VOIP AND SIP PROTOLCOLS
What is VoIP?
VoIP (Voice over Internet Protocol) also called IP communication framework as it gives
the telephone benefits over the internet. With a decent internet connection, it can give
telephone benefit conveyed through the internet connection rather than nearby telephone
organization. It transmits voice and mixed media content on internet protocol. It is the
combination of advancement and methods use to transmit call over the IP. It can be
deployed on LAN and WAN.
The two main objectives to use VOIP
2- Increased Functionality
Basically, VoIP lines cost less than the traditional PSTN lines provided by different service providers. In traditional PBX system additional cost is paid for the service and the cost increases as the features are added into the PBX system and the PBX is managed and configured by the specific company engineers, while in VoIP PBX system for instance Asterisk VoIP PBX system, it is free and it can be installed on the simple PC and can be managed and administrate by the user itself and are enrich in featuresas compare to the traditional PBX.
The call over internet protocols are free but user have to pay for their internet connection.
Also, there are service packages for calling on PSTN and GSM from VoIP which are way
cost saving and cheap as compared to the charges of the PSTN or GSM service providers.
There are many services which supports this technology like free calling for example
Skype, WhatsApp and Free world dialup.
VOIP make thing easier as on your fingertips which
are almost impossible in traditional phone networks.
Calls can be routed directly to the desired VOIP phone. Most important feature is that it
provides you a new level of mobility in case of PBX systems. User can carry its phone
anywhere and can communicate.
It is very beneficial for the call center agents as they can communicate form anywhere
and can provide service and assistance from almost anywhere.
How VoIP works?
The basic concept behind VoIP technique is
- First of all it continuously sample the audio.
- Each sample is converted into digital form.
- Then the digital stream is sent to the IP network in the form of packets.
- These packets are converted back into the audio stream for the playback.
- Before the procedure above, the system must handle call setup. Phone number to IP.
Just like every call system require some protocols to follow for signaling and call handling
for example, SS7 (Signaling system 7) is used in PSTN network, SIP (Session Initiation
Protocol) is used in VoIP network for signaling and call establishment and handling.
What is SIP?
Session Initiate Protocol, its SIP base. The SIP Channel Segment enables Asterisk to
connect by mean of VOIP with SIP phones and exchanges. Asterisk is able to act as a SIP
client. Taste is an extremely adaptable convention that has awesome profundity. It was
intended to be a universally useful approach to set up continuous mixed media sessions
between gatherings of members. For instance, notwithstanding basic phone calls, SIP can
likewise, be utilized to set up video and sound multicast gatherings, or texting meetings.
In this report, we’ll concentrate on SIP’s capacities for VoIP, and how it sets up calls that
at that point utilize RTP (the Real-time Transport Protocol) to really send the voice
information between telephones. Asterisk goes about as a Media passage between SIP,
IAX, MGCP, H.323 and PSTN affiliations. For instance, an Asterisk server can be
connected with ISDN to give your SIP customers availability to the exchanged
correspondence deal with.
Like in SS7 there are messages like IAM, ANM, and REL etc. there are six basic messages
types of SIP known as methods which are used for the establishing and handling the call.
INVITE A request is sent to the other user for call initiation
ACK The requested user responds the request for session initiation
BYE Session terminated
CANCEL Cancel the pending requests
REGISTER It registers the location of the user
OPTIONS It determines the capabilities of the end user.
There are also other protocols except SIP for routing and call management and converting
information form one form to another such as IAX AND DAHDI but we will discuss it in layer articles at knowledge-bull.com