13. Reference

Reference:

• A systematic Approach to Basics of Voice over IP : Voice over IP fundamental by Cisco Systems. Last updated : 2001
  
• Introuction to networking by Person Publication
• www.wikipedia.com on VoIP.
• searchunfiedocmumication.techtarget.com

12. Advantage and Disadvantage of VoIP

Advantage of VoIP

• Packet switching allowed several telephone calls to occupy the amount of space occupied that can be done so using circuit switching.
• It save on cost
• An IP infrastructure needs fewer adds, moves, and changes than a traditional voice or data network.
• It helps employess in getting instant access to the data irrespective of the location.
• Customers are also able to deploy network with less upfront costs or with extensive in-house expertise, so it is preferred not only by business but also by an individual.
  
• Easy to understand and operate.
  
• Also it helps in making long distance calls for free and keeps a track of all the calls, which are coming, and outgoing even recording the minutes.
  
  
  
  Disadvantage of VoIP

• VoIP is dependant on wall power. So if there is no power, means the phone cannot work and stable power source is needed for VoIP.

• VoIP is susceptible to worms, viruses and hacking.
• Another issue is that VoIP is a system that dependant on individual PCs of varying specification and power.
• Quality loss will become immediately evident
• Phone conversations can become distorted, garbled or lost because of transmission errors.
• In VoIP, all phone calls are subject to the limitations of normal computer issues.

11. VoIP network problem

1. Delay/lantency

• Defined as number of times it takes for speech to transmit from the sender to the receiver.
• There are 3 types of delay:Propogation delay, serialization delay and handling delay.
• Propagation delay is caused by spped of light in fiber or copper based network.
• Handlinh delay defines many different causes if dalay(actual packetization, compression and packet switching). It is usually caused by devices that forward frames through network.
• Sertization delay is the amount of time that it takes to place sa bit or byte onto an interface. It's influence to delay is relatively minimal.
• Delay can be controlled by assigning a higher priority for voice packets in VoIP.

2. Jitter

• Define as the variation of packets interarrival time.
• one of the issuse that exits only in packet basred network.
• In a VoIP environment, sender send voice packets at a regular interval( For e.g., sender send voice packet about 20ms).
• In order to send it in order, these packets can be delay and transmit them at the receiving end.
• Jitter can also be know as the differences between the packet expected to and when it actually received.

3. Packet loss

• It is the most common in data network.
• It has a direct result of the quencing scheme used in routers.
• The solution in reducing packet loss is to implement priority queuing, weighed fair queuing, or class-based wieghted fair queuing,whereby traffic amount are assigned to differrent to classes of data traffic.
• But, many protocols used packet loss to reduced number of packets they are sending.
• In VoIP, it is implement to control amount of packet loss in the network.

10. Voice Protocol used in VoIP

Voice Protocol in VoIP

There are two types of protocol in VoIP:

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

Real-time Transport Protocol(RTP)

• Provides the functinality to real time application
• Runs on IP over UDP
• has the capability of providing application framing by adding application-layers to datagrams.
• It uses the sequence information to detemine whether packets are arrive in order.
• It also provide time stamping to determine the packet interarrival time.
• RTP uses typically informs a source about the quality of delivery. The source then adapt its sending rate accordingly.

Types of relays used by RTP

There are two types of relay used for transmission - mixer relay and translator.

Mixer relay

• Combines data from two or more entries into single stream of data.
• It can either retain or change the data format
• Provides timming information for the combined stream of data and act as a source of timming synchronization.
• Typically used for combining audio stream in real time application and service system that is not able to handle multiple stream.

Translator

• A device that generate one or more RTP packrts for each incoming RTP packets.
• Format of outing packets may be different from the incoimg packets.
• Used in video application in which high quality siganl is converted to a lower quality in order to service receivers that support a lower data rate.
• Some time, translator can be used to transfer a multicast packet to multiple destination and the RTP entries separated by an application-layer firewall.

RTP Packet Header

The following show the RTP header :

The header fields of the RTP packet are describe below:

• Version - to indicate the version of the protocol
• Padding - Indicate the existence of a padding field at the end of the payload. It is needed in application the need the payload to be multiple of some length.
• Extension - indicate the use of an extension header for RTP
• Constributing source count (CSC) - a 4 bit field that indicate the number of contributing source identifies.
• Marker - boundaries in a stream of data traffic.
• Paylaod - indicate the specfic type of RTP payload. Alos contain information about the use of compression or encryption.
• Sequence - 16 bit field that indicates that a sender uses to identify a particular packet withihn a sequence of packets
• Time stamp - to enable the receiver to rearrange back the original data.
• Synchronization Source Identifier - randomly generated field to indicate the RTP source in an RTP session.
• Contributing Source identifier - an optional field to show the contributing source for the data.

Real-time Transport Control Protocol

• Runs on top of UDP.
• Perform multicasting to provide feedback about data quality to all the members
• The members can set an estimate of the perfromances of other members in the current active session.
• Sender can send about data rates, quality of data transmission.
• Receiver can send about packets loss rate, jitter variations and other problems encountered.
• All session members must be able to evaluate the performances of other session members.

RTCP packet limitation

• It msut be limited to less than 5%.
• If it is more than that, it would affect the voice quality and the bandwidth of the channel.
• The number of active members increased, the transmission rate of RTCP packets must be reduced

Format of RTCP packet

• Sender Report
• Receiver Report
• SEDS Source Description
• BYE
• APP - Application Specfic types

What are the reports send in RTCP session?

• There are 3 reports- sender report , receiver report and sources description (SEDS)

A receiver and sender report consists of:

• NTP timestamp - 64 bit field indicates to when a sender report was send.
• RTP tiemstamp - 32 bit field used by receiver to sequence RTP packets
• Sender packet count - 32 bit field represent number of RTP ocetets transmitted bt sender in the session.
• Sender byte count - 32 bit feild represent number of RTP octets transmitted by the sender in the current session.

The SR packets includes 0 or more RR blocks. One receiver block is included for each sender from which the members has received during the session.

9. Components of VoIP netwok and Voice Protocol

Components of VoIP network

The main components of a VoIP network are describe below

• Media Gateway
• Media Gateway/Signalling Controllers
• IP network
• IP Phone
  

Media Gateway
They are responsible for setting up call, detect call, analog to digital conversion and creation of voice packets. It also perform compression,echo cancellation and statistics gathering.

It also forms the interface that the voice content users so that it can be transported over the IP network. Each call is a single IP session transported using RTP. Trunking gateways that interface between the telephone network and a VoIP network. Such gateways typically manage a large number of digital circuits.

Types of media gateway

Residential Gateway provide a traditional analog interface to a VoIP network. Examples of include cable modem/cable set-top boxes, xDSL devices, and broadband wireless devices.

Business Media Gateway provide a traditional digital PBX interface or an integrated soft PBX interface to a VoIP network.Network access servers that can attach a modem to a telephone circuit and provide data access to the Internet.

Media gateway/Signalling Controllers

It houses the signalling controls services that control the media gateway controllers. It can be consider to that of H.323 gateway. It has the responsibility for some or all of the call signalling coordination, phone number translations, host lookup, resource management and signalling the gateway services to PSTN.The amount of functionality is based on the particular VoIP enabling products used.

The services of these devices are defined by the protocols and software they are running. There are several protocols and implementations that any number of vendors could deploy. Knowing the details of how the devices their protocols stack is important to designing the IP backbone that is to service the VoIP parts.

IP network

The IP infrastructure ensure smooth delivery of the voice and signaling packets to the VoIP elements. Due to their dissimilarities, the IP network must treat voice and data traffic differently. If an IP network is to carry both voice and data traffic, it must be able to prioritize the different traffic types.
There are several correlations to the VoIP and circuit-switching components, however there are many differences. One is in the transport of the resulting voice traffic. Circuit-switching telecommunications can be best classified as a TDM network that dedicates channels, reserving bandwidth as it is needed out of the trunk links interconnecting the switches.

IP phone

IP phone is a device that is used to receive voice packets and convert it to analog signal to the handset. A typical IP Phone perform the digitization, comperssion and packetization of analog signal . It uses the RJ 45 connector instead of the normal telephone connector. It is connected to the gateway via a port and signal passes through gateway to the network. At the same time, user can configure the IP address and that will enable communication to the distant end.

The following shows an example of the VoIP network:

8. Codec - Coder/Decoder

What is the function of Codec?

• Voice transmission is in analog signal, whereas the data network is digital.
  
• So codec is help to convert analog signal into digital form and transport it to the network. There are many standards to sample an analog signal and the process is done through PCM(Pulse Code Modulation or variations.
• How the PCM works? Firstly the analog signal is input to a low pass filter. After that, the filter signal is sampled using one of the codec. Next the waveform is converted into a discrete digital signal. This sample is than represented by a code that indicate the amplitude of the waveform at the instant the sample was taken.

The following show an example of PCM:


Types of codec

• G.711 - It is a 64kps PCM voice encoding method that encode voice that is ready in PCM format for digital voice delivery in the public network or through PBX.
• G.726 - Describe ADPCM coding at 40,32,24,and 16kpbs. It allow user to interchange ADPCM voice between packet voice and public phone or PBX.
• G.728 - A 16kpbs low-delay variation of CELP voice compression.
• G.729 - A CELP compression that enale voice to be coded into 8kpbs streams. There are two types of variation of this standard(A and B) that are differ in computational complexity and both also provide good speech quality.
• G.723.1 - It compresses speech or other audio signal components of multimedia service at low bit rate, as part of the overall H.324 Standards. There are two bit rates associated with this coder : 5.3 and 6.3kpbs.
  

7. Media Gateway Control Protocol (MGCP)

What is MGCP?

• It is protocol used within a VoIP system

• It was developed to address the demands of carrier-based IP telephone network.
• Also a complementary protocol for H.323 and SIP, which was designed as an internal protocol between Media Gateway Controller and Media Gateway.

What does MGCP consists of?

• It comprised o a Call Agent, one MG (media gateway) which performs the conversion of media signals between circuits and packets, and one SG (signaling gateway) when connected to the PSTN (Public Switched Telephone Network).
  
• MGCP is widely used between elements of a decomposed multimedia gateway. The gateway has a Call Agent, which is comprised of the call control "intelligence" and a media gateway boasting the media functions, for example conversion from TDM voice to Voice over IP.

MGCP Connection

MGCP connection can be point to point or point to multipoint. Once there is a connection, data transfer takes place between the endpoints. In multipoint connections, it is set up between endpoint and a multipoint session. In multipoint connection, it can be created over various types of bearer network.

The Architecture of MGCP

• The architecture does not involved in the frustrating work of encoding ,decoding and transfer of voice signals.
• The MGCP call agent works as a software switch for the VoIP network,.
• It does noting more than simply direct the media gateway and media signalling gateway which perform all the work.
• The main task of Call Agent is to implement numerous possibilities in the protocol.
• In every architecture, each and every components features a transaction ID, gets an anknowledgement and receives a responses.

The MGCP Packets

• MGCP packets are generally found wrapped in UDP port 2427.
  
• An MGCP packer can either be a command or a response.
  
• Commands start with a four-letter verb while "responses" start with a three number response code.

6. The Session Initiation Protocol(SIP)

Introduction to SIP

SIP is an application layer signalling protocol used to establish, maintained and terminate multimedia session. It can also perform both unicast and multicast session and sipport user mobility. It can also handles signals and identifies user location , call setup , termination and busy signal.

Since 2007, SIP has become the defacto signalling standard for VoIP and multimedia communication. It has been said that SIP has been benefited by being an IETF standard, The IETF being quicker to adapt to industry forces than the ITU. SIP is supported by both standard committees, as in H.323.

SIP is a text based protocol that is part of an overall IETF multimedia architecture. The IETF also include Resource Reservation Protocol, Real-time Transport Protocol , RTSP and SAP, SPP. it in-coperate in conjunction with other signalling Protocol such as H.323.

SIP Messages

There are two kinds of SIP messages : request from client and responses returned from servers.
Every messages contain a header that describe the details of the link. It is also able to send message syntax and header fields identical to Hypertext Transfer Protocol(HTTP). Besides that, SIP messages can sent either over UDP or TCP with multiple messages carried in a single TCP connection or UDP datagram.

What are the Message header in SIP?

There are four groups of message header which are described below:

• General headers- Apply to request and responses
• Entity headers - Define information about the content in the body and the length
  
• Request headers - Enable user to include additional request information.
• Response headers - Enable server to include additional responses information.

What are the Message request in SIP?

There are 6 kind of message request. Theses are referred to as methods, enable user agents and network servers to locate, invite and handle calls.The six SIP message request are describe below:

1. Invite
This is used to indicate that user or service is invited to join in a session. It include session description, for two-way call, the calling party indicates the media type.The user can recognize the capabilities of the other end and open a conversation session with a limited number of messages and round trips.

2. ACK
It is a correspond to an Invite request. It also represent the final confirmation from the end system and conclude the traction initiated by the invite command. If the calling party includes a session description in the ACK request, no additional parameters are used in the session. If it is absent, the session parameters in Invite request are used as the default.

3.Options
Used to enable user to query and collect user agents and network server capabilities. However this is not used to establish session.

4.Bye
Used by calling and called parties to release a call. Before ending the call, the user agent sens this request to server to indicate that the desired to end this session.

5.Cancel
Enables user agent and server to cancel any progress request. It does not affect complete request in which the final response were already received.

6.Register
This is used by the client to register location information with SIP servers.

SIP Signalling Model

The following show the signalling model between 2 user:


The following diagram can be sumarized above:

• Firstly, the caller send a signal to establish using the INVITE message to the PBX of the callee.
• The invite message contain information such as from, to via and call-id,in addition to routing information.
• Next the proxy server will communicate with the server at the callee side.
• Once the callee received the message, it will then send a TRYING message to the caller from the proxy server, indicate the signal is being routed.It is also used to keep track of the call process.
• At the same time , a ringing signal is send form the callee to caller.
• When the callee accept the call, an OK signal is send to the caller to indicate that the called party has accept the call.
• The last signal is anknowledge by an ACK messge without any response.
• After that, both user can communicate directly through the media session.
• When the callee want to end the call, it will send a BYE message to terminate the session.
• After that, the caller will send a OK message and the call is terminate.

Components In SIP

The components in SIP are:
1. User Agent
2.Back-to-Back User Agent
3.Proxy server
4.Redirect Server
5.Registrar server

1. User Agent
They are endpoint devices that initiate and terminate the session through a series of request/response quries. It can be an IP Phone and each devices can dind each other without the need of other entities.

2.Back-to-Back User Agent
It is an application that acts an intermediary between two endpoint. It also maintained the state of the call and responsible for call termination. It may function as a gateway representing as an endpoint on the IP side to an endpoint on the PSTN.

3.Proxy server
Translate request by passing hem to other SIP server by rewrite the message before passing it on if required. It can perform address translation within the domain by resolving the e-mail URL or telephone to IP address and use the DNS to find the SIP server outside the domain

It helps in call setup and termination, leaving call connect to the endpoints themselves.

4.Redirect Server
Maps a request from client to the URL of the party being called and send it back to the caller.
It does not pass request to other server.

5. Registrar Server
Register users into a databases as they come online. Information indicating the user identity and the devices on which they choose to be researched is stored by IP address, phone number or URL. Enable user to update their location.

Both Registrar and Redirect server are often resides on a SIP Proxy server.