Introduction to Session Initiation Protocol (SIP)

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Part 1/4 of blog series on SIP – Session Initiation Protocol Part 2 Part 3 Part 4

On many occasions, we have encountered a question from our customers and technology enthusiasts, what is SIP and how does it work? Here is an attempt to answer that question from a technology perspective.

What is Session Initiation Protocol or SIP

Session Initiation Protocol (SIP) is essentially an application-layer control protocol that establishes, modifies and terminates calls or multimedia sessions over IP network. The protocol can be used for various applications including instant messaging, video, voice and gaming among others. It is an efficient and powerful protocol that several organizations use for internal as well as external communications.

SIP, also known as session initiation protocol, is a signaling protocol for managing real-time communications sessions with two or more participants. Because it is transport and device independent, it can work across voice, video, messaging and collaboration sessions linking phones, PDAs, mobile devices, instant messaging clients, etc.

Unlike legacy trunking that requires enterprises to have a physical connection between themselves and the service provider, SIP connects through IP network into any region in any country effectively giving the enterprises to connect each other spanning across the boundaries. With SIP, companies can leverage multiple IP connections to maintain services and the best part is that they need to pay only for the calls made. The typical application areas of SIP include Multimedia conferences, Instant messaging, Internet Telephony, and Control applications.

Akin to other VoIP protocols, SIP addresses the signaling as well as session management functions within a packet telephony network. While signaling enables the call information to be carried over the network, session management provisions the ability to control all the attributes of a call. SIP provides several capabilities such as

  • Determination of the location of target endpoint by supporting name mapping, call redirection and address resolution.
  • Determination of media capabilities of the end-point- SIP aids the identification of the minimum level of common services between the endpoints before establishing the conference calls
  • Determination of availability of the receiver- In the event of unavailability of the receiver, SIP identifies if the called party is already on the phone or didn’t answer and accordingly sends the message to the caller
  • Establishment of a session between the originating and ending parties- SIP establishes a session between the parties and also facilitates changing of codec or media characteristic and also the addition of a new participant midway through a call.
  • Transfer and termination of calls- SIP facilitate a transfer of calls from one endpoint to another by establishing a session between the caller and the new endpoint and at the end of the call terminates all the sessions between all the parties.

SIP offers several advantages over traditional trunking systems. They facilitate cost savings by leveraging centralized SIP trunking to connect the IP phone systems in enterprises spread across various geographical locations. It also facilitates enhanced flexibility in purchasing and managing bandwidth. The IP based nature of SIP enables easier integration with unified communication applications such as instant messaging, collaboration and video conferencing.

The Pros and Cons of SIP

SIP trunking is a great way of delivering voice and other unified communications features over the internet. Together with an IP-enabled PBX system, SIP eliminates the need for PRI lines. SIP trunking can also be paired with an on-premises PBX or a cloud-based solution.

Pros: SIP services are usually significantly less expensive than PRI lines, saving some customers up to 60% of their communications costs. Many SIP trunking services let customers subscribe to channels in increments of one, making it possible for them to purchase and pay for only exactly what they need. What’s more, SIP channels can be added on-demand without the need for additional equipment. Customers can add (or subtract) channels so that their subscription always matches their current needs.

Cons: As we mentioned, most businesses today have access to internet connections that will work perfectly well with SIP trunking, but some do not and will need to stick with PRI. SIP trunking does require an IP-enabled PBX system, but it is possible to use an older PBX or key system with SIP. It simply requires an inexpensive device known as an analog telephone adapter (ATA). We wouldn’t call this a “con” per se, but it is important for customers to know that there are a large number of SIP service providers out there. Reliability, pricing, subscription options and customer service vary greatly, so it is important to be careful when choosing a SIP trunking partner.

The number of businesses using SIP trunks continues to grow, while the number staying with PRI is shrinking fast. This trend will continue as more and more businesses have access to high-speed internet and decision makers become more comfortable with the cloud approach. PRI lines are still an important part of the communications network and they are the best choice for some businesses, but the advantages of SIP trunking are compelling for many.

DoubleHorn iPhone SIP Trunk ensures the necessary translation services required to enable communications while delivering phenomenal savings. Whether you use your existing high-speed internet connection or you take advantage of the DoubleHorn Core Network and one of our broadband offerings, you can expect a robust feature rich solution. Contact us at 855-61-VOICE (86423) or solutions@doublehorn.com.

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