What is SIP Trunking? How It works ?

What is SIP Trunking? How It works ?

SIP Trunking is a paradigm shift to the digital revolution across the Globe. While this "SIP Trunking" highly offensive term sounds like it could be related to astrophysics, it’s really just a name for an internet-based replacement for traditional phone service. The SIP part refers to the fact that the service makes use of the SIP protocol for real-time communications over the internet .We can’t blame the internet for overlay technical jargon in this case. Phone companies have been baffling buyers with terms like Trunking for over a hundred years.

SIP trunking is a service that connects internet-based VoIP systems with the phone network, thereby eliminating the need for traditional landline or digital phone services. A SIP trunk isn’t a physical phone line, but rather a service provided by SIP providers like Tata Communications, TTBS, Reliance Com /GCX, Reliance Jio, Vodafone Idea, Bharti AirTel over your internet connection. Customer can use broadband internet to connect his VoIP phone system to SIP provider’s like Tata, Vodafone, Reliance network. The provider can then connect calls to and from the traditional phone network with extensions on your phone system using Dedicated MPLS/ Private VPN last mile connectivity. 

SIP doesn’t encode, decode, or transport any information during these sessions. That’s why it can be used for video conferencing and instant messaging as well as making phone calls over the internet. We’ll leave the other uses of SIP aside for now and focus on how the protocol works during a voice call. SIP is a media-independent protocol. it’s not voice, it’s not video, it’s not data, it could be anything. While it’s mostly applied to VoIP, it’s not a VoIP protocol.

SIP simply initiates and terminates an IP communication session, which could be a voice call between two people or a video conference between a team. It sets up the session by sending messages in the form of data packets between two or more identified IP endpoints, also known as SIP addresses. Every SIP address is linked to a physical SIP client ( IP desk phone) or a software client (softphone).

SIP trunking is a service offered by a communications service provider that uses the protocol to provision VoIP connectivity between an on-premises phone system and the PSTN. The growing adoption of unified communications & collaborative solutions driving the demand for the SIP trunking services market. SIP trunking is the key to connecting Unified Communications to the global telephone network, users will make more and better use of your existing UC platform. Indian Large Enterprises and SMEs are looking for a one-stop solution to meet their enterprise communication needs i.e. data, voice, video conferencing, instant messaging, etc. to have collaborated across a common platform.

SIP is an application-layer signaling protocol for creating, modifying, and terminating sessions with one or more participants over the IP network. These sessions include telephone calls, multimedia conferences, instant messaging, etc. using audio, video, and data. SIP Trunks provision VoIP connectivity between an on-premise phone system and the PSTN, SIP trunks provide phone service for the entire office so they can reach the outside world. The main role of SIP trunking is to replace PRI technology. SIP is the way you achieve a voice-over IP call. It’s an application layer protocol for setting up real-time sessions of audio and video between two endpoints like IP phones, Soft phones. Simply put, SIP is the technology that creates, modifies, and terminates sessions with one or more parties in an IP network, whether a two-way call or a multi-party conference call.

Unlike older digital voice services like Primary Rate Interface (PRI) service, SIP trunking thus doesn’t actually require you to pay for individual trunk lines. Instead of being priced according to a per-line basis, the service is priced according to the maximum number of calls that your business typically makes and takes at the same time. SIP providers route calls between customer's offices over the internet so customer don’t have to pay per-minute rates for them. Free calling between offices can be a huge cost-saver for businesses, especially ones with locations in multiple countries.

So how much bandwidth is enough to handle both voice and data traffic? According to Industry standard a typical voice call today consumes 80-100 Kbps of bandwidth if the audio data isn’t compressed by codecs like G.711a, G.729, G.722 you can follow these steps to estimate the total bandwidth

Multiply the total number of simultaneous calls customers business makes and receives when you’re at peak call volume by 100. This will give you your total bandwidth for voice traffic. Add this number to the total amount of bandwidth your company already consumes when customer having at peak data usage to estimate how much bandwidth customer will be use for voice and data together. Before being transported over the network, voice information is encoded using codecs that translate audio signals into binary data. Many codecs are used for this purpose, but the two most common are:

G.711 codec: Used for uncompressed digital voice. Audio quality is better than other codecs, but it uses more bandwidth.

G.729 codec: Used for compressed voice. It lowers the audio quality to reduce the amount of transmitted data and the resulting bandwidth consumption.

Encoded packets of audio data are carried by the real-time transport protocol (RTP), a specialized application layer protocol used for real-time streaming of audio and video data. RTP sessions are independent of SIP. RTP sessions run parallel to SIP sessions, unlike SDP, which is a payload of SIP.

RTP works alongside the RTP control protocol (RTCP), which exchanges information related to service quality, including the number of data packets exchanged, number of packets lost, and round-trip lag time. SIP doesn’t work alone during VoIP calls. Several other protocols work along with it to ensure voice data reaches its destination. The session description protocol (SDP) is one such protocol.

While SIP communicates with IP endpoints to exchange signaling details, SDP conveys session-related information to help participants join or receive details of the session. It sends three types of information: session description, time description, and media description. SDP doesn’t transport these details itself. Instead, session descriptions are included as a payload of SIP messages.

The RTP, RTCP, and SIP (with the SDP payload) data packets are transported to their destinations using transport layer protocols. The two most commonly used protocols are Transmission control protocol (TCP) Transports packets in an ordered sequence. For every packet sent, the receiving end sends back a receipt acknowledgment packet. User datagram protocol (UDP) Transports data without detecting out-of-sequence packets or retransmitting lost packets. Packets can not only be delivered in an incorrect order but can also be completely left out. 

At this point, you may be asking why is SIP so important if all it does is set up and tear down calls. Well, the telecommunication industry has standardized on SIP as the preferred protocol for VoIP communication, precisely because SIP isn’t itself involved in encoding and transmitting data. It simply establishes a session over the network. Also, protocols written to support VoIP became obsolete with time, and every time something required fixing, the protocols had to be rewritten, which was a challenge. But SIP helps overcome this challenge. 

Traditional trunk lines have fixed capacities. On a PRI trunk, for instance, customer have 23 channels that can be used for calls at the same time. If Customer business needs to handle 24 calls at once, we need to order a whole new trunk, even if you only end up using one of its 23 channels some of the time. With a SIP trunk, ISP can increase or decrease the number of simultaneous calls that customer can handle almost in real-time, so you don’t pay for lines that you don’t fully use. Customer can increase capacity for a promotional offer or product release or scale back capacity when seasonal demand wanes for certain products

The major reason to move to SIP trunking is savings. Gartner reports that SIP trunking offers savings of up to 50 percent when compared with traditional phone services. Free calls between offices on the same VoIP system. SIP providers route calls between your offices over the internet so you don’t have to pay per-minute rates for them. Free calling between offices can be a huge cost-saver for businesses, especially ones with locations in multiple countries.

Many smaller businesses can also get by without deploying a Session Border Controller. Small firms tend to be less of a target for the DOS attacks i.e. a cyberattack to take down an essential network service, such as Internet access that can be SBCs help prevent. They also have less substantial IT investments in on-premise equipment as well as fewer locations with fewer PBX systems than enterprises do. Smaller companies thus have less of a need for SBCs that ensure interoperability between PBX systems from different vendors.

The process of selecting a SIP provider can involve a bewildering array of technical details. For example, some providers ask prospective clients complex questions about the different types of SIP requests supported by their private branch exchange (PBX) systems. A PBX system is the foundation of a business phone system. It connects extensions within a corporate directory to a telecommunications network, such as the traditional telephone network or the Internet.

An Internet-enabled PBX system, designed for use with SIP trunking is known as an Internet Protocol PBX. SIP remains the protocol format for entry, transfer, reintegration of the audio signal packets into a voice product. In practice, however, the tools can be used for far more than phone calls. SIP is a specific protocol that facilitates VoIP. If VoIP refers to the type of phone calls you’re making, SIP is the protocol used for setting up that call. In other words, the main difference between VoIP and SIP is their scope. VoIP isn’t a single technology but a family of technologies. SIP sits under that VoIP umbrella.

Benefits of SIP trunking :

1. It is easy to add channels to your SIP Trunk requirement to increased calls.

2. The monthly fees to have a number of lines installed at your office drop significantly with SIP Trunks. 

3. There are many SIP Trunk providers and competition has driven down call charges significantly. 

4. SIP Trunks are not bound to a location, so it’s easy to move offices without having to change your stationary or inform your customers.

5. SIP Trunks will eliminate the need to buy and manage VoIP Gateways. 

6. Connecting an IP PBX to SIP Trunks is much easier than via the PSTN.

7. With SIP Trunks, you can easily choose the correct number of channels that you need.

8. Provide better customer service by adding more geographical and international numbers.

© Ganesh Kadam, Solutions Specialist - UCC and Cloud domain