SIP Trunking- a paradigm shift to the digital revolution

Calling is a technology that is ever-changing in business, but it remains incredibly important nowadays. In the era of the Cloud and other latest online technologies, the landline phone is going away, but business communications are more important. The Covid-19 pandemic has accelerated the adoption of digital technologies across the world.?

The COVID-19 pandemic has acted as a catalyst for the development of automation technologies by global companies, especially in the area of drones for the purpose of surveillance and medicine delivery, IT industry. This lockdown provides us an opportunity to adopt these new trends, thus increasing the opportunities for the global SIP trunking service market in the upcoming years.

The growing adoption of unified communications & Collaborative solutions in enterprise environments along with the significant reduction in telephony costs and a rapid Return on Investment (ROI) facilitated by the trunking services is 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 Small & Medium-sized Enterprises SMEs, and Industry Verticals (BFSI, Healthcare & Life Sciences, Telecom & IT, Government & Public Sector, Manufacturing, Consumer Goods & Retail, Media & Entertainment ) 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.

The Indian Government's Digital India campaign that trend would provide a strong network for Indian enterprises to integrate voice and data over an IP Network which would encourage them to adopt Unified communications, SIP-based services and leverage the benefits of a strong IP backbone. As enterprises in India are expanding rapidly overseas with branches and offices at foreign locations. Enterprises would look for a centralized connectivity solution on IP background to connect remote work locations in a cost-effective way.?

Evolution of SIP Trunk

SIP Trunking today owes a lot to the initial design of audio conversion to digital packets for network transmission. It represents the refinement of all those steps made earlier to run audio over computer lines, and SIP Trunking is the reason today so many companies can easily handle phone networks in multiple countries across the globe.

SIP Trunking is a service tool that gives a user the ability to run VoIP communications entirely on its Internet network as voice packets in digital form. The two requirements are a PBX system for input and output, as well as an Internet gateway to make connections with. While traditional landline phones connected people via PRI phone lines, SIP allows for the PRI lines that make these work to be bypassed. This saves businesses money since the bills are much lower than traditional landline bills. It also is much more efficient because the connection is of a higher quality than PRI calls have.

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. PRI, or Primary Rate Interface, has been used for decades to deliver lines of voice and data using physical copper lines. Essentially, it's a bundle of analog phone lines put together. PRI can be costly to maintain since the hardware is becoming outdated rather quickly.

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 can invite both persons and services (such as a media storage service) to participate in a session and can also be used to set up calls between PSTN subscribers using gateways.

What Is SIP Trunking?

Session Initiation Protocol, or SIP, is the way you achieve a voice over IP (VoIP) call. It’s an application layer protocol for setting up real-time sessions of audio and/or video between two endpoints (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.

A SIP trunk is the virtual version of an analog phone line. Using SIP trunks, a SIP provider can connect one, two, or twenty channels to your PBX, allowing you to make local, long-distance, and international calls over the Internet. If you have an on-premises PBX in your office, a SIP trunk provider can connect to you and allow you to make outbound calls on your existing system, without restrictions on the number of concurrent calls.

SIP Architecture :

SIP uses two basic network entities namely clients and servers. A client is an application program that sends SIP requests. A server is an application program that accepts and services SIP requests and sends back responses to the requests. Figure 1 shows a logical architecture for a SIP call. The signaling passes via one or more servers while the media stream takes a direct path. User-Agents are SIP network terminals like SIP Phones and Gateways.

A Proxy Server acts as the initial point of contact for all SIP requests. It acts as a server and a client for the purpose of making requests on behalf of other clients. A Redirect Server accepts a SIP request from a client for the called party destination address. A Registrar Server accepts SIP REGISTER requests from a user, indicating that he is available at a particular address in the network. A Location Server is used by a SIP redirect or proxy server to obtain information about a called party's possible location. The Policy Server is designed to use Common Open Policy Service to provide Quality of Service (QoS), bandwidth reservation.

SIP Trunk is an advanced Voice Connectivity product, with the best in class IP solution. It replaces traditional multiple fixed PSTN lines with a single physical link that can support up to 1500 simultaneous calls. Scaling up and down call capacity on our SIP Trunk is very easy, quick, and can be done in multiples of 10 channels. SIP Trunk reduces the cost of multiple lines, maintenance, as well as hardware requirements for multiple PRI ports. Enterprises can opt for SIP Trunk solutions to overcome issues related to scalability, maintenance, and convergence. This innovative, best-in-class SIP Trunk Services has won industry awards and offers easy integration with other IP technology platforms.?

SIP Trunking adoption is at a growing phase and would lay a strong foundation for unified communications encouraging collaboration. However currently due to India’s regulatory landscape; the integration of the Internet with PSTN and centralized SIP architecture is not possible. But this is expected to change with the liberalization of government regulations going ahead in the future.

Growth in mobility, network developments, and foreign expansion of enterprises would encourage enterprise customers to migrate towards SIP solutions moving away from the costlier and less flexible traditional voice ISDN circuits. Trends like BYOD, Digital India dream of government, spectrum advancements, unified communications goal & overseas expansion of Indian enterprises would be the main driving force of VOIP & SIP Trunking services in India. SIP Trunking services also open doors for future technologies like Microsoft Lync calling, WebRTC, Omni-channel approach & wearable technology. SIP Trunking services with their gamut of benefits can change the face of enterprise communication and evolve in a platform to enable unified communications to support high-value applications and result in cost savings

Traditionally, Enterprises used to have a separate infrastructure for fulfilling their data & voice needs. A PBX was connected through trunks, these were connected to a PSTN Network which took it to other branches. This traditional architecture hiked up the infrastructure cost of dedicated single PRI Lines with scalability and flexibility constraints. SIP trunking communication protocol can help in alleviating the problem related to cost as it connects internal IP-PBX to the PSTN and enables to carry out the communication over an IP network. SIP Trunking provides flexibility and scalability by creating and maintaining a user session only when required, thus helping in bandwidth optimization. SIP can be deployed by installing an IP-PBX or a mediation server and communication through the channels is facilitated by soft-switches.

How Does SIP Trunking Work?

SIP is the dominant format used in IP telephony. It’s a protocol that establishes a VoIP session over the internet.?Be on a voice call with one other person, Have multiple people on a conference call, Run video calls.SIP trunks work as an intermediary between your business phone system and the Internet Telephony Service Provider (ITSP). See the diagram below showing how calls flow from a SIP phone (VoIP) in a business to the outside world—it's fascinating.

SIP and VoIP both refer to internet telephony and are technically different. They’re often seen as two different options you can choose from. However, you shouldn’t directly compare SIP to VoIP, and here’s why. VoIP is quite a broad term that refers to any phone call made over the internet instead of the traditional, physical telephone line. VoIP stands for Voice over Internet Protocol.

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. The definition of SIP trunking is a VoIP based on SIP, which enables internet-based telephony vendors to provide telephone services and unified communications to customers with SIP-based IP-PBX.

SIP messages & Responses :

What is the Pilot number & DID/DoD?

A pilot number is an address, extension, or location of the hunt group inside the PBX or IP PBX. It’s generally a blank extension number or one extension number from a hunt group of extension numbers that don't have a person or telephone associated with it

Direct inward dialing (DID) is a telecom service that forwards incoming calls directly to VoIP or to another telephone number anywhere in the world. DID enables callers to directly dial into a phone extension without the assistance of an operator. Direct Outward Dialing (DOD) is a service that enables outbound calls from a DID number. DOD can complement direct inward dialing service to enable two-way voice communication with a fixed caller identification

What is E.164 number plan?

E.164 is the international telephone numbering plan that ensures each device on the PSTN has a globally unique number. E.164 is a global PSTN and data network standard that prescribes how telephone numbers should be arranged for successful routing. It is officially referred to as ITU-T Recommendation E.164. This number allows phone calls and text messages can be correctly routed to individual phones in different countries.?

E.164 numbers are formatted [+] [country code] [subscriber number including area code] and can have a maximum of fifteen digits.

The E.164 standard establishes a common framework that every country can use to create international phone numbers. The standard limits the number of digits that a telephone number can have to 15, excluding the international call prefix. E.164 is designed to adapt the public telephone numbering plan to the demands of the internet age. It makes dialing easier, among other benefits.?

What are audio codecs??

An audio codec is a digital electronic device or computer-based software application that aids in the compression and decompression of a digital audio data stream. Most SIP trunking services use either G.711 codec, which consumes 64 Kbps per call, or G.729, which consumes 8 Kbps per call. There are other codecs that be used, but these two are the most popular.

G711 is the standard codec used in SIP. For better quality of speech, this codec transmits the audio signals without compressing the voice data.?G729 is a reduced quality codec, where the voice data is compressed. G729 is typically seen in scenarios where bandwidth is restricted or unavailable for dedicated voice use.?

What is SBC?

The SBC is a piece of hardware or software that governs how phone calls or sessions are initiated, conducted, and terminated on an IP phone system. It works much like a router does on a data network, sitting between the customer and the carrier network and allowing only authorized sessions to pass through the border. It also provides QoS functions, ensuring that calls go through properly and emergency calls get top priority. Multiprotocol label switching (MPLS) and virtual private networks (VPNs) reduce the risk but an SBC is still an important addition to a Voice over IP (VoIP) system.

An SBC can help boost security by using its QoS rules to identify incoming threats such as a digital denial of service (DDoS) attack. It also offers deep packet inspection, policy enforcement, and other security functionality, providing more control than an application-layer firewall. SBCs are now used to regulate all forms of real-time communications including VoIP, IP video, text chat, and collaboration sessions. SBCs can also improve interoperability between disparate VoIP systems and legacy analog and digital PBXs. Generally, SBCs are deployed on both the customer and carrier side of the connection to improve security. SBCs on the customer side is known as enterprise SBCs (eSBCs).

SIP trunk to PRI converter :

A SIP trunk to PRI gateway, also known as a SIP trunk to PRI converter, connects today's SIP trunking phone services to legacy PRI phone systems. This is essential for businesses to keep older equipment that is still functional. It also helps businesses sign up for a new phone service while being able to use their old phones. Many organizations still have these older phone systems. But they want to begin using VoIP service through SIP trunking.

Few SIP to PRI convertor models used - AudioCodes Mediant 500, Mediant 800, Matrix, Terratel, etc.

SIP Trunk Security?:

One of the ways SIP safeguards the information is by using encryption. This mechanism involves sending SIP data over a transport layer security (TLS) channel. Using this method, the data sent between the PBX and the VoIP phone system will transmit over a secure connection where the voice data will be split into encrypted IP packages.

This form of secure communication is called SIPS or SIP secure. These data packets can only be read using the right encryption key. The benefit of using SIPS is that, in addition to the voice data, the details of both the caller and the recipient are also encrypted. Encrypted SIP is one of the many features that set SIP apart from other telephony systems.

SIP also uses SRTP which is an extension of the real-time transport protocol (RTP). SRTP uses authentication and encryption to minimize the risk of cyberattacks such as denial of service (DoS) and protect against replay attacks. The protocol encodes communication data from the sender into encrypted packages, which can only be decrypted by the receiver via a key.

Benefits of SIP trunking :

1. Flexibility – It is easy to add channels to your SIP Trunk to cope with increased calls. A simple phone call will allow you to add channels, and often this can be done immediately.?
2. The monthly fees to have a number of lines installed at your office drops significantly with SIP Trunks.?
3. There are many SIP Trunk providers and competition has driven down call charges significantly. Some SIP Trunks even come with unlimited calling and you can save on international call charges, too.
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. There is no longer any need to pay to forward phone calls to the new offices.
5. SIP Trunks will eliminate the need to buy and manage VoIP Gateways. All phone calls come in via IP. No extra conversion often means better quality too.
6. Modern IP PBX / Unified Communications solutions will give customers increased productivity, mobility and boost sales. Connecting an IP PBX to SIP Trunks is much easier than via the PSTN. You can go on-premise or host, the choice is yours.
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. Quickly and easily add numbers to your SIP Trunk and terminate them on your IP PBX you can give customers more options to dial in at a significantly lower cost, even long distance. Customers can contact you more easily and sales will increase.

#Reference- Technology White papers, My relevant work experience with Tata communications- UCC & Enterprise domain, Reliance Jio- Enterprise solutions & delivery, and Tata Teleservices- Enterprise solutions.

Ganesh Kadam_Solutions specialist_UCC & Networking domain