SIP-Based VoIP: Learning the Basics

Understanding VoIP

VoIP has become a very well-known term these days, particularly among businesses that require telephony communications. VoIP, the abbreviated form of Voice over Internet Protocol is a methodology and group of technologies that are used for voice and multimedia communications over IP networks, such as the internet.

SIP-based-VoIP-basics

VoIP has been developed as an alternative to the standard circuit-switching telephone networks, that is to say, the traditional wired phone networks. VoIP has been adopted by many companies for its highly affordable long distance and international calling rates as well as the myriad of functionalities it provides that the traditional phones cannot.

Knowing the Data Paths

Ever since digital voice coding has been made possible, there was a lot of thought on trying to converge telephony and IT environments in order to transmit data, voice and video applications using the same channels. However, each of these applications had different needs. Data needed variable line bandwidths, is not bothered about the connection’s reliability while at the same time, voice and video transmissions which have real-time applications required a constant bandwidth and a guaranteed reliability of the connection.

There were many technologies developed to accommodate both these applications each having its own inner workings. However in the end most of them were eliminated, leaving two technologies that improve interaction and compatibility of products from different companies: H.323 and SIP

Among them H.323 is capable of delivering a more unified communications during the same session and SIP is more about maintaining a session between the two points for a simplistic Internet architecture. VoIP technology separates voice and signal communications during transmissions. Signal sessions re-handled by a server that replaces a standard PBX while the voice stream is created point-to-point between the end sides.

What is SIP?

The Session Initiated Protocol (SIP) is a communications protocol for signaling and managing multimedia communication sessions. The most common uses of SIP are in the fields of internet telephony for voice calls, video calls and instant messaging over Internet Protocol (IP) networks.

SIP is a highly flexible protocol with a huge depth. It has been designed to act in a general purpose manner in order to establish real time multimedia sessions between a group of participants. That is to say, SIP can be used to set up an audio or video multicast meeting along with simple telephone calls.

One of the best features of SIP is that it is a text-based protocol modeled on the request/ response model that is also used in HTTP. Due to this SIP is very easy to debug as the messages are easy to construct as well as easy to see. Unlike H.323 the other protocol used for VoIP, SIP is very simple while still providing powerful features.

Why is SIP better for VoIP?

Besides its simplicity, there are other advantages of using SIP for VoIP. The following list details all the main functions performed by SIP from a VoIP point of view:

  1. User Location and Registration: End points (telephones) are able to alert SIP proxies of their locations. This helps SIP to know that which of the end points will be participating in a call.
  2. User Availability: The end points use SIP to determine whether they will be able to “answer” a call. This works in the same manner when a ‘busy tone’ comes up in traditional telephony.
  3. User Capability: End points can use SIP to negotiate media capabilities with each other, for example deciding upon a mutually supported voice codec.
  4. Session Setup: SIP informs the end points that the phone should be ‘ringing’ when there is a connection request waiting to set up the session. SIP is used to agree on session attributes used by the caller as well as receiver.
  5. Session Management: Activities like call transfer, call termination, and changing of call parameters in the middle of a session (Ex: adding a 3rd party to make a conference call) are handled by SIP.

Conclusion

SIP and VoIP go hand in hand and makes it an amazing calling environment. The easy functionalities that SIP provides to VoIP are unmatchable. Using SIP for all your IP telephony needs is a must. It helps in having a reliable, easy to use and secure connection.

2 replies
  1. David Herrera says:

    Really informative post Arsh. SIP is highly suited for VoIP based communication and offers a lot of benefits. While H.323 was good too, SIP with its easy usage has made it virtually obsolete.

    Reply
  2. Frank Klein says:

    The dual nature of SIP makes it the best for VoIP. The multiple signal and voice lines allow for reliable and quality calls to be made.

    Reply

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