Developping service voip in Viet Nam

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Ha Noi open university
Center For International training Co-operation
Thesis:
Teacher : Nguyễn Thái Nguyên
Group 3 : Đồng Xuân Thắng -Cap
Lê Trọng Nghĩa
Nguyễn Xuân T
Mai Trọng Dũng
Bùi Thanh Nhàn
Ngô Thị Nhàn
Hà Nội ngày 15/1/2003
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Glossary
ATM : 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
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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 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 Cisco
’
s 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)
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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 and eliminating
the ineffectiveness of time-consume on connecting. In the channel shifting
network, the user is provided a conductive channel to exchange information

together. When the exchange completed, the conductive channel is released. This
could lead to loss because of limits of conductive channel. The utility 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 designed optimum for real
transmission time with high service quality. In the channel switching network, all
terminal equipment and switch board are inserted a fixed number so no need to
enter address for information exchanging process. The switching system in
channel switching network will base on the address of called subscriber to define
the conductive line. Because the band width is ensured 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 that are
not exchanged according to the real time; Package delay doesn’t effect strongly
on service quality like email and file transmission. Package switching networks
don’t set aside a fixed line between two users, so, not ensure the service quality.
All information on the network are divided into packages, these packages contain
the destination address and its order.
Channel fixer and host on the network will send these packages to the
targeted address. On Internet, all packages are treated the same with out
distinguishing their contents. When packages to the destination address, they will
be arranged according to the initial number. By form of package information
transmission, the utility is maximum. However, real time applications will be
greatly effected on service quality. The fee is not calculated on distance or time
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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 terminal stations. Channel
fixing will be controlled by the IP destination address. To create a
understandable, convenient address type for the IP address by name like service

of regional name or email address.
Because the limit of IP address, the users are temporarily inserted IP while
dialing. The IP address is only for one terminal equipment while connecting
Internet and deleted while not connecting. The deleted IP address will be used for
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. But with
VoIP, the time for call establishment is independent to subscriber’s fee. One
subscriber could have calls to different ones and exchange data, dialogue, pictures,
paintings and video with other subscribers.

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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 and
Internet, we see that it is typical to accumulate real time signal into the package
switching 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 transmission
equipment.
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 PC
should have 1 PC that is equipped audio, a software and connected to Internet.
This service no need gateway and PCTN because PCTN never switch these
calls, the main transmission tool is public Internet. Due to sound quality and
complexity of use, the PC to PC has a litter affect on traditional telephone

service.
+ PC to Phone expands the number of users but for exploiters, the call of PC to
Phone is more complex than that of PC to PC.
+ Phone to Phone is very important market including mainly commercial
services, because, people prefer to communicate by phones. However, the 3
rd
model of IP requires more investment capital because it needs input gateway to
PSTN near places providing service. Services of Phone to Phone are nearly
similar 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 voice
transmission, fax and services relating to package switching networks on IP.
Internet phone and VoIP are basic types of IP. Internet phone is IP in which the
main transmission network is public Internet (global super-network).
Voice over IP is IP in which the main transmission network is private-used one
basing on IP.
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Besides, being the replacing tools for distance and international phone, the IP
technology 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 terminal
equipment 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, the
terminal systems can control the compressed volume on the network bandwidth
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 transmission
method. 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:
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The VoIP technology has potential for future development, ability to replace
the existing PSTN network. Due to differences in features of channel switching
network 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 for application
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 much
when at distance, to reduce expenses for distant calls, use public data network or
private data network for communication. The package switching network that
applies IP is example. Using the package switching network by IP to transmit the
talking signals. Voice over IP-VoIP is good basis to design global multi-
instrument transmission system that can replace the infrastructure of existing
network. Accumulating Audio, Video, data, fax... into a single common network
on IP technology. It is possible to apply the Frame relay or unsynchronous
transmission technology ATM to replace IP technology. The VoIP is more
economic for distant call, because the fee is calculated by the width 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 as that of PSTN.
The biggest difference when applying into the multi-instrument network is
actual 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 of talk
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 by Rule
A or U with speed of 64 Kps recorving sound rather actual. For some specific
applications such as transmitting talk signals on TP network, sounds are
transmitted with lower speed, so, there should have coding techniques and talk
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signal compression to lower speed according to standard of ITU and ETSI like
G723.1; G729; G729A; GSM.
+ Standard G7213. According to the standard of ITU, the coding has 5.3Kbps and
6.3Kbps. The compression technicque uses MP-MLQ for high bit speed; for
coding with low bit speed using ACELP. Delaying against algorithm is 67.5ms.
+ Standard G.729. According to the ITU standard, this coding has speed of 8
Kbps. This compression techniques uses algorithm predicting coded linear
linked structure algebra excitation. Delaying against algorithm is 25ms.
+ Standard GSM06.10. According to ETSI, this code has 13Kbps. This
compression 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 intensity
that is transmitted by automatically detecting the dead space on the talk and
stopping transmitting at that time. There are space approx. 50-60% of almost
talks. 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 of talk
signal frequency. The difficult of VAD is to define the exact time of talk ending

and of talk signal. The double VAD is nearly 200ms after recognizing talk signals
and stop and detect package processing. This top prevent VAD from missing 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 of
address: address in CSN and in IP.
a. Numbering on SCN network:
On the channel switching network, all terminal and switchboard are fixed a
number. Number E164 is telephone numbers subject to the structure and
numbering program that were described on the proposal E164 by International
Telecommunication Union. The line fixing process on the channel switching
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network 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 + Identification
Code + Subscriber number.
Code of Access Caller + International Post + Country Code + Destination Code
+ Subscriber.
Code of Access Caller + International Post + Country Code + Global
Subscriber’s Number.
b. Numbering on IP:
+ Prefix is an identifier including one or more numbers allowing the used
numerical types, network and service and can be used to select service
provider, 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 the selected
code of service provider. Incase, a lot of service providers on IP network,
Gatekeeper is able to identify and process the selected code of service provider.
To get the most common address types on Internet, it can use name address like
email address: user@domain , user@host , user@IP-address , phone-
number@gateway .
2.4. Fee:
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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 collecting
and reserving all detail of call from gateway or MGC. These data are used to
make invoices for customers. Customers ca access into the host for their fee
details on the website. The fee will be calculated by the used time. The fee
calculating system should be able to calculate on 2 types of service: pre-paid and
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 signal news
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.
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Although new signaling, ssuch as H.323 ans SIP, exist for VoIP net works
the standard in traditional telephony and in mobile networks is SS7. Therfore,
if a VoIP based network is to communicate with any traditions network, 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 inter
working 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. If
other call control entities use SS7 then the MGC must use SS7 at least to the
extent that the other call control entities can communicate freely with it. This
means that the MGC does not necessarily need to support the whole SS7- just
the 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 be
used 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.
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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
12
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Figure 3: SS7 Stack
SCCP provides an enhanced addressing mechanism to enable signaling
between entities even when those entities do not know each other’s signaling
addresses (known as point codes). This addressing is known as global title
addressing. Basically it is a means wherby some other address, such as a
telephone number, can be mapped to a point code, either at the node that
initiated the message or some other node between the originator and
destination of the message
Figure 3 provides some examples of communication between different

SS7 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 to peer
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 exists between
point codes 1 and 2 and between point codes 2 and 4.
At the TCAP and Application layers, a peer to peer relationship can only
take place between point codes 1and 4. In other works, the application at point
code 1 is only aware of the TCAP layer at point code 1 and application layeer
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 the corresponding
TCAP layer at point code 4. It is not aware of any of the MTP layer. Equally, if
we consider communication between point code 2 and point code 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 as the SCCP layers
are concerned, nothing else exists. Therefore, SCCP neither knows nor eares
that point code 3 exists. Consider Scenario C, where ppoint code 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 the lower SS7
layeers at all- just an MTP emulation over IP. Provided that the MTP 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
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layers at point code 1 and 2 do not care. Consequently, it is possible to
implement SS7 based applications at point code 4 without implementing the
whole SS7 stack. This is the concept behind the Sigtran protocol suite.

Point code 1 Point code 4
Scenario A - Communication Between Adjacent Signaling Points

Application Application
ICAP ICAP
Sccp sccp sccp
MTP3
mtp3
MTP3 MTP3
MTP2
mtp2
MTP2 MTP2
MTP1
mtp1
MTP1 MTP1
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ISUP
MIP3
MIP2
MIP1
ISUP
MIP3
MIP2
MIP1
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Point code 1 Point code 2 Point code 2 Point code 4
Scenario B - Communication Between non- Adjacent Signaling Points
Application Application
ICAP ICAP
sccp sccp sccp
MTP3
mtp3

MTP3
MTP MTP
MTP2
mtp2
MTP2
emulation emulation
MTP1
mtp1
MTP1
over IP over IP
Point code 1 Point code 2 Point code 2 Point code 4
Figure 4 Example SS7 Communication Scenarios

2.6. Confidence:
The IP service active on the base of IP switch, the requirement of confidence
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:
- Confirmation.
- Acceptance.
- Refuse.
- Privateness.
- Security.
An IP system can have one oral above services depending on each specific
case and even each specific subscribe. For network exploiers, protecting
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important information from illegal access is put on top. Below are some

suggestions:
+ Data Coding: This is the most effective practical method to protect information
that are transmitted through different networks. Regularly, the data is
compressed by different standards by Gateway, may be, no need to code the
data. If necessary, information on network are advised to code by DES (Data
Encryption Standard) with the key of minimum 56 bit long.
+ Anti-virus: Virus can cause significant consequences to the software of all
system. Virus could be spread from other system or customers’. This also
carries significant meaning when the system operates on base of the dispersion
processing structure. Anti-virus software should be installed on Gateways 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 allow
Firewall 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 and
confirming 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 be
cancelled.
- Connecting limit: Limiting on connecting times and starting points of
connect.
+ Confidence policy:
Confidence plan should include following elements:
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- 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:
+ Delay:
- Algorithms delay: This is caused by Codec and naturally - created by coding
algorithm.
- Package delay: This is necessary time to delivery a IP package. And also
suffer from delay when passing saving equipment and transition equipment,
for example, passing line fixer or switcher.
- Wave transmission delay: This is necessary time for optical or electric
signals 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 should move
from Gate to Door across server body. There is a minimum delay through
server body and changeable delay by in line and processing of line fixer.
+ Echo suppression.
The first trouble caused by the delay is echo impact. The echo can be
occurred on a talk network by chain-jointing between the listening and speaking
parts of the complex. This delay is called auscosic delay. This also occur when a
part of power energy is reflected to the speaker by a exotic line in PSTN, that
called 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 need
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echo suppression if one-way delay is smaller 25ms. However, the one-way delay
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 very
difficulty when the delay is too long causing voice superposition. This occurs when one
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, the
data network that badly transmit and can’t supply the fixed delay because
different packages have different delay, so, different delay frame. Resources
create regularly frames, the Destination gate can’t collect these frames regularly
because of Jitter. Jitter interrupts the call and difficult the talk content. To remove
the changeable delay, it should receive frames and keep them for enough time. So
that the latest frames come timely for reading in order. The buffer 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) late
for 12ms and 5ms against expected relatively.

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Figure 5: Jutter description
+ Package loss:
The IP network doesn’t ensure to fully and orderly distribute packages.
Package will be lost if blocking (be broken by transmission line or insufficiency
in capacity). Due to, the sensitive of talk transmission, the transmission rules are

based on TCP, it will be no effective. If talk sample is lost on the terminal talk,
ignore the gap at this line. If too much package are lost, the voice will be broken.
To cover, replaying previous packages. This is only done if some samples are lost.
In case of group faults, take interpolation by using previous packages and re-
coding set will product what lost package is. In reality, to apply IP network for
high service like video, mobile and high-quality talks, another signal 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. This
compression 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, for
compressing the RTP starter may reduce the IP starter to 2 bytes. The bandwidth
depends on byte coding speed and talk package size. The private IP network has
more 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 dead suppression.
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 some
unfavorable matters when using the TCP mode. Due to the reliability of leyte
service and retransmission of lost packages increasing the delay of network.
TCP has a lot of properties and complexity, this is not benefit for VOIP
technology. When transmitting talk signals, they should be distributed to users
at the same time. On TP network, there should have effectiveness for
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distributing multi transmit-feedback data, however, TCP can’t supply this. If

the data are distributed to destinations on TCP, single TCP will be required to
connect causing cost of bandwidth.
+ UDB is protocol simpler than TCP, just an expanded ID mode, only used when
no requirement for high quality service. This protocol has advantage that no
waste of time for re-transmission of lost packages.
It can use the property of multi-transmit and feedback and save bandwidth
when data sent to a lot of destination. UD Palso has disadvantages, no
synchronous mechanism and no means to control flow and block. To solve this
matter, 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 transfer protocol. When a
host desires to send a package, it should know transmission measure to make
package shape, add the specific transmission measure into the title of package to
pre-decide the RTP’s title and put into the lower layer transmission measure.
Then, send to network by multi transmit-feedback or single transmit-feedback
ways to other participants.
Format of RTR fields are described as follows:
P header
20 bytes
DP
8 bytes
TP header
12 bytes
CODEC sample
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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 more
Octet padding adding to the terminal that not belong to pay load. The final
Octet of padding includes number of ignored octet padding. Padding may need
more other coding algorithms with changeable sizes of block or bring some
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 supplies
information to re-create and release package in case of defining the first
package on released voice.
+ Payload (PT, 7bytes). Fixing the transmission measure (Editing, changing the
bandwidth to be sufficient for transmission on each travel). RTP and detailed
description.
+ Sequence number (16 bytes). Sequence number increases each value for each
data package sent by RTP, and search for lost packages and recover them in
order.
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+ Time stamp (32 bytes). Time Stamp feedback a sample for the first octet on
RTP 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 in one
RTP session.

+ CSRC list (0 - 15times, 32 bytes/field): CSRC list defines, show resources for
load (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 define
contributing resources.
+ RTR Header Extension (variable length). An optional extended mechanism is
supplied with RTP allow each implementation to test new functions requiring
more information on RTP Header.
3.1.2. Real Time Control Mode:
The RTCP is the basic to control continuous transmitted packages to
participants on communication session by using the same distribution mechanism
for data packages. The low modes must pill up data packages and control by
using different port number and UDP. Functions of RTCP are described 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 coding sets.
However, testing with IP multicast also give out results against transmission of
feedback from the receiving end to diagnosing distribution errors. Sending a
feedback to all supervising points to define problems, errors by local or central.
By distribution mechanisms like IP multicast, can do for each unity as service
providers and not be attracted into other aspects on communication sessions,
receive feedback and act like the 3
rd
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 receiving
points of C.Name keep way for terminal. Receiving points also require C.Name
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conjugate to data lines from each giving point in mutual relations on RTP
session, for example, audio, and video.
+ Two first functions require all points participating into communication session
send RTCP, so, the speed must be controlled by RTP to arrange a great number
of communication points. Each communication point can send information
package to other points, each point can supervise independently to others.
+ An optional function for a minimum post session, like fixed communication
points to display on user’s interface. It seems very suitable, useful on loosely
control sessions where communication points in and out don’t need member
controlling measures or negotiation parameters. RTCP serves as a useful
channel to reach to communication points. But it thinks that not necessary to
satisfy 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 with time
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 block for
each RTP. Each block describes a immediate coefficient and the fitter (like
phase drift) from this sources. The tensioning block shows the final label and
the delay from receiving sender’s report, allow resources estimate their
distances.
- SDES: Resources labeled packages for controlling session. It include C.Name,
the unique identification like frame an entail address. The standard name used
to resolve conflict in synchronized source value and deferent combined
communication protocol current is created by such user. SDES packages 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.

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- APP: Specific applying elements (APP) to add give further specific information
in to packages .
Identifying header park as follows:
0 8 16 31
V = 2 P RC PT Length
Figure 7: Preface of RTCP
- Version (2 bit) determining version of RTCP. At present it’s installed equal
2.
- Padding (P.1 bit) when being installing, it will determine RTCP information
package, including some octet added at end part of control information the
latest octet of padding part, will count, how many octet added are left.
Padding may be uses by some cipher algorithm with the sire of data block
changed. In permissible RTCP information package, padding can be
required on final information package because information packages will
couple complete code.
- Reception Report Court (RC). Volume of reception report Court will lump
all to RTCP package value is equal zero is legal. May have to 200 constant
determinations 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, including header
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 to the
application. The host transmit data that need to reach any QSS, it send the call
to destination address owning to newscasts include information on character
and flow. RSVP not is line-de fixing protocol, simple selects a most optimal
line. This can’t give an ideal QsS. RSVP is an important instrument to QsS,

but not resolve all necessary problems related to QsS. On the transmission to
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aim of RSVP allows router save information on transmission newscast, with
this way, its used when prior keeping data from newscast sender on the
transmission line. When the user receive, transmission line newscast, it can
decide to receive data or not of the sender with QsS fixed. For meeting QoS
fixed, RSVP will send periodically a requirement of prior keeping under prior
keeping line of transmission line newscast.
Precise prior keeping line attained owing to information in RSVP. The
prior keeping newscast includes 2 parts: containing QsS that collector wants
to reach and describe data pack but will be received by that QoS. There are
some prior keeping types are supplied by RSVP. A host receives data from
some sources that can set forth on prior keeping requirement to distribute
separate communication band to each source. It maybe one communication
band is shared to every user in case of on-line discussion often only has 01
person talk at a timing date.
3.1.4. Conclusion:
This chapter has mentioned main issues in VoIP technology. To have a
talk network in IP complete, it needs to have standards of multi-means
telecommunication, it will mention in next chapter.
4. Introduction of standards:
4.1. Introduction of standards:
For standard of multi-means telecommunication bring pack base
including both VOIP and standards related to telecommunication. International
telecommunication organization (ITU-T): set forth recommendations H323.
H2323 is complicated protocol and not ensure good quality of service (QoS).
The technical expert force of Internet (IETF) has set forth 2 standard of
simple protocol standard more than SID and MGCP and H.248 of ITU will
provide quality more ensure and more flexible MGCP and H248 will provide to

H323 and STP.
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