SIP Calling – A Guide to Understand How it Works and its Business Use Cases

Over the last few years, businesses have been widely adopting cloud-based SIP calling services over traditional telephone lines. Owing to its numerous benefits and rising acceptance, it is estimated that the global SIP trunking services market will grow at a promising CAGR of ~13% from 2020 to 2030. In the future, as everything will be based on the cloud, SIP will be the new standard for establishing multiparty and multimedia communication over various channels.

What is SIP?

SIP, which stands for Session Initiation Protocol, works by cutting out the middleman; and thus, it provides a direct connection to the rest of the world. In a traditional phone system, three units work together to connect the dialer and receiver. These include the Private Branch Exchange (PBX), Primary Rate Interface (PRI lines), and Public Switched Telephone Network (PSTN). However, a SIP trunk makes the PRI lines redundant and uses the internet to link the PBX to the PSTN directly. This way, it saves money on copper cabling and provides numerous benefits of an advanced cloud-based communication network.

Unlike the traditional telephone connections, a SIP calling account lets us collaborate over video calls, text messages, and conference calls. This advanced system is highly effective for businesses that need to establish direct communication with anyone inside or outside the organization, regardless of their location or technical setup.

While often SIP calling is confused with VoIP, it’s noteworthy that SIP is a protocol that describes how VoIP calls are connected, maintained, and ended.

How does it work?

Now, let’s understand how SIP calling works and its role in VoIP.

SIP helps in establishing a call by sending signals to each terminal. Once a call is established, other protocols monitor the transfer of audio and data between the two phone systems. In this way, SIP doesn’t work alone during a VoIP call.

Among the numerous protocols used in a VoIP call, the session description protocol (SDP) is another important one that transfers session-related information to assist participants in joining or receiving details of the session. Here, you need to understand that voice information is encoded via codecs that translate audio signals into binary data before being transported over the network.

Real-time transport protocol (RTP) is another specialized application layer protocol that helps by encoding packets of audio data and video data in real-time. It works along with the RTP control protocol (RTCP) to exchange information related to the service quality of sessions.

Now, all the protocols mentioned above are transported to their destinations via transport layer protocols, including transmission control protocol (TCP) and User datagram protocol (UDP).

Now, you might be wondering how SIP is significant for businesses if all it does is establish and end calls. This is because the telecommunication industries around the world have standardized SIP as the preferred protocol for VoIP communication because it isn’t involved in encoding and transmitting data.

So, here is everything that SIP does:

1. It establishes, modifies, and ends multimedia sessions like VoIP calls.
2. It helps send voice, data, and video packets to the receiver terminal.
3. Even without using any internet connection, it can establish independent phone calls.

What are the main components of a SIP Network?

There are five main components of a SIP network, which help establish SIP calls. These are as follows:

1. User Agent: These include a SIP network’s hardware, devices, and networking elements, such as mobile, softphone, and laptop. All these elements can initiate, modify, or terminate the calling session.
2. Proxy Server: Similar to a router, a SIP calling server or proxy server is the primary network element that initiates a request from a user agent and forwards it to another.
3. Registrar Server: This server accepts registration requests from user agents and helps a new user to authenticate itself within the network. Moreover, storing the URI and the location of users in a database helps other SIP servers within the same domain.
4. Redirect Server: This server receives requests from the intended recipients and responds with 3xx to the user.
5. Location Server: It provides details about the caller’s location to the redirect and proxy servers.

Business Use Cases of SIP Calling

Many organizations worldwide have been using SIP calling for years, and it has been playing a pivotal role in saving their money and providing many additional benefits, such as reliability, scalability, and customization. Here are some business use cases built on SIP calling.

• SIP trunking helped GoDaddy establish direct and easy communication channels with small business owners.
• Call center software provider, Aircall used SIP domains to direct inbound calls through the conventional PSTN networks. This was an easier and cost-effective method than porting the various clients’ contact details available with numerous telecommunication providers. Moreover, the company also utilized SIP I/O interfaces to use its carriers in high-traffic regions.
• For Weave, SIP trunking was a natural choice that helped it manage its business more efficiently.
• Splice Software improved its customers’ experience by using SIP trunking with its proprietary internal software. It also improved the company’s ability to manage its brand.
• SIP trunking enabled WW, formerly Weight Watchers, to scale rapidly globally without worrying about reliability.

We can conclude that customers want to reach businesses via numerous online and offline channels in this digital era. Therefore, the SIP call is an obvious choice for both established and new companies. After all, it enables employees, customers, and vendors of a business to connect their mobile devices via flexible VoIP apps and communicate more safely.

To avail of all the surprising benefits of SIP calls and if you need a reliable service provider, please contact us.