Question

(terza/prima) Question: You are required to design a VoIP application. Discuss the requirements for such an application in terms of its protocols and performance requirements. What types of protocols you would need to design a complete VoIP application (Hint: real time streaming protocols and signaling protocols, etc.)?

PLEASE write the answer in your own words! cheers

Answer 1

Given :

You are required to design a VoIP application. Discuss the requirements for such an application in terms of its protocols and performance requirements. What types of protocols you would need to design a complete VoIP application (Hint: real time streaming protocols and signaling protocols, etc.)?

Introduction :

Voice over Internet protocol (VoIP) is a new way of communicating. It is a technology that allows users to make telephone calls over an IP network. This paper will describe Voice over Internet Protocol (VoIP) to a level that allows business concerns of implementing VoIP, components of a VoIP system. The business concerns will be those that affect Quality of Service (QoS). VoIP components will include end-user equipment, network components, call processors, gateways and two of the more common architectures: Session Initiation Protocol (SIP),This paper gives a brief introduction of VoIP technology: the network structure, protocols, echo and delay, jitter, and packetloss in VoIP network. Finally, the survey concludes with a discussion on the feasibility of providing VoIP over challenging satellite links.

Requirements :

While making phone calls over internet, signalling protocol plays essential role since it empower the network components to communicate with each other, hence set up and tear down calls. For IP telephony, a call can be prescribe as the multimedia session between multiple participants, while on the other hand signalling conjoined with a call is referred to as a connection. Key roles of a signalling protocol can be divided into four functions:

1)Session establishment: The callee decides, if to accept, reject or redirect the call. User location: The caller first has to find the location of the callee.

2)Call participant management: It allows endpoints to join or leave an existing session.

3)Session negotiation: The endpoints involved in the call should concur upon a set of properties for the session.

Protocols :

a)The H.323 Protocol Stack:

H.323 suite comprising of a set of standards. G.711 (64 kbps channel) is the minimum requirement for audio applications. Various voice codec standards illustrated by H.323 are G.723 (5.3 and 6.3 kbps channels), G.728 (16 kbps channel), G.722 (48, 56, and 64 kbps channels), G.729 (8 kbps channel) [4]. The control protocol H.245 for multimedia communication is used during an initial deal among the machines to find the terminal capabilities, audio encoding algorithm and media channels. The Real-Time Control Protocol (RTCP) has a function to provide quality of the sessions and connections and also feedback information across the communication parties. The support and data packets can operate over UDP or TCP [4].

b) Session Initiation Protocol (SIP):

SIP (Session initiation protocol) is a communication protocol used for signaling and controlling multimedia communication sessions such as online gaming, instant messaging and various services. It is similar to web protocol HTTP since messages comprises of headers and a message body. SIP generally uses port 5060 as its default protocol for either TCP or UDP. SIP can be interpreted as the authorize protocol for voice, telephony and video over IP (VoIP) services. Network Elements Server network elements are defined by SIP. Though multiple SIP endpoints can communicate without any involvement of SIP infrastructure but this commence is often not practical for a general service. The main network elements involved in the SIP communication can be illustrated as follows: Proxy server It is a mediator entity which reacts as server (UAS) as well as client (UAC) for raising requests on behalf of various clients. It also performs routing to transmit the job assigned to another entity next to the targeted user. User Agent The User Agent (UA) is used in generating or receiving SIP messages. It can also act as User Agent Client (UAC) for transmitting SIP messages and the receiver will act as a User Agent Server (UAS). Elements of SIP network sometimes save this information, since it can be used in identifying SIP compatibility problems. Redirect server It allows proxy servers to connect SIP session invitations to external domains.

c) RTP – (Real Time Protocol):

RTP is drafted for all over real-time stream data transfer. It enables data transfer to multiple destinations through IP multicast and treated as primary standard for audio/video transport within IP networks. Generally RTP is used in alliance with a signalling protocol that assists in build up connections across the network.The RTP protocol is relevant for audio and video streaming.Two RTP sessions establishes for video streaming each with different SSRC identifiers out of which one is used for audio transmission whereas another for video transmission. Also, there is downside of RTP that it neither assure delivery of packets nor Quality of Service (QoS).

d)RTCP – (Real Time Control Protocol):

RTCP stands for Real Time Control Transport Protocol and is defined as a protocol that works with real-time protol to observe delivery of data over large multicast network. RTCP can monitors the fraction lost, jitter, packet loss and one way delay. The basic functionality and structure of packet is defined in RFC 3550. One of the major drawbacks in the RTCP does not report the late arrival of packets. This has been overcome in an improved version of RTCP-XR (Real Time Control Protocol Extended Reports).