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Protocols play a vital role in networks. Mainly they are divided into two types. Protocols which carries call controls and used for signaling purposes Protocols used for voice payload like RTP RTCP UDP and IP. When we connect a call between two end points after setting up call process the source and destination can transmit and receive data by using the voice payloads. We can use different combination of protocols depending upon the source and destination ends.
These are used as interfaces between the IP telephone ,Networks and PSTN. One end of the gateway is connected to the controlling and signaling protocols, used for the interfaces with the PSTN and other end of the gateway supports IP Telephone networks through call manager which acts as like a protocol translator. Call manger communicates with the IP Phone using Sccp and also other side it will be communicating to gateway using MGCP and H.323
H.323 standard technology used for a LAN based networks along with multimedia applications. In 1996 the H.323 specification was approved and two years later they released version 2 which supports wide range of technologies includes stand alone devices, point to point and multipoint conferences. In addition to this it supports call control functions along with multimedia and bandwidth management(TOBY J Velte,2001)
"H.323 specification is used to carry real time voice traffic such as telephone calls, video and data over IP network" (Ramesh kaza,2005). By using H.323 multimedia applications can be used without having overhaul the whole network. IP LANs are getting more powerful from speeds of 10 mbps then 100mbps and now 1gbps. It allows internetworks functionality between two different networks for communication. By increasing network manageability we can limit the amount of bandwidth and efficient use of network resources and also supports multicasting that reduces bandwidth consumption. H.323 constant improvements of processors, memory and chips benefits many computing and communications companies Microsoft , Cisco and IBM biggest names backing H.323 (Anthony T.velte,2001).
H.323 defines four major components for network based communications system
Terminals: These are the endpoint devices used by the client. It supports voice communication but video and data are optional. H.323 specifies of how different audio video data terminal will be work together and also it supports H.245 which is used for negotiation of usage of channel and its capabilities.
Gateways: It is used as a translator between H.323 endpoints and other different terminals With different audio and video codecs performing other function like call setup. It acts between transmission formats and communications procedures. In H.323 gateways are optional component used when you are connected to other networks because endpoint can communicate each other in a same network.(frank ohrtman, 2004)
Gate keepers: H.323 gatekeeper is the central point for all call controls within its zone and it provides services to the registered endpoints. We can manage on the usuage of bandwidth is a optional function of routing calls which helps more effectively and efficient calls in a network(salman asadullah,2005)
An alternative of H.323 in the VOIP is Session Intiation Protocal. It is a application layer protocol used to establish, maintain and terminate the call between terminals. SIP does not needed a dedicated sever to manage its sessions. Peers in a sessions are called user agents(UAs) functions in two roles.
User Agent Client(UAC) - It is used to initiate the SIP request
User Agent Server(UAS) - When you receive a request from SIP it will be returns responses on behalf of user.
Endpoints can function as a UAC either UAS depending upon the request by user. It cannot be both at a time so it functions only as UAC or UAS
SIP can be divided into two sections
SIP Clients -> Phones, Gateways
SIP Server -> Proxy Server, Redirect Server, Registrar Server
Phones: Soft phones and Cisco SIP IP phones will be initiating the SIP request and it will be responds to the request. These devices can be used for either UAC or UAS
Proxy Server: When Server Receives a SIP messages and it forwards to the next server in a network. It provides authorization routing, security authentication
Redirect Server: It contains the next hop information that message should take. client will be contacting directly the next hop server or UAS
Registrar Server: This processes the request given by the UACs for registering their current location
Basic functions of SIP
Invite Message -> For initiating a session it was sent by UA
Ack Message -> Used to acknowledge after the successful session is initiated
Bye Message -> For terminating a session which is already established
Register Message -> It tells to the network where it can be located and sent by the UAS. This location can be one or more addresses.
Cancel Message -> Used for terminate the session which is not progressed
Option Message -> States the capabilities of other UA
Calling party will be sending a invitation to the called party.
Proxy server responds to calling party request which states response "100 trying" and proxy server invites a message to called party on behalf of calling party.
Called party gets two responses "100 trying" & " 180 ringing"
Proxy server sends response to calling party "180 ringing"
Called party responds "200 ok" if call is successful
Proxy server passes information to calling party the success response "200 OK"
Calling party sends ACK message and then proxy passes ACK message to called party.
SIP successfully establishes path between two end points.
Designed as internal protocol within a distributed system this appears outside as a single gateway. It uses session description protocol for describing and negotiating the media capabilities.
Media gateway: It bridges the networks with different media types such as analog networks, circuit switched and IP networks. It is a interface between telephony network or PBX and VOIP network