password  Select Submit at the bottom of the screen to save this setting.

98 .[Integrated VOIP System] Select the ―Restart‖ button to save all settings after changes have been made  The device is ready to use when you can see your go talk username displayed on the screen of the DPH-120s.

5 Softphone Configuration Start X-Lite. click on the Menu button and choose "Sip Account Settings" 99 .[Integrated VOIP System] 6. If this is the first time you have used X-Lite then the "Sip Accounts" screen will appear. Otherwise.

[Integrated VOIP System] Click on the Add button to open the Properties screen. Enter the following fields: 100 .

[Integrated VOIP System] 101 .

[Integrated VOIP System]

102

[Integrated VOIP System]

103

[Integrated VOIP System]

Appendix

104

[Integrated VOIP System] 105 .

[Integrated VOIP System] 106 .

[Integrated VOIP System] 107 .

108 .[Integrated VOIP System] The main MCU module PCB Pattern.

109 The CO-line Component layout. .[Integrated VOIP System] The main MCU module The CO-line PCB pattern.

[Integrated VOIP System] The Slic Module PCB pattern. Power supply PCB pattern. The Slic Module component layout. Power supply component layout. 110 .

[Integrated VOIP System] 111 .

[Integrated VOIP System] 112 .

[Integrated VOIP System] 113 .

[Integrated VOIP System] 114 .

[Integrated VOIP System] 115 .

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[Integrated VOIP System] 117 .

[Integrated VOIP System] 118 .

Competition Schedule: The MIE competition starts in August 2008 where teams will pass through phases starting by a summer training camp and ending with the competition finals in August 2009. Applying and Selection. .20th August 2008. Tiba Com 4-04. Made In Egypt (MIE) Competition is organized by IEEE Egypt GOLD Section "Graduates Of Last Decade". 20th June 2008 – 1st August 2008. Yes. (IEEE Competition). Summer Training. Yes. foster business-oriented results and develop prototypes for new products with the signature "Made In Egypt". 16th August 2008 . 30th August 2008 – 4th December 2008. • Sponsorship Details: o Having a sponsor is a must to continue in the competition phases. Yes. The competition aims at orienting Research and Development activities in Egypt to tackle real needs and problems. Made In Egypt (MIE) MIE is a competition that aims to bridge the gap between university researchers. 3-Governmental Organizations. o Sponsors Can be: 1-Industry Entities. Yes. foster business-oriented results and develop prototypes for new products with the signature "Made In Egypt". 119 Date. Done. 4th December 2008. What is the MIE? Made In Egypt is an IEEE GOLD program founded in 2005 and aims to orienting Research and Development activities in the Egyptian universities to tackle real needs and problems and develop prototypes for new products with the signature Made In Egypt (MIE). Submit Final Idea (1st Filtration Point).[Integrated VOIP System] MIE Competition. MIE vision and objectives Our slogan this year is "Think…Plan…Act" as we believe that to achieve any of your goals firstly you should think about what you want to do… plan for achieving this goal…start acting and working to achieve your goal. professionals and business community towards a better impact on our societies. The competition aims at orienting Research and Development activities in Egypt to tackle real needs and problems. 2-Private business companies. Basic Rules • Team Formation: Participants should form teams of minimum of 2. The competition phases are as follows: Activity. Market Research and Sponsor Engagement Phase. Overview This category is for Undergraduate students in their final graduation year at any governmental/private faculty of Engineering or Computer Science can participate with their graduation projects.

Yes. 2 Teamwork Create a team that can work together effectively. The training topics are: ## 1 Training Topics Communication and Presentation Skills Learn how to communicate and show your ideas professionally with different stakeholders in 1-to-1 and Panel situations (sponsors. business skills and technical training.[Integrated VOIP System] Concept Development and System Level Design. Entrepreneurship Get the vision and spirit of Entrepreneurship and know how to be an entrepreneur. Yes. Applying and Selection Students from each team should go to the registration page and fill the competition application form then submit the application electronically. Yes. Yes. 30th January 2009 – 1st February 2009. Yes. The MIE Committee will select the team who will continue in the competition phases based on the info included in the application form. 3 4 5 120 . March 2009 – July 2009. Stand by to it. Mid Year Training. handle conflicts and survive united to the end. The training will include soft skills. 23rd May 2009. 6 Thinking Hats Utilizing a variety of approaches within thinking and problem solving. Negotiation Skills Negotiate with sponsors and different stakeholders targeting a Win-Win deal. 1st August 2009. Summer Training The selected 350 students will attended a summer training camp in Cairo or Tanta with accommodation support. judges…etc). Submit 1st Report. Detailed Design and Testing Phase. The training represents an orientation phase for teams to help in enhancing their entrepreneurial skills to take their innovations into market. understand each others. 7th February 2009. thus servicing the needs of all individuals concerned. December 2008 – March 2009. Finals (Prototype is ready). Submit Final Report (2nd Filtration Point).

Industry Analysis & Competitive Strategy Understanding market analysis and how to reflect research and analysis result on developed product/service. Submit Final Idea (1st Filtration Point) The final proposal consists of: 121 . put an initial technical design and get a sponsor for this idea. company information and sponsorship deal to continue to next MIE phases. financial. 6 7 8 9 10 11 12 Market Research and Sponsor Engagement Phase Teams should study the market and make a market research with the target of having a project idea that satisfies real need. You have 2 scenarios to get a sponsor for your project. added values and competitive edges. tools…etc) and plan for different risks. the sponsor should send to MIE sponsorship confirmation email with the final proposal attached. either to contact sponsors on your own or send to one or more of the companies in contact with MIE. Project Management Learn how to plan the project time plan. How to get a sponsor Learn how to get in contact with potential sponsors and how to present your project idea effectively to get a winning deal. Create innovative ideas.[Integrated VOIP System] Fundamentals of Marketing Understand basic marketing concepts and customer/market needs. allocate resources (Human. Product Design and Development Learn how to get into a professional product development lifecycle starting from product planning and gathering customer needs until making your prototype. At the end of this phase every team must have a sponsor and apply a full proposal containing project details. Intellectual Property (IP) and Copyrights Know the concept and importance of IP and learn how to protect your innovations through a set of best practices. In both cases. Business Writing Articulate your ideas clearly and effectively through abstracts and proposals to get a winning deal and communicate professionally with sponsors and different stakeholders using different types of writing.

It represents a very informative and useful source for you during this period. Sponsor Info Form: It shows sponsor company information. It's very important to read the Product Design and Development Book 122 . The filtration will be based on the final proposal. it‘s highly recommended that you fill it first then review it with them till both of them accept it. Sponsor and the Team. IUL form: It helps in shaping the relation between the team and the sponsor by defining the sponsorship deal. Make sure that you add all required contacts correctly. Supervisor. o Business Vision. It also shows some general terms for the sponsorship deal. Teams should refer to the book ―Product Design and Development‖ for more information about this stage. It‘ll cover these main points: • Sponsorship Deal o You must have a sponsor by this time and must have submitted all the required forms. o The sponsorship deal will be examined to make sure it‘s a true deal and represents an added value for the Team and/or the Sponsor. To make it easy for the sponsor and the supervisor. Project Info Form: It shows the project technical and business aspects. the progress they have achieved. Concept Development and System Level Design In this stage each team should generate different design concepts for their project and find an integral solution for any problem related to the project.[Integrated VOIP System] 1. Submit 1st Report This is mainly concentrating on the market research phase. problems that they overcame and problems that still exist to the committee and to other teams. 2. o You will be evaluated by the committee and get advised for the problems either in the team management or in the project progress. 3. This is to be filled by the sponsor. • Project proposal o Technical Approach. Mid Year Training In this training there will be seminars to attend with topics suitable to this stage: ## 1 2 3 4 Training Topics Training Topics Public Speaking Business Planning Technical Writing How to start your own business 5 Entrepreneurship and Business Networks There will be also an evaluation meeting at the same time as: o Teams will present their projects. The IUL Form should be filled and accepted by all the 3 involved parties.

LEGEND Technologies. 2. Jelecom. Triple A. Eng\Michael Nasif (faculty of Engineering Shobra). Eng\Ahmed Abaas (Telecom Egypt). Detailed Design and Testing Phase Teams should finish complete specifications of the product – materials. Don‘t postpone it till last days. You should start working on this report during the market research phase to make best use of it.Egypt A.  The Judges will have your projects‘ abstracts and reports to have a clear picture about the project. We are :- ADNAN ADEL AMIN MOHAMMAD ENAS ADEL YOSEF EL TIGANY OTHMAN MOHAMMAD HANAFY ALI 123 . Also should finish prototype forming and evaluation on several copies before yielding the real product. Thebes Academy. This report will guide you through the book and will be very supportive for you through this phase. The filtration will be based on your project‘s technical and business aspects in more details. Dr\Ahmed El-shazly (Chair man of electronics & Communication in Thebes Academy). purchased parts…etc – and prepare the process plan for the fabrication and assembly of the project. Finals (Prototype is ready) Judging Panels The teams will present their projects to judges‘ panel:  There will be 2 judges in your projects‘ fields. geometry. We had Succeeded all this Stages & We waiting to finals MIE to select the best five (5) Teams to connect to Made In Arab (MIR). Submit Final Report (2nd Filtration Point) This report will cover the technical design and the prototyping issues. 3. Us in MIE Our code is Tiba com 4-04. Eng\Tamer Saleh. ASK PC. 1. 4. 7. 9. Under the Supervision of: 6. Sponsor with Us.D Panasonic. and more. 5.[Integrated VOIP System] and apply it practically in your research. Qualified teams will make it to the MIE Finals. It‘ll be very useful to guide you to reach a strong design and get used to make technical design documents. 8. one will judge it from the technical and other from business points of views.

[Integrated VOIP System] AHMED REFAT TWFIQ MOHAMMAD ADEL MORSY MOHAMMAD ABD EL-MAKSOUD MOHAMMAD IBRAHIM MEDHAT IBRAHIM AMGAD EHAB SCANDAR 124 .

[Integrated VOIP System] References [1] Trixbox made easy by Barrie Dempster and Kerry Garrison [2] Trixbox without tears by Ben Sharif [3] Cisco press at Cisco.wikipedia.com 125 .com [4] Triple a Training Company [5] VoIP for Dummies by Timothy Kelly [5]www.

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