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Session initiation protocol (SIP) is protocol way to setup sessions across Ip networks. It can also define as connection between two IP end points. It can be conference call; videochat etc.The features of Session initiation protocol made it more important in fast growing telecomindustry. Flexible, Open SIP capturing power of internet and mobile IP networks to create new generation of services. It takes communication flexibility to next level. One of best function is it is independent of transport protocol that it has ability to operate over many protocols like TCP, UDP etc.It rather defines how a user can create, modify and terminate the connection of voice, video .Though Session initiation protocol does not solve every problem it gets job done from other protocols easily . It is friendly protocol plays nicely with other protocols. It plays main role in modifying and terminating session. Session initiation protocol reuses many exiting protocols. It uses 404 for address not found and SMTP for address schemes.The main reasons for implementing Sip is it locates users and resolves their SIP address to an IP address. During session setup it acts as signaling protocol providing services in internet text.The flexibity of Session initiation protocol made more helpful than other protocol that it uses text based with lower head.It can change session parameters during call that a user can enable even video function on call.It can handle call setup and teardown of call users in the session.Though Sip works similar to H.323 protocol, flexibility and scalable features made it more helpful to telecom industry.The extension of SIP can be more useful for instant messaging and managing events.
From many years Protocols like SS7,H.323 have ruled public switched telephone network but the problem with these protocols is that it is difficult in time consuming developing applications like call waiting, call forwarding, voice mail etc.call waiting and multi conferencing which made people to think of their comfort. These are useful especially when a person is willing to talk to many people simultaneously to share information.voice mail are highly useful when a person busy in their business meetings or emergency situation.Though the person does not respond he can send or check information on voice mail.H.323 protocol supports this feature. Though its supports call waiting, implementation is not flexible and design is very heavy where as sip is flexible and easy to implement. Voice over IP is another thing which is very flexible and lowcost.As it support internet to internet calls including video conferencing it could not assure quality. Skype and fring are examples of voice over Ip.For voice over Ip definitely a internet connection should be established. It could not transmit any data with out internet connection. Internet plays a vital role in voice over ip but the problem with voice over IP is it does not guarantee quality of service. voice over IP needs high speed internet connection to transmit information properly if not there will be a delay(voice can break because poor performance).These new technologies have challenged the traditional telephony and telecommunications.From past many years there are no protocols which supports advanced features like call waiting. Many protocols have tried but they have their own limitations. In order to overcome limitations of some protocols a new protocol was developed called session initiation protocol which establishes session across Ipnetworks.
Session initiation protocol is a way to set up session across IP networks that its a simple text based used for establishing multimedia session. It is simple as it uses 4 methods in its modules.
1. USER AGENTS (UA):It initiates sessions. It can be personal computers, cell phones or IP phones. Sip request are generated by user client where user agent server responses back to client.
Examples: IP phones, cell phones.
2. REGISTRARS:It can be referred as database of location of user agents(UA) within domain that it keep track of users.
3. PROXY SERVERS: It handles call routing authentication, loop detection per domain.It is application layer that forwards session requests and responses.
4. REDIRECT SERVERS: These are used by proxy servers if the call is off domain.
The working of Sip is very simple as it uses fewer resources.
Proxy and redirect servers use logical entity location server to route requests. Sip mainly relies on RTP (real time transmission protocol).Whenever a user sends a request it sent to proxy server. Proxy server checks the sip address with the help of registrars. If it finds the address then responses to request and connection will established that it gives a acknowledgement (Ack).After connection setup to release it responses by Bye function so that connection is released. In order to have a connection between users there must sip address .As it is textbased similar to HTTP with fewer resources overcomes the limitation of H.323 protocol.As it is independent of transport layer protocol it can use many protocols like tcp,udp but sip prefers udp as it avoids connection setup. Different protocols can be used between proxyâ€™s end user agents while forwarding a single message.
Proxy servers and registrar can be combined or independently operated.Another protocol in which Sip relies is SDP(session description protocol)it describes what to receive.SDP in SIP is unicast it indicates receiving capabilities and destination address and port for any number of streams. It initially generates a plaintext password. Sip are carried using transport layer or IP layer security.
Example of sip without proxy server: In this example during call setup user agent sends invitation to another agent. A connection will be established with ringing signal and give response with acknowledgement.
Sip uses urls have specified email address format as user@domain.
Examples of successful sip call set up:
When ever user agent want to establish connection with another person its send invitation,the invitation is first sent to proxy server it can be connected to multiple server where it check the database location of registered sip address.If already sip address is registered then its connects to other agent by ringing.Then acknowledgement will given as request response by ACK.To release a connection session it responses by BYE.
My idea is to make call from internet to landline phones, cell phones using session initiation protocol. let us elaborate with the example
SKYPE:skype is voice over Ip application in which a connection will be established with software installing in pc or mobile.It provides video conferencing with audio but the problem with this it needs high speed internet.If the speed of internet is low there will be delay in transmitting voice.It cannot assure quality of service.Though it is more flexible and lowcost it is limited to only where internet is available.VOIP will not work during power outages. Session initiation protocol establishes a session across Ip network .As session initiation protocol uses URL ie email based it is easy to implement even registration format of sip is just like email.(user@domain)can easily establish a call between user and called agent. This can be implemented by having a sip account. If once we have sip account a call can be made to any place irrespective of network provided sip address is available for users.This is free of cost and more flexible it even supports video calling .With the help of sip address a user can make a call cell to cell ,internet to cell, cell to fixed line. Sip initially provides a port number while registering sip account.The extension of Sip can be useful for instant messaging and managing event.
SIP utilizes transport protocol like TCP , UDP so it is vulnerable to tcp session hijacking, spoofing.
Client/server likes to keep information confidential but due to open and distributed architecture of VoIP cannot keep it private.
Less complex compared to H.323 protocol.
Lightweight uses text based similar to HTTP and SMTP
Uses URL and fewer resources.
Sip does not make PSTN easy and cannot assure quality of service.
By implementing security mechanism like IPSEC,TLS it can be overcome.
By improving quality of service of sip it makes more flexible for users to use.The interworking of PSTN should be solved and made easy to work.Apart from these proper security mechanism should be followed to avoid session hijacking ,spoofing .This can be improved by making authentication tight especially in keeping information confidential.If all certain precaution are followed then SIP can be sensation in fast growing telecom industry .This will be more useful for people due to its flexibility and free of cost.
User agent software module shall be integrated with soft phones and live voice communication over internet can be realized.
Proxy server can be augmented to handle multiple user agents simultaneously.
The importance of session initiation protocol is explained including its working. Establishment of sessions during call is mentioned. Protocols like SDP,RTP are explained. Improvement of SIP is explained to make more efficient in coming years.