Telecommunications And Internet Technologies

Published Date: 02 Nov 2017

This paper will concentrate on a network element is often known as Session Border Controller and it has the relation of the IP multimedia standards and standard organizations. The Main goal of this thesis is, the functions of the session border controller when compared to the standards and figure out, What is SBC functions on standard manner and what is on not standard?

We characterize even which functions of SBCs carried out and why they are carried out? The methodology selected to compare the functions of SBCs to the determined functions in the standards.

A second goal of this study is to compare the views or thoughts that various standards organizations defining IP multimedia communications infrastructures have this sort of functionality that runs a session border controller.

Introduction

During the last few years the perspective of IMS as the paragon for an omnipresent QoS enabled network for multimedia services has developed. The present reality is a miscellaneous world incorporating both IMS and not IMS controlled network, with connects to a plurality of uncontrolled and legacy networks, and different access technologies and user demands.

Session Border Controllers (SBCs) are built up as a fundament of the multimedia network environs managing and accommodating signalling and media flows as they traverse network borders. In reaction, the IMS standards have extended to include a lot of the attributes that SBCs provide and the mobile, the different necessities of fixed line and Cable Companies.

Table of Contents

The Session Border Controller (SBC)

Session border controller is a non-standardised collection of functions. Rather the Session Border Controllers (SBCs) have designed to address the broad range of matters that may occur when voice and multimedia services are superimposed on IP infrastructure. These comprise security and avoidance of service misuse to guarantee Quality of Service (QoS), Policy for trafficking and billing purposes, protection the privacy of the bearer and user information, resolving power of VOIP protocol problems coming up from the expansive comprehensive use of firewalls and network address translation (NAT), and the variety of various protocols used in VoIP networks. These issues are used to access the bearer and enterprise networks, and on both User Network Interface (UNI) to end users and access networks, and Network Network Interface (NNI) to peer networks. The diagram below indicates where the function of SBC is distinctive required.

Signaling path border element (SBE)

Data path border element (DBE)

C:\Users\Jussif\Desktop\white_paper_c11-540690-2.jpg

Figure 1: NNI, UNI and SBC positions and where can be deployed in an enterprise. [1]

The graph draws one single device on the border of each network (conventional SBC), but there is really high flexibility how this function is dispersed. As for instance, a device on the access network would be able to execute incipient user authentication, a device on the edge of the network could impose policy accessibility to restrict Denial of Service (DoS) attack and preclude bandwidth stealing, core devices could restrict the entire usage for specific group of users and find distributed DoS attacks. The position of the individual function depends on the whole system design, as well as the availability of proceeding resources and the level of confidence between the various devices.

Session Border Controller Functions

Security

A vulnerable unsecured network cannot afford an assured QoS service or charge for its use, for the simple reason that an unauthorized users cannot be prohibited from misusing confined network resources. The SBCs can be responsible for security and protection by preventing such as following issues from being occurred, unauthorized access into the confidential network, inacceptable or malicious calls, together with Denial of Service (DoS) attacks, attempts of the authorized users to steal the bandwidth, unfamiliar network conditions, such as a serious emergency. The Bandwidth on access links and the capacity of processing on network servers are common resources that usually have need of protection. In the main, these resources can be prevented from becoming bottlenecks whereas the network links can be inexpensively provided. To achieve these security services, the SBC identifies and authenticates every single user and specifies the corresponding priority of particular call, stops any unauthorized access to the network traffic regardless whether media or signalling traffic, confines the use of resources and call rates in order to avoid overloading, approves the flow stream of the media and categorises and routes the data to guarantee appropriate desirable QoS. The QoS through the core of the network is typically managed by an accumulated classification mechanism, an example is Differentiated Services (DiffServ), since this eliminates the reserved bandwidth overhead for every single flow. The QoS in the access network may impose by guiding the endpoint to reserve required resources within the access network transversely or by signalling to the access routers, to perform that a SBC can also be used. Instead of that and by investigating the call signalling messages a sophisticated smart access network could autonomously specify appropriate QoS for the media streams.

Policing and Monitoring

Due to both regulatory reasons (wiretapping and QoS observing) and commercial reasons (billing and stealing detection) the monitoring of the network usage is essentially required.

The devices that perform the monitoring function has to fulfill two important basic requirements, they must have adequate intelligence to recognize the signaling and media protocols and have to be located in a position where all the signaling and the media flow through. SBCs accomplish all these requirements thereby all traffic must pass through a SBC to access the network.

Privacy Protection

In order to ensure maintaining privacy the protection of associated information is needed, the most confidential information are: the user information, that belong to a particular user who doesn’t want to be published and the core network information, which could deliver commercially sensitive information to contender or some particulars precise information that could be used to aid an attack. A SBC may also be used for removing sensitive and personal information from messages, in addition to the details of interior network topology as well as the signalling routes, earlier before leaving the core network. It can even hide the actual address of the user by serving as a connector for the signalling and media. [2]

1.2 VoIP Protocol Problems & Solutions

Due to the capability of SBCs devices to hide any dissimilarities between the access and the core networks protocols, can the SBCs serve as gateways for heterogeneous networks. Moreover SBCs may provide some additional functions just like voice transcoding (such as converting voice from G.711 to G.729A), but SBCs include a number of functions exclusive to the IP telephony world. Key among those functions are signaling conversion between H.323 and SIP or between IPv4 and IPv6 or even various versions of ITU VOIP protocol H.323. A SBC device is a typical solution to hide the network topology, including the complicated routings because of NATs and the firewalls and to provide interconnection for sessions between VPNs with overlapping addresses to both VPNs and private IP [2].

This chapter roughly presents the general functionality and technologies involved.

IP Multimedia Overview

This chapter presents a detailed analysis of the real time IP multimedia approaches and the time dependency in the context of interactional dialog oriented communication. In addition, the chapter gives a description of the quality of service alongside with implementation in IP networks. Aside from that, list of International Organization for Standardization that have contributed to the development of IP multimedia subsystem standards are exposed. The evolution of the Internet and its widespread facilities is driving telecommunications operators to be responsible for providing commensurable services to their subscribers. The traditional voice communicating service that operates over a circuit-switched network is not sufficient anymore to entice mobile users to spend money with their mobile network carrier. In order to support a modern new and more enticing services for the users, these operators have to use IP based networks. Design a completely new network architecture was essentially required to provide such a closed network model. This telecommunication technology is called IMS. IMS stands for IP Multimedia Subsystem. The meaning of IP multimedia is the capability of communication parties to exchange their digitalized information between each other and quite apart from the fact that they may have a variety of different communication modes, such as interactive communication, streaming, and sharing. Examples for possible applications where IMS may be used are Multimedia advertising, Presence services, Unified messaging, Full Duplex Video Telephony, Instant messaging, Multiparty gaming, Video streaming, Web/Audio/Video Conferencing, Push-to services, such as push-to-talk, push-to-view, push-to-video [3].

"IP Multimedia Subsystem (IMS) is a set of specifications that describes the Next Generation Networking (NGN) architecture for implementing IP based telephony and multimedia services. IMS defines a complete architecture and framework that enables the convergence of voice, video, data and mobile network technology over an IP-based infrastructure. It fills the gap between the two most successful communication paradigms, cellular and Internet technology. Do you ever imagine that you can surf the Web, play an online game or join a videoconference no matter where you are using your 3G handheld device? This is the vision for IMS; to provide cellular access to all the services that the Internet provides." [4]

Real-time communications (RTC) means that users can exchange information instantly or with minimal latency in any way of telecommunications. That means the transmitting content has some kind of time dependence and must be transmitted without any transmission delays.

IMS uses open standard IP protocols to boost the compatibility between the Internet and the IP Multimedia Subsystem. The multimedia traffic is carried by networks using the Internet Protocol (IP). IP is a packet-based protocol which is defined by RFC 791. All networks that use IP are known as IP networks a typical example for such IP based network is the Internet, but usually commercial enterprises and large companies use an Intranet. An intranet is an IP based private network that provide connections to an outside network such as the Internet. The applications of the IP multimedia are applicable over both networks the Internet and intranets. The IMS is indicated in many documents by IETF (Internet Engineering Task Force) and ETSI (European Telecommunications Standards Institute). The reason for this is that IMS offers a way to use existing open technologies and integrates them altogether, to fulfill the requirements of service providers and User Equipment flexibility. They are typically two planes located on the IMS network, signaling and media plane. These planes are not interdependent on each other and both of them are essential to provide IMS services (IMS network can also provide services using only a signaling plane).

IMS Signaling

The main IP Multimedia Signaling Protocols are the Diameter and the SIP protocols. Diameter is an Authentication, Authorization and Accounting protocol and also used for supporting the user information and capabilities of billing to the network functions. SIP is responsible for communication management and the connection of the User Equipment.

Diameter Protocol in the SBC

Diameter is an Authentication Authorization Accounting (AAA) protocol and is an improved version of the RADIUS (Remote Authentication Dial-In User Service) protocol. Diameter is defined in RFC 3588 as a successor to RFC 2865 and it is a chosen protocol developed for the next generation IMS network. It is used in a range of shapes, called interfaces, between various IMS network functions in order to provide exact information wherever are needed. This handling of policy information and media reservations are done in our case by SBC at the border of an access network. As soon as the Diameter protocol being implemented on a network, the Policy Charging and Rules Function (PCRF) performances as the Diameter server and the Application Function (AF), in our case SBC, performances as the Diameter client. SBC Diameter offers users with the option to configure one of two kinds of routing:

• Host based routing

• Realm based routing (several peers can be configured)

In IMS network the interfaces are designated as reference points. These reference points are titled using unique acronyms, such as Rx (receiving reference point). In other hand a SBC Diameter has some limitations,

a SBC Diameter does not repeat states or pending requests for the duration of redundancy switchovers. After a switchover from a failed active to a backup connection all states and pending requests get lost. [5]

Session Initiation Protocol (SIP)

The Session Initiation Protocol is a signalling protocol has been developed by the standardization organization of the Internet (IETF). SIP v2 has been accepted in 1999 as a RFC 2543 SIP: Session Initiation protocol by the IETF [6]. During the first years of SIP v2 implementations a number of problems were detected, because of this a SIP v2 has been republished in 2002, which is the present used SIP version [7]. SIP is an application layer protocol which has been designed to be run independent of underlying transport layer. For the simple reasons that SIP is an open protocol which has attained widespread recognition and a lot of free available open sources of software solutions, SIP has become very popular for VoIP (Voice over IP) services. The basic provided functionalities of SIP are setting up, managing and tearing down communication sessions between users in the network. Therefore the protocol is also referred as a rendezvous protocol. SIP may also be run on different transport protocols, like UDP, TCP and SCTP and has been designed to deal with both IPv4 and IPv6 [7]. The main focal functions are defined in SIP RFC 3261 [8]. Because of the cumulative importance of the Internet in the last ten years SIP has enabled many services to be implemented in the Internet community. SIP allows every interested Internet users to use or even provide telecommunication services like telephony (audio and video). SIP protocol has extensions within IMS, these extensions of SIP protocol are required to provide signaling in order to ensure the functionality of QoS. In place of user equipment a SIP component stringently follows the condition and supposedly the IMS mechanisms. But inside the IMS network, the SIP signaling is able to adjust the client communication to the underlying network, to examine subscriptions and billing functions.

SIP Functional Network Elements

The SIP standard defined by a functional perspective of five different network elements. This is just a functional definition and real implementations typically comprise a combination of functional Elements. A typical example is a SIP Proxy Server with a Registrar server functions. Which has also the ability to act as a User Agent server in case of handling some specific failure situations.

2.1.2.2 SIP Architecture

It consists of two essential SIP entities. They are SIP User Agent (UA) and SIP Servers. User Agent exemplifies end point entity and is commonly a software application running on the system of the user device which can be for example a mobile phone, fixed telephone(hard phone), web phone(soft phone), messaging clients or even a network element like a media gateway. This SIP logical entity can act as both parts of SIP User Agents: The User Agent Server (UAS) as well as the User Agent Client (UAC). SIP distinguishes between these two couple User Agent functions. A UAS receives requests and originates responses while a UAC receives responses and originates requests. In real implementations, a user agent will always operate as both UAC and UAS by containing their functions. SIP operation is based on a client server model but with different approach in comparison with other client server Internet protocols like HTTP. A web server is always a HTTP server while a web browser is always an HTTP client. UA has the ability of session’s initiation and termination by exchanging requests and responses.

2.1.2.3 Back-to-Back User Agent

A back-to-back user agent (B2BUA) is one kind of SIP network element that receives a SIP request, then re-originates the request at its own policy and sends it out as a new request. Responses to the request are also re-originated and sent back in the reverse direction. The media stream can be also manipulated by a B2BUA. It generates also the message body on its own while the SIP proxy server must not. Since it is a combination of a UAC and UAS, no explicit definitions are necessary for specifying its behavior. A B2BUA is an important network service element and has various applications such as Prepaid Service, Transcoding, Anonymisation and Session Border Controller (SBC).

Figure 2: Back-to-Back User Agent (Request-Response message flow)

Figure 3: B2BUA general overview

Figure 3 shows a general overview of request and response sequence between two user agents through a B2BUA. Requests are represented by arrow lines from left to right while responses from right to left. In the figure 3 the dashed arrows indicate the stateful processing by the B2BUA and illustrates how to re originate the request (2) and the response (5).

2.1.2.4 SIP Gateways

SIP Gateways are used to interconnect networks with different technologies (SIP and non-SIP networks). The interconnection between a SIP network and non-SIP PSTN network (Public Switched Telephone Network) are the most prominent SIP gateways usage. Where the SIP gateways here and from SIP network view point behave as a UA’s. SIP gateway network element contains a User Agent. The User Agent here is not a human but either network interface to another protocol such as SS7-ISUP or a User Interface such as ISDN-PRA is typically used. A gateway has the capability of signaling as well as session initiation and termination [9]. At the media gateway between an IP network and circuit switched PSTN/PLMN a media termination is required but in case of a gateway between SIP and H.323 there is no need for media termination due to the SIP and H.323 endpoints are exchanging their media directly on the IP network and a gateway has process only the appropriate signaling passing through [10].

2.1.2.5 SIP servers

There are three logical elements [11]. They are Proxy server, Redirect server, Registrar server. Each of these servers has a specific functions that are explained below in this chapter.

Proxy Servers

SIP Proxy Servers are entities that route SIP requests and responds between SIP user agents. Just like a network router, SIP proxy but a SIP proxy server forwards SIP messages at the OSI application network layer while a network router forwards IP packets at the OSI network layer (IP layer). SIP proxy Server routes received requests from UAC to UAS and responses from UAS to UAC. It is also responding to the requests or forwarding them by acting on behalf on a UA. A proxy server usually has access to a database or a location service that help it during the request processing for determining the next hop destination. A proxy doesn’t need to understand the entire contents of a message to route the requests or the responds. In comparison to a Back-to-Back User Agent (B2BUA) a SIP proxy Server is restricted to manipulate the content of requests and responses (e.g. add, modify or delete header fields). It is only allowed to modify requests/responses and to respond requests from a user agent. Therefore it only relies on header fields. The operation of a SIP Proxy Server can be classified into two transactions mode for each new request, either a stateful or a stateless mode.

A stateless proxy server processes each SIP request/response based only on the message contents (information header field) and there is no information about the message is stored after processing and forwarding a message. Furthermore there is also no state information retained, a stateless proxy does not retransmit a message.

A stateful proxy server keeps track of requests/responses received and uses this information in processing future requests/responses. This type of SIP transaction proxy server is typically the most common one. However there is another type of proxy servers is a forking proxy. It is actually a stateful SIP Proxy Server with forking. Forking is a very valuable feature which enables storing two or more IP location addresses to a Location Database in case of multiple simultaneous registrations of user agents for one generic address which also called Address of Record (AoR) in order to reach the user [10]. Otherwise it comes for the price of high complexity of the implementation.

Registrar Server

A Registrar, or a registration server, is a special kind of User Agent that only accepts REGISTER requests, processes them and responds consequently. There are no any exchanging for other additional SIP messages within. In case of receiving non REGISTER requests will be answered with a 501 Not Implemented response.

For the duration of register transaction, the entire contact information of the request acquired by a registrar, which will be available to other SIP servers where usually the Registrar collocated with, such as redirect and proxies Servers [10]. The reason of this collocation is to ensure the straightforward publication of contact information [12].

Redirect Server

A redirect server is a user agent server that accept SIP requests, but instead of actively forwarding the request to the direction of the target like a proxy, it notifies the initiator of the original request on the location of the target. This is done by including all new target addresses in a Contact header field of a 3xx redirection class response after querying the location server. This method is an active alternative method for SIP Proxy Server in order to resolve the address binding. The redirect server uses the databases or any other corresponding Location Service that aid it to determine where is the next hop or target can be reached at [10]. It generally contains information about the called party’s possible location that allows a Proxy or Redirect Server to use a URI containing an AoR (general address) as input and receive a set of zero or more URIs containing contact addresses as output that tell the inbound SIP proxy or Redirect Server where to send the request. This information can be created, updated or removed in many ways [7].

2.1.2.6 SIP Addressing Scheme

In SIP users and devices (UAs) are identified by Uniform Resource Identifier (URI). Using of this SIP URI to address the UAC and UAS give them the capability to communicate with each other. SIP proxy server and DNS are able to resolve a SIP URI to a physical IP address. The typical SIP URI syntax is look like "sip:user:password@host:port;" [13].

SIP has two main types of addresses (URIs):

1. Generic Address (AOR). This is the address people use to contact the corresponding user, which its format looks like an email address. The SIP URI contains the generic address of the receiver (destination of the request) e.g. sip:[email protected].

2. Contact Address is the address that correspond to an end device and is determined by what IP address a user currently has, the device name, and the port number a user is using for SIP. This is usually a short-term address and stored in memory. When a particular user register with an appropriate SIP server, the server maps this physical address onto his SIP generic (AOR).There is a set of URI schemes that SIP can support such as telephone, presence, and instant message, SIP and secure SIP using TLS. The last two schemes are commonly used. [10].

The following are some of simple SIP URI examples:

sip:[email protected]

sip:[email protected]

sip:[email protected];user=phone

2.1.2.7 SIP Messages

SIP is similar to Hypertext Transfer Protocol (HTTP) in structure and a lot of its protocol functions are borrowed from HTTP and SMTP. This is why SIP is a text based protocol which means the message context can be human readable, clearly visible and easy to be changed. According to RFC 2822 there are two styles of SIP messages namely SIP request messages from a client to a server and a SIP response messages from a server to a client [13]. The sip message format contain a start-line which is either a request-line in case of request message or a status-line in case of response message. Next follows the message headers which is itself followed by a CRLF sequence (Carriage Return and Line Feed character). And an optional message body which is depending on the semantic of the message [7].

Figure 4: The generic format of a SIP message

SIP Requests Messages

In RFC 3261 there are six request messages have been defined INVITE, REGISTER, BYE, ACK, CANCEL and OPTIONS.

The REGISTER request message: it is used to register the UA and to tell SIP Servers about its current location.

The INVITE request message: is the first request message which is used to establish a session and usually includes a session description protocol in the message body of the INVITE request [14].

The ACK request message: the acknowledgment request message is a confirmation message sent by the UA telling the other partner that its final response has been received.

The BYE request message: it is used to tear down an established sessions. The BYE request message is sent by SIP UAs, all other SIP network elements are not able to terminate a session.

The OPTIONS request: it allows a User Agent to query another User Agent or a SIP Proxy Server about its capabilities. This is how a User Agent explores information about the supported (methods, content types, extensions, codecs, etc.) [11]. All User Agents support the OPTIONS method.

The CANCEL request message: it is used to cancel the pending sessions. A typical example for a CANCEL message is canceling a previous request sent by a UAC by asking the UAS to discontinue processing the request and to originate an error response to that request. A forking SIP proxy may also use CANCEL request message in order to cancel calls progressing with other end points when a session with a UA has been initiated.

SIP response messages

When a user agent or proxy server receives a SIP request it generates a SIP response for that request sent by a UAC. An ACK request does not get any response at all while all other SIP request messages have to be replied with an appropriate SIP response. Status information what a SIP response message contains, which is related to a particular request. SIP responses are quite similar to the SIP requests with the exception in the first line format which contains protocol version (SIP/2.0), status code, and reason phrase. The Status code specifies the response type and consist of 3 integer digits value from 100 to 699. In case of a SIP response the first line (start line) is a status-line. An example of a status-line is:

" SIP/2.0 180 Ringing "

SIP response messages are classified into 6 classes defined by the first digit as follows: 1XX, 2XX, 3XX, 4XX, 5XX and 6XX [7].

1xx

Provisional

request received, continuing to process the request

2xx

Success

the action was successfully received, understood, and accepted

3xx

Redirection

further action needs to be taken in order to complete the request

4xx

Client Error

the request contains bad syntax or cannot be fulfilled at this server

5xx

Server Error

the server failed to fulfill an apparently valid request

6xx

Global Failure

the request cannot be fulfilled at any server

Some response classes are described below [10]:

100 Trying: A 100 Trying response is provisional response that typically sends by a SIP proxy Server to inform the associated UA that its related request was received and the Proxy Server now take care of this request. Now the sender has been informed about its request and there is no need for retransmitting. Otherwise the UA (sender) will keep retransmitting the request periodically.

180 Ringing: this provisional response is typically originated by a UA and means that the phone of the callee is ringing.

200 OK: There are two possibilities here to send a positive 200 OK final response. Either to accept session INVITE request and in this case a message body is usually required to describe the media properties of the callee or to indicate that the SIP request messages are successfully completed.

Sometimes even though a UAC sends only one INVITE request, it my get several 200 OK responses back from different UASs. This is happened only when a UAC has sent its request to a forking SIP proxy which can split the request to reach many UAS at the same time. Each UAS will accept the invitation and response with 200 OK message to that associated UAC.

300 Multiple Choices: Means that the related address in the request URI resolved to more than single choice. When UA had multiple matches and it could not determine which destination to choose, the user has to choose the desired target for communication to complete the call and redirect the request to that location.

403 Forbidden: This is a negative response telling the caller that its request has been denied. The request may be understood from the server but not processed.

415 Unsupported Media Type: This is a failure response sent by a UA to inform the sender that this type of required media in its request is not supported. A typical 415 Unsupported Media Type response includes a set of header fields with supported media type to aid the UA.

513 Message Too Large: The UAS send a 513 Message Too Large response to the sender if the request size was too large and exceeded its capabilities to process it.

603 Decline: The called party does not want to participate in the call, or just this destination or any other alternative destination is unable to accept it (e.g. voicemail server).

2.1.2.8 SIP Message Body

Both SIP messages requests and responses may also have message body. SIP message body itself includes different type of messages. It can be for example an XML message body which contains the current user status or its configuration data, text based message body, status information message body. A typical message body in SIP carries session description information (SDP information) which is used in INVITE requests and corresponding destined responses to that INVITE request in order to describe the session. Since the content of SIP messages body is transparent to SIP proxies, there is no need to investigate it at all. This either means that the transmission of SDP data based on end-to-end communication model between user agents [12].

3. Session Description Protocol (SDP)

Session Description Protocol is a defined protocol by RFC 2327, which has been developed by the MMUSIC workgroup of the IETF. The functionality of SDP is not a protocol functionality as it named, hence SDP defines the syntax for textual description of a full range of session features and capabilities. The content of SDP is inserted within SIP messages bodies when the Content Type-header field of such messages shows "application/sdp". The initial SDP purpose is to provide specific detailed description about all parameters of multicast session set up over the multicast backbone network of the Internet, the MBONE. The first defined SDP by RFC was RFC 2327. Meanwhile it has been replaced by RFC 4566 [15]. When SIP required a means for session description IETF decided to use the available SDP again although the application was a little different and SDP was not a 100% fit for this application. In case of multicast there was only one direction for the media stream in contrast to SIP sessions that means SDP does not have the capability of a full range media negotiation, where media streams flow among peers in both directions. Also multicast sessions were scheduled with time and sometimes several source streams mixed with each other. SDP includes a complete set of information that is required to explicitly establish media channel such as media types, list of codecs and their parameters, IP addresses, stream direction, available bandwidth etc…

An SDP session description be made up of a set of textual lines attribute / value pairs, one for each line where the coding of the attribute names was as single character and values were either an ASCII string, or a sequence of certain types detached by space. The exact form of all lines must just look like the given order in the specification. A typical session description structure is illustrated in Figure 5.

Figure 5: SDP structure example with description [16]

Figure 6: Annotated SDP Session Description [17]

4. Real-Time Transport Protocol

The Real-Time Transport Protocol (RTP) was developed to carry real-time packets containing voice, video or other time sensor data information over IP. RTP is defined in RFC 3550 [18]. Since there is never time for retransmitting the lost packets, the RTP implementations are generally built on UDP. Because of that RTP does not guarantee any quality of service (QoS) over the IP network. This means, that RTP packets are handled like all other packets in an IP network. RTP does not either guarantee delivery through the network. RTP does, however, prevent Out-of-sequence packet arrival to the application by checking the sequence number of the received packets. Some of the impairments, such as Asymmetric routing, Variable transport delay and Packet loss, can be detected by RTP protocol [10]. RTP is an application layer protocol which typically runs on UDP and relies on it for transport, but it may also be used by other appropriate underlying protocols. The encoding of RTP protocol is not textual, but just like a UDP and IP, the RTP uses a bit oriented header. RTP has been planned to be very generally and flexible protocol for real-time data transmission such as audio and video. RTP has a set of valuable protocols features and real-time nature, which requires a minimum latency (delay) across the Internet. This is way it has an efficient performance of supporting multimedia conferencing and media streams merging. The RTP Control Protocol (RTCP) is an associate protocol to RTP and its basic functionality and packet structure defined in RFC 3550. It provide monitoring of data delivery by allowing the participants in an RTP streaming multimedia session to send quality feedback and static information reports between each other periodically. Even port numbers are used by RTP, while the next higher odd port number is used by the associated RTCP stream [19]. RTP is the most important prepotent transmission protocol in real time multimedia on IP networks and all other signaling protocols in this dissertation depends on its usage.