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: