Developing service VOIP in Viet Nam

VoIP ( Voice over IP- that is, vioce delivered using the Internet Protocol) is a term used in IP telephony for a set of faccilities for managimg the delivery of voice information using the Internet Protocol(IP). In general, this means sending voice information in digital form in discrete packets rather than in the traditional circuit – committed protocols of the public switched telephone network (PSTN). A major advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary telephone service. VoIP, now used somewhat generally, derives from the VoIP Forum, an effort by major equipment providers, including Cisco, Vocltec, 3 Com, and Netspeak to promotethe use of ITU-T H.323, the standard for sending voice (audio) and video using IP on the public Internet and within anintranet. The Forum also promotes the user of directory service standard so that user can locate other users and the use of touch-tone signals for automatic call distribution and voice mail. In addition to IP, VoIP uses the real-time protocol (RTP) to help ensure that packets get delivered in a timely way. Using public networks,it is currently difficult to guarantee Quality of Service (QoS). Better service is possible with private network managed by an enterprise or by an Internet telephony service provider (ITSP). A technique used by at least one equipment manufacturer, Netspeak, to help ensure faster packet delivery is to Packet Internet or Inter- Network Groper (Ping) all possible network gateway computeres that have access to the public network and choose the fastest path before establishing a Transmission Control Protocol (TCP) sockets connection with the other end. Using VoIP, an enterprise positions a “VoIP device” (such as Ciscos AS5300 access server with the VoIP feature) at a gateway. The gateway receiver packetixed voice tranmissions from users within the company and then routes them to othe parts of its intranet (local area or wide area netnork) or using a T- carrier system or E-carrier interface, sends them over the public switched telephone network (PSTN)

Ha Noi open university

Center For International training

Co-operation

Thesis:

Teacher : NguyÔn Th¸i Nguyªn

Group 3 : §ång Xu©n Th¾ng -Cap

GlossaryATM : Asynchronous Trasfer mode

ACELP : Algebraic Code Excited Linear Predictive

ARQ : Automatic Rrepeat Request

ACF : Admission Confirm

DES : Data Encryption Stadard

PSTN : Public Switched Telephone Network

PC : Personal Computer

PCM : Pulse Code Modulation

IP : Internet Protocol

ITU : International telecommunication Union

IETF : Internet Engineering Task Force

ISUP : ISDN User Part

INAP : Intelligent Network Application Part

ITSP : Internet Telephony Service Provider

MAP : Mobile Application Part

MGCP : Multimedia Gateway Control Protocol

MTP : Message Trasfer Part

MP : Multi point

MCU : Media Control Unit

OLC : Open Logical Channel

QoS : Quality of Service

RC : Report Court

RSVP : Resource Reservation Protocol

RTCM : Real Time Control Mode

RTP: Real Time Post

SIP : Session Initiation Protocol

SS7 : Signal No.7

SCCP : Signaling Connection Control Part

STP : Signaling Transfer Point

TCP : Transmission Control Protocol

TCAP: Transaction Capabilities Application Part

UDP : User Data Package

VAD : Voice Activity Detector

VoIP : Voice over Internet Protocol

General of the thesis VoIP -Voice over Internet protocol

VoIP ( Voice over IP- that is, vioce delivered using the Internet Protocol) is a termused in IP telephony for a set of faccilities for managimg the delivery of voiceinformation using the Internet Protocol(IP) In general, this means sending voiceinformation in digital form in discrete packets rather than in the traditional circuit– committed protocols of the public switched telephone network (PSTN) A majoradvantage of VoIP and Internet telephony is that it avoids the tolls charged byordinary telephone service.

VoIP, now used somewhat generally, derives from the VoIP Forum, an effort bymajor equipment providers, including Cisco, Vocltec, 3 Com, and Netspeak topromotethe use of ITU-T H.323, the standard for sending voice (audio) and videousing IP on the public Internet and within anintranet The Forum also promotes theuser of directory service standard so that user can locate other users and the use oftouch-tone signals for automatic call distribution and voice mail

In addition to IP, VoIP uses the real-time protocol (RTP) to help ensure thatpackets get delivered in a timely way Using public networks,it is currentlydifficult to guarantee Quality of Service (QoS) Better service is possible withprivate network managed by an enterprise or by an Internet telephony serviceprovider (ITSP)

A technique used by at least one equipment manufacturer, Netspeak, to helpensure faster packet delivery is to Packet Internet or Inter- Network Groper (Ping)all possible network gateway computeres that have access to the public networkand choose the fastest path before establishing a Transmission Control Protocol(TCP) sockets connection with the other end

Using VoIP, an enterprise positions a “VoIP device” (such as Cisco’s AS5300access server with the VoIP feature) at a gateway The gateway receiver packetixedvoice tranmissions from users within the company and then routes them to otheparts of its intranet (local area or wide area netnork) or using a T- carrier system orE-carrier interface, sends them over the public switched telephone network (PSTN)

Chapter1:Voice over IP (VoIP) Technology

1 Fundamental features of channel switching network and Internet:

1.1 Fundamental features of channel switching network:

The channel switching network is designed for rapid connect andeliminating the ineffectiveness of time-consume on connecting In the channelshifting network, the user is provided a conductive channel to exchangeinformation together When the exchange completed, the conductive channel isreleased This could lead to loss because of limits of conductive channel Theutility is low but ensures the calling quality because a two-way 64 kbps channel

is set aside for caller and receiver The channel shifting network is designedoptimum for real transmission time with high service quality In the channelswitching network, all terminal equipment and switch board are inserted a fixednumber so no need to enter address for information exchanging process Theswitching system in channel switching network will base on the address ofcalled subscriber to define the conductive line Because the band width isensured not be changed during calling, calling fee of channel switching network

is based on distance and calling time

1.2 Fundamental features of Internet:

Internet is the package switching network suitable with applications thatare not exchanged according to the real time; Package delay doesn’t effectstrongly on service quality like email and file transmission Package switchingnetworks don’t set aside a fixed line between two users, so, not ensure theservice quality All information on the network are divided into packages, thesepackages contain the destination address and its order

Channel fixer and host on the network will send these packages to thetargeted address On Internet, all packages are treated the same with outdistinguishing their contents When packages to the destination address, theywill be arranged according to the initial number By form of packageinformation transmission, the utility is maximum However, real timeapplications will be greatly effected on service quality The fee is not calculated

on distance or time but on used band width On Internet, on address of package

is marked by IP address, the IP address will be named for the host and terminalstations Channel fixing will be controlled by the IP destination address Tocreate a understandable, convenient address type for the IP address by name likeservice of regional name or email address

Because the limit of IP address, the users are temporarily inserted IP whiledialing The IP address is only for one terminal equipment while connectingInternet and deleted while not connecting The deleted IP address will be usedfor another connecting on the network

1.3 Advantages of VOIP against PSTN:

The users will pay for used time of PSTN if more time for call establishment,more increased fee to be paid At one time, they can contact to one person Butwith VoIP, the time for call establishment is independent to subscriber’s fee Onesubscriber could have calls to different ones and exchange data, dialogue, pictures,paintings and video with other subscribers

Figure 1: The basic structure of telephone network by IP

1.4 Outlook of VoIP technology:

+ Some technical features of IP telephone:

By analysis of fundamental features of channel switching network andInternet, we see that it is typical to accumulate real time signal into the packageswitching network and IP telephone Firstly, we should classify IP telephones.All IP telephones change according to 3 characters: type of terminal equipment,position of gateway, between IP and PSTN networks and main transmissionequipment

a Terminal equipment and gateway: There are 03 main types of IP They are PC

to PC, PC to Phone, Phone to Phone

+ PC to PC is the first model of IP telephone Users at two ends of PC to PCshould have 1 PC that is equipped audio, a software and connected toInternet This service no need gateway and PCTN because PCTN never switchthese calls, the main transmission tool is public Internet Due to sound qualityand complexity of use, the PC to PC has a litter affect on traditional telephoneservice

+ PC to Phone expands the number of users but for exploiters, the call of PC toPhone is more complex than that of PC to PC

+ Phone to Phone is very important market including mainly commercialservices, because, people prefer to communicate by phones However, the 3rd

model of IP requires more investment capital because it needs input gateway

to PSTN near places providing service Services of Phone to Phone are nearlysimilar to that of traditional telephones

b Transmission equipment: The classification between IP and VoIP telephone is

based on the nature of main transmission equipment IP telephone is for voicetransmission, fax and services relating to package switching networks on IP.Internet phone and VoIP are basic types of IP Internet phone is IP in whichthe main transmission network is public Internet (global super-network) Voice over IP is IP in which the main transmission network is private-usedone basing on IP

Besides, being the replacing tools for distance and international phone, the IPtechnology creates a plenty of other services that can transmit every service by

IP This part only mentions the technology of VoIP and interests in the terminalequipment that is telephone on the channel switching network (Phone to Phone)

Figure 2: IP call: Phone to Phone

+ Special features of VoIP:

a Adjustable quality: The quality of VoIP depends on each part (coding and

low speed re-coding for each part) Internet is not specific service network,the exchanging methods are entirely selected by terminal systems Thus, theterminal systems can control the compressed volume on the networkbandwidth or content for transmission

b Security: Using SIP to order a password and confirm messages indicating the

terminal RIP make and the password to be the password of transmissionmethod Therefore, all program is coded to secure transmission

c Users interface: Terminal systems of VoIP have plentiful indications and can

give out instructions and various graphic interface

d Connecting telephone and computer: Available to solve these complex

connections

1.5 Conclusion:

The VoIP technology has potential for future development, ability to replacethe existing PSTN network Due to differences in features of channel switchingnetwork and Internet, to apply VoIP for users of channel switching network(Phone to Phone), these differences should be solved Concretely, there should

be address changes, indication of two networks and proper inter-code forapplication of time transfer on network

2 Problems relating to VoIP technology and talk quality on VoIP:

Using the traditional channel switching telephone network will cost muchwhen at distance, to reduce expenses for distant calls, use public data network

or private data network for communication The package switching network thatapplies IP is example Using the package switching network by IP to transmitthe talking signals Voice over IP-VoIP is good basis to design global multi-instrument transmission system that can replace the infrastructure of existingnetwork Accumulating Audio, Video, data, fax into a single commonnetwork on IP technology It is possible to apply the Frame relay orunsynchronous transmission technology ATM to replace IP technology TheVoIP is more economic for distant call, because the fee is calculated by thewidth of bandwidth, not by distance In IP, it uses talk compressing technology

to save band width leading to cost reduction but the IP’s quality not as good asthat of PSTN

The biggest difference when applying into the multi-instrument network isactual time service non-actual one With actual time service and like Audio,Video not allow over-delay on the network; in non-time network like email,file transmission, the delay is not worthy worrying So, to carry out VoIP,special compressing and coding methods should be used to reduce the speed oftalk signals that can’t be use 64 tps like channel switching

2.1 Coding techniques and talk signal compression:

In talk transmission, voice is usually numberidized and coded PCM byRule A or U with speed of 64 Kps recorving sound rather actual For somespecific applications such as transmitting talk signals on TP network, sounds aretransmitted with lower speed, so, there should have coding techniques and talk

signal compression to lower speed according to standard of ITU and ETSI likeG723.1; G729; G729A; GSM

+ Standard G7213 According to the standard of ITU, the coding has 5.3Kbpsand 6.3Kbps The compression technicque uses MP-MLQ for high bit speed;for coding with low bit speed using ACELP Delaying against algorithm is67.5ms

+ Standard G.729 According to the ITU standard, this coding has speed of 8Kbps This compression techniques uses algorithm predicting coded linearlinked structure algebra excitation Delaying against algorithm is 25ms

+ Standard GSM06.10 According to ETSI, this code has 13Kbps Thiscompression technique is regular pulse excitation and long-term predictor.Delaying against algorithm is 40ms

2.2 Voice Activity Detector (VAD):

VAD is carried out by numeric signal processor to reduce the talk intensitythat is transmitted by automatically detecting the dead space on the talk andstopping transmitting at that time There are space approx 50-60% of almosttalks This always occurs because when one speaking, the other must listen to.VAD allows band width for dead space saved for reserving other data

VAD actives by controlling power of talk signals; power change is change oftalk signal frequency The difficult of VAD is to define the exact time of talkending and of talk signal The double VAD is nearly 200ms after recognizingtalk signals and stop and detect package processing This top prevent VAD frommissing the end talk or in the middle of small interrupt in talks

2.3 Number and address:

Due to cooperation between IP and SCN networks, there will be 2 types ofaddress: address in CSN and in IP

a Numbering on SCN network:

On the channel switching network, all terminal and switchboard are fixed anumber Number E164 is telephone numbers subject to the structure andnumbering program that were described on the proposal E164 by International

Telecommunication Union The line fixing process on the channel switchingnetwork is controlled by the address system of E164 Before dialing, the users

of channel switching network have to dial E164 and callee’s number

+ Local number:

Code of Access Caller + National Post + National Destination Code +

+ Subscriber number

+ For international numbers, we can use 03 following structures:

Code of Access Caller + International Post + Country Code + IdentificationCode + Subscriber number

Code of Access Caller + International Post + Country Code + DestinationCode + Subscriber

Code of Access Caller + International Post + Country Code + GlobalSubscriber’s Number

b Numbering on IP:

+ Prefix is an identifier including one or more numbers allowing the usednumerical types, network and service and can be used to select serviceprovider, type of service in a nation

+ Selecting service provider including numbers that allow to select service by

IP network or SCN and there of to select appropriate switching

+ Selecting service provider can be done by ways: pre-select by user or dialing,password

Incase, the Gateway connects to SCN where there are a lot of service providers,both Gateway and Gatekeeper should be able to identify and process theselected code of service provider Incase, a lot of service providers on IPnetwork, Gatekeeper is able to identify and process the selected code of serviceprovider

To get the most common address types on Internet, it can use name address likeemail address: user@domain , user@host , user@IP-address , phone-number@gateway

2.4 Fee:

To ensure the effectiveness of network, the fee calculating will be done by

a separate host system The fee-calculator host will be responsible for collectingand reserving all detail of call from gateway or MGC These data are used tomake invoices for customers Customers ca access into the host for their feedetails on the website The fee will be calculated by the used time The feecalculating system should be able to calculate on 2 types of service: pre-paidand post-paid This software must be able to carry out some following function

- Accepting call

- Informing the amount of account

- Fee calculating based on pre-fixed level for different directions

- Informing the maximum time of call

- Updating account’s amount after calling

2.5 Signal cooperation:

The standard of signal communication of IP Phone to PSTN is suggested to

be signal No 7 (SS7) The SS7 is used to transmit following information:

- Information o call establishment

- Information about call control

- Property and application

The signal communication between 2 IP networks and signal network 7 of PSTN

is carried out by signal Gateway The signal gateway connects to STP on the SS7

as a SP and transfer signals fully The signal Gateway should support signalnews ISUP and SCCP/TCAP

Using the signal communication No 7, IP telephone network will bring benefits

as follows:

- Fully connecting to PSTN

- Supplying additional services

- Improving call control

- Improving maintaining property for trunk

- Speeding up call establishment

Although new signaling, ssuch as H.323 ans SIP, exist for VoIP networks the standard in traditional telephony and in mobile networks is SS7.Therfore, if a VoIP based network is to communicate with any traditionsnetwork, not only must it network at the media level through media gateways,

it must also interwork with SS7 To support this, the IETF has developed a set

of protocols known as Sigtran

In order to understand Sigtran, it is worth considering the type of interworking that needed to occur Imagine, for example, an MGC that control one

or more media gatways The MGC is a call control entity in the network and,such as uses call control signaling to and from other call control entities Ifother call control entities use SS7 then the MGC must use SS7 at least to theextent that the other call control entities can communicate freely with it Thismeans that the MGC does not necessarily need to support the whole SS7- justthe necessary application protocols

Consider figure 3 which shows the SS7 stack The bottom three layer are

called the Message Transfer Part (MTP) This is set of protocols responsible

for getting a particular SS7 message from the source signaling point to the

destination signaling point Above the MTP we find either the Signaling

Connection Control Part (SCCP) or the ISDN User Part (ISUP) ISUP is

generally used for the establishment of regular phone calls SCCP can also beused in the establishment of regular phone calls but it is more often used for

the transport of higher layer applications, such as the GMS Mobile

Application Part (MAP) or the Intelligent Network Application Part (INAP).

In fact most such application use the services of the Transaction Capabilities

Application Part (TCAP) which in turn uses the services of SCCP.

Figure 3: SS7 Stack

Application Part ISDN User Part

(ISUP) Transaction Capabilities

Application Part (TCAP) Signaling Connection Control Part (SCCP)

MTP Level 3 MTP Level 2 MTP Level 1

SCCP provides an enhanced addressing mechanism to enable signalingbetween entities even when those entities do not know each other’s signalingaddresses (known as point codes) This addressing is known as global titleaddressing Basically it is a means wherby some other address, such as atelephone number, can be mapped to a point code, either at the node thatinitiated the message or some other node between the originator anddestination of the message

Figure 3 provides some examples of communication between differentSS7 entities Consider scenario A In this case, the two entities, represented

by point code 1 and point code, communicate at layer 1 At each layer, a peer

to peer relationship exists between the two entities Scenario B has a peer topeer relationship at layer1, layer 2, and layeer 3 between point codes 1 and 2,

2 and 3, and 3 and 4 At the SCCP layer, a peer to peer relationship existsbetween point codes 1 and 2 and between point codes 2 and 4

At the TCAP and Application layers, a peer to peer relationship canonly take place between point codes 1and 4 In other works, the application atpoint code 1 is only aware of the TCAP layer at point code 1 and applicationlayeer at point code 4.Similarly the TCAP layer at point code 1 is aware only

of the application layer above it, the SCCP layer below it, and thecorresponding TCAP layer at point code 4 It is not aware of any of the MTPlayer Equally, if we consider communication between point code 2 and pointcode 4, the SCCP layer at each point code knows only about the layeer above(TCAP), the layer below (MTP3), and the corresponding SCCP peer As far asthe SCCP layers are concerned, nothing else exists Therefore, SCCP neitherknows nor eares that point code 3 exists Consider Scenario C, where ppointcode 3 is replaced by a gateway that supports standard SS7 on one side and an

IP based MTP emulation on the other side Point code 4 does not supportr thelower SS7 layeers at all- just an MTP emulation over IP Provided that theMTP emulation at point code 4 appears to the SCCP layer as standard MTP,then the SCCP layer does not care, not do any of the layers above SCCP.Equally the SCCP layers at point code 1 and 2 do not care Consequently, it ispossible to implement SS7 based applications at point code 4 withoutimplementing the whole SS7 stack This is the concept behind the Sigtranprotocol suite

Point code 1 Point code 4

Scenario A - Communication Between Adjacent Signaling Points

Point code 1 Point code 2 Point code 2 Point code 4

Scenario B - Communication Between non- Adjacent Signaling Points

Point code 1 Point code 2 Point code 2 Point code 4

Figure 4 Example SS7 Communication Scenarios

ISUP MIP3 MIP2 MIP1

ISUP MIP3 MIP2 MIP1

2.6 Confidence:

The IP service active on the base of IP switch, the requirement ofconfidence is very important for:

+ Protecting exploiters from bad activities

+ Protecting exploiters from network troubles by faults of network components.+ Protecting users from bad activities

To ensure above targets, the network should protect for 5 following services:

+ Data Coding: This is the most effective practical method to protectinformation that are transmitted through different networks Regularly, thedata is compressed by different standards by Gateway, may be, no need tocode the data If necessary, information on network are advised to code byDES (Data Encryption Standard) with the key of minimum 56 bit long

+ Anti-virus: Virus can cause significant consequences to the software of allsystem Virus could be spread from other system or customers’ This alsocarries significant meaning when the system operates on base of thedispersion processing structure Anti-virus software should be installed onGateways and hosts of gatekeeper

+ Using Firewall: This is important method to protect the network of exploiter.There are 2 basic mechanism of Firewall are to stop information and allowFirewall information to-

- Stop all coming data except the resource is confirmed

- Release all data except for propaganda and regional checking data

Even, using firewall is effective, to ensure high confidence, coding andconfirming methods should be used

+ Confidence for distant access:

To control distant access, there are following methods:

- Confirmation: Distant subscribers should be controlled

- Access Limit: Fixing each distant subscriber a specific position on server

- Time limit: Fixing connecting time, if it is over, connecting will becancelled

- Connecting limit: Limiting on connecting times and starting points ofconnect

+ Confidence policy:

Confidence plan should include following elements:

- Definition of access levels regulating user to access into relevant resource

- How a subscriber on subscriber group access into the network

- Access Regulations: Time, place and how to use services

- Instructions for fee calculation

- Requirement on network accessing and connecting

- Ability to strengthen confidence methods in specific cases

- Instruction on confidence for users

2.7 Troubles relating to calls quality:

- Wave transmission delay: This is necessary time for optical or electricsignals on transmission environment to certain geographic distance

- Structuring delay: This is delayed time created by different components in

a transmission system For example, a frame across a line fixer shouldmove from Gate to Door across server body There is a minimum delaythrough server body and changeable delay by in line and processing of linefixer

+ Echo suppression

The first trouble caused by the delay is echo impact The echo can beoccurred on a talk network by chain-jointing between the listening and speakingparts of the complex This delay is called auscosic delay This also occur when apart of power energy is reflected to the speaker by a exotic line in PSTN, thatcalled echo

If the time of one-way delay or terminal delay is short, every echo created

by talk line is back to the speaker rapidly and non-noticeable In reality, no needecho suppression if one-way delay is smaller 25ms However, the one-waydelay of VoIP almost over 25ms, so the echo suppression is required

+ Superposition of voice

If the best ability of echo suppression, switching 2-way talk become verydifficulty when the delay is too long causing voice superposition This occurs whenone party reduces voice of the other when the delay is too large

+ Jitter - Changeable delay

While phone services require to transmit according to the fixed delay, thedata network that badly transmit and can’t supply the fixed delay becausedifferent packages have different delay, so, different delay frame Resourcescreate regularly frames, the Destination gate can’t collect these frames regularlybecause of Jitter Jitter interrupts the call and difficult the talk content Toremove the changeable delay, it should receive frames and keep them forenough time So that the latest frames come timely for reading in order Thebuffer can remove the fitter No worry on this for PSTN, because, the bandwidth

is fixed Volume of Jitter is more big, the longer frame kept on the buffer and

create more time delay If the Jitter is small, use small buffer If Jitter increased

by increase of loading, the size of buffer will automatically increase

Packages will destine after some fixed time (for example, after 20ms).Incase of Jitter, this is not true The figure below illustrates the Package 1 (P1)and package 3 (P3) coming timely; but Package 2 (P2) and Package 4 (P4) latefor 12ms and 5ms against expected relatively

Figure 5: Jutter description

IP network for high service like video, mobile and high-quality talks, anothersignal system is required to solve this, it is signal system No 7

+ Bandwidth:

A traditional talk uses a 64Kbps flow When the talk flow is on IP network,

it will be compressed and numericalizied by Digital signal processor Thiscompression reduce speed of talk to 5.3Kps for a talk, then, packed into IP

network, IP/UDP/RTP starters are added This large the band width for each call(about 40 byte for each package) However, technology for example, forcompressing the RTP starter may reduce the IP starter to 2 bytes The bandwidthdepends on byte coding speed and talk package size The private IP network hasmore advantages than Internet does because of more bandwidth so, voice quality

is better Defining the bandwidth on the network, number of call at peak time.VoIP can reduce the bandwidth by talk signal compression and deadsuppression

3 Transfer modes:

TCP and UDP are two modes for data transmission on IP network

+ TCP is good protocol for data transmission that can control flow and block,protect from over-loading on the network However, there are someunfavorable matters when using the TCP mode Due to the reliability of leyteservice and retransmission of lost packages increasing the delay of network.TCP has a lot of properties and complexity, this is not benefit for VOIPtechnology When transmitting talk signals, they should be distributed tousers at the same time On TP network, there should have effectiveness fordistributing multi transmit-feedback data, however, TCP can’t supply this Ifthe data are distributed to destinations on TCP, single TCP will be required toconnect causing cost of bandwidth

+ UDB is protocol simpler than TCP, just an expanded ID mode, only usedwhen no requirement for high quality service This protocol has advantagethat no waste of time for re-transmission of lost packages

It can use the property of multi-transmit and feedback and save bandwidthwhen data sent to a lot of destination UD Palso has disadvantages, nosynchronous mechanism and no means to control flow and block To solve thismatter, cooperate UDP and modes controlling the real time

3.1 Real time mode:

3.1.1 Real Time Post:

RTP can distribute among terminals of real time services like audio, vide.The typical RTP is used to transmit data through UDP (User’s Data Package).RTP and UDP supply functions of protocol transfer UDP supplies multi-

elements and error checking service RTD is also used with other transferprotocol When a host desires to send a package, it should know transmissionmeasure to make package shape, add the specific transmission measure into thetitle of package to pre-decide the RTP’s title and put into the lower layertransmission measure Then, send to network by multi transmit-feedback orsingle transmit-feedback ways to other participants

Format of RTR fields are described as follows:

Figure 6: News on real time Post Mode

Fields of RTP header are:

+ Version (V, 2 bytes) defines version of RTP

+ Padding (P, 1 byte) If padding is installed, a package contains one or moreOctet padding adding to the terminal that not belong to pay load The finalOctet of padding includes number of ignored octet padding Padding mayneed more other coding algorithms with changeable sizes of block or bringsome RTP packages in low layer data unit mode

+ Extension (X, 1byte) If X byte is fixed, Fixed Header will allow Header have

an extension

+ CSRC Count (CC, 4bytes) CCRS Count include some CSRC defining quantity

of resource participants, shown on Fixed Header

+ Marker (M, 1byte) Marker is defined by a profile, it means to allow signal,events like marking frame margin on information package M Byte suppliesinformation to re-create and release package in case of defining the firstpackage on released voice

+ Payload (PT, 7bytes) Fixing the transmission measure (Editing, changing thebandwidth to be sufficient for transmission on each travel) RTP and detaileddescription

+ Sequence number (16 bytes) Sequence number increases each value for eachdata package sent by RTP, and search for lost packages and recover them inorder

+ Time stamp (32 bytes) Time Stamp feedback a sample for the first octet onRTP data package This sample should be taken from a information package

by a simple o’clock and linear in a period for synchronization

+ SSRC (32 bytes) In case SSRC defines, show out the synchronous resources,this definition is selected at random to avoid two synchronous resources inone RTP session

+ CSRC list (0 - 15times, 32 bytes/field): CSRC list defines, show resources forload (volume) in information package The quantity of fixed sets is recorded

on the CC field If there are more than 15 resources for information package,only 15 set are defined CSRC show out and insert, using SSRC to definecontributing resources

+ RTR Header Extension (variable length) An optional extended mechanism issupplied with RTP allow each implementation to test new functions requiringmore information on RTP Header

3.1.2 Real Time Control Mode:

The RTCP is the basic to control continuous transmitted packages toparticipants on communication session by using the same distribution

mechanism for data packages The low modes must pill up data packages andcontrol by using different port number and UDP Functions of RTCP aredescribed as follows:

+ Supplying feedback on the distributed data quality This is major part of RTR;the protocol transports and relates to the flow controlling function and block

of other controlling mode Feedback are very useful for controlling codingsets However, testing with IP multicast also give out results againsttransmission of feedback from the receiving end to diagnosing distributionerrors Sending a feedback to all supervising points to define problems, errors

by local or central By distribution mechanisms like IP multicast, can do foreach unity as service providers and not be attracted into other aspects oncommunication sessions, receive feedback and act like the 3rd representative

to diagnose network errors

+ Bring a fixed load to RTP resource called C.Name When SSRC is defined to

be changeable if a conflict is found or the program is reset; required receivingpoints of C.Name keep way for terminal Receiving points also requireC.Name conjugate to data lines from each giving point in mutual relations onRTP session, for example, audio, and video

+ Two first functions require all points participating into communicationsession send RTCP, so, the speed must be controlled by RTP to arrange agreat number of communication points Each communication point can sendinformation package to other points, each point can supervise independently

to others

+ An optional function for a minimum post session, like fixed communicationpoints to display on user’s interface It seems very suitable, useful on looselycontrol sessions where communication points in and out don’t need membercontrolling measures or negotiation parameters RTCP serves as a usefulchannel to reach to communication points But it thinks that not necessary tosatisfy all transmission control required by application

There are 5 package identifiers:

- SR: Sender’s news is created by users, they also send transmission measures(RTP resources) They describe the sent data quality like correlation withtime stamp, RTP sample and absolute time for synchronize different means

- RR: Receipt’s news is to create components participating into RTP session.They’re receiving transmission measures Each such news contains a blockfor each RTP Each block describes a immediate coefficient and the fitter(like phase drift) from this sources The tensioning block shows the finallabel and the delay from receiving sender’s report, allow resources estimatetheir distances

- SDES: Resources labeled packages for controlling session It includeC.Name, the unique identification like frame an entail address The standardname used to resolve conflict in synchronized source value and deferentcombined communication protocol current is created by such user SDESpackages also identify members through its name, email and talk data,supplying simple control form

- BYE If an user leaves participating in own-self session with BYE message,

so each member can know the total members participated in

- APP: Specific applying elements (APP) to add give further specificinformation in to packages

Identifying header park as follows:

of data block changed In permissible RTCP information package, padding

can be required on final information package because informationpackages will couple complete code.

- Reception Report Court (RC) Volume of reception report Court will lumpall to RTCP package value is equal zero is legal May have to 200 constantdeterminations of RTCPSR package

- DT: Determining the load type is which information in 5 kinds of newscast

- Length: The length of news package is a number of 16 bit, includingheader and added padding

3.1.3 RSVP:

RSVP not provide separate transmission protocol but still use IR,RSVP is only control protocol to supply quality of service (QoS) ensuring tothe application The host transmit data that need to reach any QSS, it sendthe call to destination address owning to newscasts include information oncharacter and flow RSVP not is line-de fixing protocol, simple selects amost optimal line This can’t give an ideal QsS RSVP is an importantinstrument to QsS, but not resolve all necessary problems related to QsS Onthe transmission to aim of RSVP allows router save information ontransmission newscast, with this way, its used when prior keeping data fromnewscast sender on the transmission line When the user receive,transmission line newscast, it can decide to receive data or not of the senderwith QsS fixed For meeting QoS fixed, RSVP will send periodically arequirement of prior keeping under prior keeping line of transmission linenewscast

Precise prior keeping line attained owing to information in RSVP Theprior keeping newscast includes 2 parts: containing QsS that collector wants

to reach and describe data pack but will be received by that QoS There aresome prior keeping types are supplied by RSVP A host receives data fromsome sources that can set forth on prior keeping requirement to distributeseparate communication band to each source It maybe one communicationband is shared to every user in case of on-line discussion often only has 01person talk at a timing date

3.1.4 Conclusion:

This chapter has mentioned main issues in VoIP technology To have atalk network in IP complete, it needs to have standards of multi-meanstelecommunication, it will mention in next chapter

4 Introduction of standards:

4.1 Introduction of standards:

For standard of multi-means telecommunication bring pack baseincluding both VOIP and standards related to telecommunication.International telecommunication organization (ITU-T): set forthrecommendations H323 H2323 is complicated protocol and not ensure goodquality of service (QoS)

The technical expert force of Internet (IETF) has set forth 2 standard ofsimple protocol standard more than SID and MGCP and H.248 of ITU willprovide quality more ensure and more flexible MGCP and H248 will provide

to H323 and STP

The relationship between reference protocolls OSI with functions andprotocol in VoIP

Figure 8: Relationship between OSI protocoll and protocols in VOIP

4.2 Standard H323:

4.2.1 Introduction on H323:

Standard H323 is a heart technology determines compositions, protocoland procedure, to provide to real time multimedia telecommunication servicesuch as talk, image, data through pack switch network which relies on IPprotocols H323 can apply for multi point telecommunication and provide manykinds of service, so, it may be applied for many fields

4.2.2 H323 elements:

Standard H323 includes 4 kinds of fixed elements, when they worktogether in network they will provide multi media telecommunicationservice from point to point, and point to multi points In the terminal,Gateways, Gatekeepers, multi point controller unit (MCU) A Proxy H323 isthe fifth composition may in the rate with protocol activity

Layer order Name of class

OSI Functions and protocol in VOIP

Figure 9: Structure H323

+ Terminal: Using to multi-media telecommunication two-way-real time Theterminal can an individual computer or independent equipment are usingH323 All terminal equipment must implement the transmission of sound tofundamental service provided by the terminal H323 being talktelecommunication All terminal equipment must by provided H245, Q931,RAS, KTP, H245 with capacity to control means flow, H225 (origin fromQ931) use to control the signal of call, establishing and deleting a call RAS

is used at the terminal to register/permit/state, to be a protocol used to contact

to gatekeeper RTP / used as a protocol transferring means for bringing talkflow The compositions not oblige of terminal H323 are video pressedstandards T120, conference data protocols and MCU properties

+ Gateway A gateway connects to 2 different networks Gateway H323 providethe connection between a network H323 and a network that not be H323 (forexample PSTN) For implementing the connection between two differentnetworks, it need to convert protocol for the establishment and deletion ofcall, converting information and means through gateway No need to usegateway to the telecommunication between 2 terminal in H323 network.Gateway includes both hardware and software aiming to implement the duty

of code and decode the talk signal and sign to packing talk signal into IPpack, to establish calls through gatekeeper, to treat connections with PSTNnetwork

Main functions of gateway:

- Communication PSTN: Provide many different protocols, to meet referenceconnection of PSTN network

- Communication ID: Provide communications with ID network through partialnetwork and wide network

- Treating talk signal, including function of pressing talk signal under manystandards such as G729, G729A, and then to pack signal into ID pack

- Sign: Only use sign R2, SS7 to connect PSTN and IP gateway

- Destroying echo: Provide the function of destroying echo under standardG165 of ITU

+ Gatekeeper: Implementing the control and fixing line of call, being a veryimportant part of telephone network ID, it play a role to supervise and controlall activities of network Gatekeeper must have ability to provide almostfunction of PSTN In additional it must also have high rank functions on thebasis of characteristics of network IP functions

- Converting address: Is ability to move digital of public telephone networkand separate network to addresses ID and verso versa

- Managing network: Managing, supervising the state and parameters ofnetwork compositions To allow executive person setting forth adjustments

on time

- Controlling the access: No permit machines that no register to use naturalsources of system

- Treating the call: Treating the calls between subscribers and gateways such

as establishing, connecting, deleting connecting, notifying to move the call

- Data base: Archives information of each subscriber and systemconfiguration

- Mobil describer: To allow describers to connect and use network from anypoint on the network IP

- Counting fee: From data base to count fee for each subscriber at the sametime to provide information on fee and service to customer

+ Elements of control and treatment of polypoint MCU A MCU will providethe capacity to keep multi media dialogue with many components participated

in MCU is the combination 2 basic system components to permit polypointtelecommunication Multi control (MC) and treating multi point (MP)

- MC provides the control of means flow such as dialogue code andestablishing sessions of transmitting signal real time, multi-transmittingand feedback, single transmitting-feedback through signal H245 when aconnection (terminal or gateway) taking part in dialogue, it must establish

a connection H245 with MC

- MD sending and receiving protocol flow (for example: talk form in PTDParks) arrive and members who take part in the dialogue MD may convertprotocol between different identifications, with ability to combine differentprotocols (such as mixing sound from many sources)

+ H323 proxy: To be established to the protocol H323 H323 proxy also act asother normal proxy, it’s often placed at firewall and management as well asdisplay all words called H323 between partial network and Internet Proxyalso ensure that only have connect void H323 it will be passed firewall Proxyoperate at layer of application and control package between 2telecommunication applications Proxy may determine the aim of call andimplement the connection requested Proxy may manage less with RSVP.Proxy H323 must satisfy requirements of gateway H323 and point outinterfaces with the functions that have the presence of gateway

4.2.3 H323 structure:

This structure uses UDP unreliable telecom protocol like audio, video andregistering packs Reliable protocol but later than TCD is used to data andcontrol packs in call signal, protocol T120 uses to conference data

Data Signal and control the call Audio and

Figure 10: Structure of protocol H323 4.2.6 Signal and control system in H323:

H323 provides 3 main protocols on control Call signal H225/Q931, H325/RAS and control protocol H225/Q931 is used to sign a call H225/RAS uses toestablish the call from place that sends to the host received after the call isestablished, H245 will be used to read just protocol flow

+ H225/RAS implementing functions like registration, to allow service, changethe wide degree of frequency band, notifying activity state between terminalsand gatekeeper, using protocol of unreliable transfer UDP

In LAN network without gatekeeper, RAS signal channel, not exist RASsignal channel In LAN network, with gatekeeper, RAS signal channel will beestablished between a terminal and gatekeeper

+ Call signal H225, this channel uses to bring control newscast H225, using toestablish the connection between 2 terminals H323 This signal channel isindependent to RAS signal channel and H245 control channel In a systemwithout gatekeeper, call signal channel is established between 2 terminalstaking part in the call And in system with gatekeeper, call signal channel will

be established between terminals and gatekeeper or between 2 terminal ofeach other The selection of project on establishing signal channel is depend

on the decision of gatekeeper

+ Control on protocol H245 After establishing the call, system of using H245protocol control measure, to exchange the capacity, on, off logic channel,requesting on priority regime, controlling flow, ordering and instructing This protocol is also used to implement the function like to decide masta/slave which aims to avoid the conflict occurred when 2 terminals implementsimultaneously the same things but only have a thing permits taking place at atiming date

4.2.5 Establishing the call in H323

Figure 11: Treating the call in H323

1 Connecting end H323 registering with gatekeeper

2 When the user picking up the receiver and dial need to contact, thisrequirement will be sent to gatekeeper through RAS newscast

3 If accepted, the call, the gatekeeper will answer IP address of describer/callerand send information band requested to the call

4 Starting the call by sending newscast that establishes the call to cable sidethrough newscast H225

5 Describer/cable receives information with the call come, the terminal H323(telephone) ring a bell

6 The two parties discuss owing to the capacity of control channel H245, whichaims to ensure the signs transmitted are the signs but the terminal receivedwill have capacity to solve

7.8 RSVP requirement will be sent to cable, later on RTP is opened between 2sides, the call is established

In figure 11, two terminals (H.323 end points) need to establish a VoIPcall between them, and different gatekeepers control the two terminals As afirst step, the calling terminal requests permission from its gatekeeper to

establish the call This is done with the Admission Request (ARQ) message.

The terminal indicates the type of call in question (two-partyor multi-party),the endpoint’s own identifier, a call identifier (a unique string), a callreference value (an integer value also used in call signaling messages for thesame call), and information regarding the other party or parties to participate

in the call The information regarding other parties to the call includes one ormore aliases and/or signaling addresses One of the most importantmandatory parametes in the ARQ is the bandwirth parameter This specifiesthe mount of bandwidth required in units of 100 bps

Note that the endpoint should request the total media stream bandwidthneeded, excluding overhead Thus, if a two-party call is needed, with eachparty sending voice at 64 Kbps, then the bandwidth rerquired is 128 Kbps,and the valuecarried in the bandwidth parameter is 1280 The purposeof thebandwidth parameter is to enable the gatekeeper to reserve resources for thecall

The gate keeper indicates a successful admission by responding to the

endpoint with an Admission Confirm(ACF) message This includes many of

the same parameters that are included in the ARQ The difference is thatwhen a given parameter is used in the ARQ, it is simply a request from theendpoint, whereas a given parameter value in the ACF is a firm order fromthe gatekeeper For example, the ACF includes the bandwidth parameter,which may be a lower value than that requested in the ARQ, in which case

the endpoint must stay within the bandwidth limitations imposed by thegatekeeper.

Another parameter of particular interest in both the ARQ and the ACF is

the call Model parameter, which is optional in the ARQ and mandatory in the

ACF In the ARQ call Model indicates whether the endpoint wants to sendcall signaling directly to the other party, or prefers that call signaling bepassed via in the gatekeeper.In the ACF, it represents the gatekeeper’sdecision as to whether call signaling is to pass via the gatekeeper or directlybetween the terminals In the example of figures 11, the calling gatekeeperhas choosen not to be in the path of tha call signaling

The Setup message is the first call signaling message sent from

one-terminal to the other to establish the call The message must contain the Q.931Protocol Discriminator, a Call Reference Setup, a Bearer Capability , and theUser-User information element Although the Bearer Capability informationelement is mandatory, the concept of a bearer, as used in the circuit switchedworld , does not map very well to an IP network For example, no B-channelexists in IP and the actual agreement between endpoints regarding thebandwidth requirements is done as part of H.245 signaling, where RTPinformation such as the payload type is exchanged Consequently, many of thefields in the Bearer Capability information element, as defined in Q.931, arenot, used in H.225.0 Of those fiejds that are used in H.225.0, many are usedonly when the call has originated from outside the H.323 network and has beenreceived at a gateway, where the gateway performs a mapping from thesignaling received to the appropriate H.225.0 messages

A nember of parameters are include within the mandatory User-to-Userinformation element Those include the call identifier, the call type, aconference identifier, and information about the originating endpoint Amongthe optional parameters, we may find a source alias, a destination alias, anH.225.0 address The User-to-User information element is included in allH.225.0 call signaling messages It is the inclusion of this informationelement that enables Q.931 messages, originallydesigned for ISDN, to beadapted for use with H.323

The Call Proceeding message may optionally be sent by the recipient

of a Seup message to indicate that the Setup message has been received andthat call establishment procedures are underway When sent, it ususlly

precedes the Alerting message, which indicates that the called device is

“ringing” Strietly speaking, the Alerting message is optional

In addition to Call Proceeding and Alert, we may also find the optionalProgress message(not shown) Ultimately, when the called party answersthecalled terminal returns a connect message Although some of the messagefrom the called party to the calling party, such as Call Proceeding andAlerting, are optional, the connect message must be sent if the call is to becompleted The User-to -User information elementcontains the same set ofparameters as defined for the Call Proceeding, Progress, and Alert message ,with the addition of the Conference Identifier These parameters are also used

in a Setup message and their use in the Connect message is to correlate thisconference with that indicated in a Setup Any H.245 address sent in aConnect message should match that sent in any earlier Call Proceeding,Alerting, or Progress message In fact, the called terminal must include atleast an H.245 signaling address to which H.245 message must be sentbecause H.245 message are used to establish the media (that is voice) flowbetween the parties

In the example of figure 11, H.245 message exchange begins after theConnect message is returned This message exchange could In fact, occurearlier than the Connect message It is important to note that H.245 is notresponsible for carrying the actual media For example, there is no such thing

as an H.245 packet containing asample of coded voice That is the fob ofRTP Instead, H.245 is a control protocol that message the establishment andrelease of media sessions H.245 does this through messaging that enables theestabliment of logical channels, where a logical channel is a unidirectionalRTP stream from one party to the other

A logical channel is opened by sending an Open Logical Channel (OLC)

request message This message contains a mandatory parameter calledforward Logical Channel Parameters, which relates to the media to be sent inthe forward drection, that is, from the endpointissuing this command It

contains information such as the type of data to be sent, an RTP session ID,

an RTP payload type, and an indication as to whether silince suppression is to

be used If the recipient of the message wants to accept the media to be sent,then it will return an Open Logical Channel Ack message containing the samelogical channel number as received in the request and a transport address towhich the media stream should be sent

Strictly speaking, a logical channel is unidirectional Therefore, in order

to establish a two-way conversation, two logical channel must be opened-one

in each direction According to the description just presented, this requiresfour messages, which is rather cumbersome Consequently, H323 defines abidirectional logical channel This is means of establishing two logicalchannel, one in each direction, in a slightly more efficient manner Basically,

a bidirectional logical channel really means two logical channels that areassociated with each other The establishment of these two channels can beachieved with just three H.245 message rather than four In order to do so, theinitial OLC message not only contains information regarding the media thatthe calling endpoint wants to send, but it also contains reverse logical channelparameters These indicate the type of media that the endpoint is willing toreceive and to where that media should be sent

Upon receipt of the request, the far endpoint may send an Opne LogicalChannel Ack message containing the same logical channel number for theforward logical chanel, a logical channel number for the reverse logicalchannel, and descriptions related to the media formats that it iswilling tosend These media formats should be chosen from the optionsoriginallyreceived in the request, thereby ensuring that the called and willonly send media that the calling end supports

Upon receipt of the Open Logical Channel Ack, the originatingendppoint responds with an Open Logical Channel Confirm message toindicate that all is well.RTP stream and RTCP message can now flow in eachdirection