How SIP Is Used In VoIPPosted on: 2016-06-20 | Categories: SIP SIP Trunking
In the world of telephony, VoIP and SIP are often used interchangeably by users but technically speaking, they are not the same. Though they are related, they are not substitutes for one another. You cannot simply replace SIP with VoIP or vice versa. In reality, SIP is one of the most popular ways of implementing or delivering VoIP service to consumers, households, enterprises and other entities. VoIP is a much broader term that can be applied to any form of voice calls that happens over the Internet.
What exactly is SIP?
In the world of telecommunications and network engineering, a protocol is a set of rules that can be used by two entities to transmit data, information or signals. These rules define the syntax, method, standards and semantics of communication, to minimize confusion and make sure communication sessions can proceed efficiently.
The Session Initiation Protocol – which is abbreviated to SIP – is one such protocol that is used for signaling, setting up and controlling multimedia sessions. These sessions may involve audio, video, instant messages, 2 callers, multiple callers or any combination of the above variables.
What the end-user considers a single phone call is actually divided into two phases – ‘call set up’ and ‘data transfer.’ SIP is used in the call set up phase and implemented in the application layer. SIP is used to initiate the call between two or multiple parties via SIP proxy servers. Once the initial parameters have been established, the actual call and voice data transfer happens directly between the endpoints in a peer to peer fashion.
The relationship between SIP and VoIP
As stated above, SIP is one of the most common protocols used to deliver VoIP services. It is an IETF standard that was first introduced in the 90s and it continues to evolve today. To visualize the relationship between SIP and VoIP, take the example of Internet or email. These are pretty universal services used by practically every person and organization today.
Email is commonly implemented through SMTP – Simple Mail Transfer Protocol. By Implementing the services through universally accepted and standardized protocols, end users are able to send email or surf the web without barriers. For instance, a Gmail user can send email to an Outlook user or a Yahoo user or an enterprise user in much the same way. It doesn’t matter if the person is using a standalone desktop client or a webmail dashboard.
This is pretty much how SIP and VoIP are related and work. When an organization utilizes SIP to deploy their VoIP network, it is easy for other entities to work with the same equipment. For instance if the business purchases new SIP phones, they are already compatible with the network because they use the same protocol.
How SIP is used in VoIP
SIP is a text based protocol that was developed on a similar model to HTTP responses. When compared to alternatives such as H.323, it is very simple to construct and debug. But it is also extremely flexible and has powerful features that can be utilized in very complex PBX systems. It is a general-purpose protocol that can handle a variety of sessions. In VoIP specifically, SIP is used to perform a variety of functions including:
SIP endpoints – basically your telephones – register themselves with proxy servers that maintain lists of individual endpoint locations. When a particular phone initiates a call, the request is sent to the proxy server which then contacts the other endpoint to set up the session.
SIP determines if the various parties are available to answer the call or participate in the session. It means that different endpoints can set up rules stating when or how they would like to be contacted and when they are available for calls. For instance, a particular phone may not accept calls from 8 AM to 5 PM or will only accept calls from within the same network etc.
The various endpoints involved in a particular session use SIP to determine what kind of audio codecs and other media capabilities (video, file transfer etc.) can be utilized in the call.
Session set up
SIP tells the phone that is being called to start ‘ringing’ and manages the initial contact between all parties.
SIP also performs other management functions such as ending the call, adding another caller, switching to video conferencing etc.
How did SIP become so popular?
SIP is not the only protocol that can be used for implementing VoIP, there are plenty of others like H.323. At present however, very few organizations and equipment manufacturers implement VoIP on other protocols. SIP has become the defining standard and universally accepted protocol for VoIP. How did this happen? There are several reasons for this development.
Even though SIP is used for voice communication, it was not developed by or within the telecommunications industry. Instead it is an IP-based protocol that is maintained by the IETF which is the Internet Engineering Task Force. SIP is flexible and because it has its roots in the IT industry, is much more capable of being integrated with other Internet applications, software and systems. Thus it is highly suitable to the way businesses function in today’s world, rather than being a relic of telecom systems from the last century.
Organizations do not need to hire telecommunication experts for VoIP deployments. It is a simple text based protocol and similar to HTTP. This means that many IT professionals can immediately become familiar with SIP as they have been working with HTTP for many years. Since VoIP essentially treats voice calls as simply a form of data, enterprise deployments are generally managed and maintained by the IT department itself.
In addition to this, it is an open source protocol and has been developed to ensure interoperability from the beginning. Manufacturers of VoIP equipment frequently test SIP compatibility before launching their products. For all the above reasons SIP has gained significant traction and is now the first choice for any organization wishing to deploy VoIP networks.