5. The H.323 signaling protocol

The three commonly used signaling protocols in VoIP are:

• H.323
• Session Initiation Protocol(SIP)
• Media Gateway Control Protocol(MGCP)

The H.323 Signaling protocol

H.323 is an ITU-T specification for transmitting audio, video and data across an IP network . It is implemented at layer 5 of the OSI reference model. It provides ideal telephone communication, support simultaneous voice and data transmission and can also transmit binary message that are encoded using basic encoding rules. It also relies o 2 additional signaling protocols, the H.225 and H.245 for call setup and management.

H.323 components include the following:

• Terminals
• Gateway
• Gate keeper
• Multipoint control Units.

Terminals

• It is an end point device such as IP telephone, softphone or a dedicated conferencing device.
• It is designed to support voice, data traffic and IP telephony.
• It mus H.245 for channel and negotiation, RAS(Registration Admission stations)
• Q931 is used for signalling and setup. It also provide setup and support RTP and RTCP on the stream of the media
• Terminals must support audio with video and data transmission.

Gateway

• It perform call setup and tear down of the session between two parties
• It also help to translate audio and video signals.
• Besides that, it provide RAS for registration .
• It reflects the characteristics of a switched circuit network(SCN) endpoint and H.323 endpoint.
• Provides an interface between IP telephony system and PSTN.
• Can be implemented on a gate keeper , a MCU or on a voice enabled router and switch.

Gatekeeper

• Provides pre-call and call level control services to H.323 endpoints.
• It is separated from other network elements in H.323 environment.
• If more than one gatekeeper is used, an intercommunication is accomplished in an unspecified manner.
• Manage all the registered terminals, gateways and MCUs in a single H.323 zone.
• Services such as addressing, authorization of H.323 components , bandwidth management , accounting and billing can be configured on the gatekeeper.
• It can be implemented on a dedicated server.

Multipoint Control Units(MCU)

• Support conferences between 3 or more endpoints in a multi point conference.
• Consists of a Multipoint controller(MC) and an optional Multipoint Processor(MP)
• The MC is responsible for H.245 functions while the optional MP handles the actual mixing of media streams to avoid bandwidth contention.
• It act as a bridge between centralized and decentralized enviornment.
• It receives audio, video and data and distribute them to endpoint participating in a multipoint conferences.
• Both MC and MP can resides on to a dedicated components , a gateway, terminals or gatekeeper. But it is recommended that MC be utilized on a gateway.

The 5 steps of signaling in H.323 is as followed:
1. Call setup
2. Initial communication capability
3. Audio/video communication establishment
4. Communication
5. Call termination.

At step 1, when user dial the number of the other party, the first set of signal that are exchange is the TCP connection. TCP-SYN, TCP-SYN-ACK and then the TCP-ACK are generated between two user. At this point the H.225.0 SETUP ON TCP signal informs the party that the connection can be set up on TCP. Next, the users can now request a certain bandwidth from it associated gateway server of the called party.

The requested bandwidth from the gateway granted the bandwidth if there are sufficient bandwidth, id not the call has to find another gatekeeper to register. This phase is handle by H.225 and H.245 protocols.

At step2, all the end point's communication are available over TCP. This phase is necessary as the type of communication service requested depends on the communication capabilities at both ends.

At Step3. it implements the establishment of a logical channel, which H.323 is unidirectional, so a logical channel must be establish in either direction in order to have a full duplex communication.But at the same time more band width is needed

Step 4 handles the communication between two user. This phases is control by using RTP over UDP. This step can be any kind of media flow depending on the size and type of the channel establish in step 4.

At step 5 , is the termination by either user and the termination is usually done by gatekeeper.

The following shows the signalling process of H.323:

H.323 Protocol suite

The H.323 protocol suite is based on several protocols. The protocol family support admission, call setup, status ,tear-down , media streams and message in H.323 systems. These protocols are supported by both reliable and unreliable packet delivery mechanism over data networks.

Although most of the H.323 implement TCP as the transport protocols for signalling, H.323 version 2 does not implement TCP but instead use UDP as the transport protocol.

The following shows the H.323 protocol suite

Fig 5 H.323 Protocol suite

This protocol suit can be divided into three main areas of control:

• Registration, Admission , and Status(RAS) Signalling
• Call control Signalling
• Media control and Transport

4. VoIP Signalling Protocol

In VoIP, a signaling Protocol handle call setup, conversion of phone number to IP address mapping and proper call terminating . It usually involves the user location phone number , finding route between a caller and a called party . It also handle things like all forwarding and other call features.

IP telephony signaling can either use a distributed or a centralized signaling scheme.

In distributed signaling scheme, it enable two IP Phone to communicated using a client/server model. It works well with VoIP network within a single company. In a centralized approach, it uses the conventional model and provides some level of guarantee. Signaling is usually done between two IP phones and each phones will exchange those packets to ensure there is a communication link.

In the next section, we are going to look at different types of signaling protocols used in IP telephony.

3. Packets

A packet is a unit of data that is routed between an origin and destination on the internet or other packet switched network. Each packet has a header and a trailer. The header usually consist of the sources and destination address. It may also provide error checking . The trailer usually consists of check sum and error detection function. In VoIP, a voice packet usually consists of several protocol such as Internet Protocol(IP), User Datagram Protocol (UDP) and Real-time Transport Protocol (RTP).

Another similar term to packet is called Datagram which is used in User Datagram protocol. Packet is usually transported using IP, and the re sequencing is usually done by Transmission Control Protocol.

The following diagram show the IP Packet format:

2. Differences between Packet switching and circuit switching.

Circuit Switching is the basis of a conventional telephone system. A circuit between two user must established for a communication to occur and it must maintained for the duration of the call until one of the party hang up. This network need resources to be reserved for each pair of end user. Also, there must be no other user that can use the already dedicated resources for the duration of network use.

Circuit switching is ideal when data must be transmitted quickly, arrive at a sequence order and at a constant rate. So, it is used for audio, video and real time application. This type of network is establish as a virtual circuit between endpoints and the single route which does not vary during connection and the nest connection may take a difference route.

The following diagram show an example of circuit switching network. In the diagram, the communication link is set by thick blue line for the conduit of data from device A to device B and a matching purple line form B to A. Once set up, all communication between devices take plcaes over the conduit , even though there are possible ways that data could be conceivably passed over the network of devices between them.









Fig 1 Circuit Switching Network


Packet switched network is a unified, integrated data between infrastructure that can provide a variety of communication services that needs a different bandwidth.. It keeps the connection long enough to send packets form one end to the other. It is used for the integration and transmission of voice and data . Besides that, it span a large geographical are and comprise a web switching nodes interconnected through transmission links. It allocate resources when required.

The following diagram show a packet switching network. In this network, no circuit is set to prior to sending data between devices. Blocks of data, even from the same file, may take any number of paths as it journey form one devices to another


Fig2 Packet Switching network

The advantage of packet switching is that both parties can transmit at different rates. But the problem is that too many packets can cause congestion and packets that cannot be stored or delivered might be discarded by the packet switching exchange.

VoIP uses packet switching as It needs to send those voice packets at a faster rate while the conventional telephone uses a circuit where it needs dedicated path between two user.

1. Introduction to VoIP - Voice over IP

VoIP, also know as Internet Telephony, is the technology that make it possible to have a telephone conversation between two parties over the internet or a dedicated IP network instead of voice transmission lines. Early Voice over IP system requires the dedicated user PC to be eqqupied with the sound capability and a software application.

This software application convert speech signal into digital packet using a specified type of protocol as the transport medium. It also translates into free or very low cost long-distance calling.

The basic scheme of IP telephony starts with Pulse Code Modulation(PCM). The encoded data is transmitted as packets over packet switched network. Onces it reaches the receiver side, the data is decoded and convert back to analog form which travel through the handset. The process of digitization, compression and packetization is handle by Codec - Codec/Decoder.

In VoIP, voice are transported in packets form through the internet and usually an IP phone or software is used between two parties. These devices will convert the analog signal into digital format, followed by compression. The purpose of compression is to save bandwidth and the more it compressed, the lesser quality of the voice. After that, the binary value is input to a packetizer which consists of IP packets and transmit through the internet.

At the receiving end, the packets will input to a depacketizer that will only keep the payload. After that the payload(digital signal) is converted to analog signal using DAC and input to a standard.

In short, to transmit voice over the IP network, the three steps must be followed:
1. Digitization
2. Compression
3. Packetization

There are three methods of connecting to a VoIP network:

• Using a VoIP telephone
• Using a "normal" telephone with a VoIP adapter
• Using a computer with speakers and a microphone

VoIP telephone calls can be placed either to other VoIP devices, or to normal telephones on the PSTN (Public Switched Telephone Network).

There are two types of PSTN to VoIP services: Direct inward dialing and access numbers.

Direct Inward Dialing(DID) will connect a caller directly to the VoIP user ,while the access numbers need a caller to provide an extension number for the called VoIP user.