SIP Trunking VoIP with WebRTC SDK: Transform your Business Applications Across Multiple Platforms

Think of all the times you have thought of exactly how the phone lines work. What goes on behind the scenes of those phone lines used by multiple users connecting each other over a network? How do the messages go through those long phone wires?

When the Internet replaced phone lines, a new system was required to deliver voice and multimedia communications. Enter Session Initiation Protocol (SIP), an application layer protocol that allows you to run your phone networks through an Internet connection. And the system that manages all the phone lines used by multiple users connecting them to a telephone network is called Trunking. Together they are called SIP Trunking.

According to a 2023 report by Lumen Assets, SIP Trunking has helped businesses save up to 65%, i.e., almost $39,000 per year.

How Does SIP Trunking Work?

SIP Trunking is an IP-based infrastructure that allows businesses to establish and manage telephone calls, video conferences, and other real-time communication sessions. It is the digital equivalent of analog phone networks for businesses to unify digital communications.

Simply put, it enables your organization to upgrade its offline Private Branch Exchange (PBX) systems and digitize the phone lines with the power of the Internet. Three changes managers can experience  by leveraging SIP Trunking to empower their communications:

1. Enhance Voice over Internet Protocol (VoIP)
2. Save infrastructure costs
3. Improve call quality

SIP Trunking is a revolutionary technology that can bring about massive transformation by replacing Private Branch Exchange (PBX) systems with Internet-powered phone systems. How? Let us explain through an example.

Traditionally, your office building might use physical telephone lines, such as primary rate interfaces (PRIs) or analog lines, to connect their PBX systems with Public Switched Telephone Networks (PSTN). PBX connects internal telephone lines and enables internal and external communication with PSTN.

SIP trunking replaces these physical connections with virtual connections over an Internet Protocol (IP) network. Think of it as an HTTP – the central technology driving browsing web pages. However, SIP has extended capabilities than only commercial phone services. It also transforms your messaging functionalities through Voice over Internet Protocol (VoIP).

The primary purpose of using SIP is to replace PRIs – a technology that works on Time-Division Multiplexing (TDM). TDM is a traditional system enabling multiple voice conversations channeled through a single physical connection. However, it’s not cost-effective, and that’s why so many organizations have switched to SIP. It replaces their TDM services by delivering SIP through Ethernet and reduces costs. Moreover, it simplifies the complexity of managing legacy systems and providing additional services.

How Does SIP Establish VoIP Session?

Implementing SIP trunking needs a business to integrate a compatible IP-based PBX system or a session border controller (SBC) to manage the connection between the IP network and the PSTN. A reliable internet connection with sufficient bandwidth is crucial for quality and uninterrupted communication.

SIP protocol dominates the IP telephony to establish a VoIP session in any of the below scenarios:

1. When there are two persons on a call
2. When there are more than two persons on a conference call
3. When you are in a video conference

Here is a three-step process:

SIP trunks are intermediaries between your organization’s telephone system and The Internet Telephony Service Provider (ITSP). Earlier companies used ISDN circuits to install on-premises. However, now SIP trunks take over and send them over data networks.

SIP brings a cost-effective way to assign and manage your business phone numbers with Direct Inward Dialing (DID). It allows you to have multiple phone numbers with a specific SIP phone. Instead of having multiple lines, your service provider will send an incoming call over SIP Trunk.

SIP trunks are extremely helpful for provisioning Voice over Internet Protocol (VoIP) to establish a connection with PSTN and on-premise phone networks. SIP Trunking leverages the signaling protocol to bring modifications in real-time sessions between parties. It allows businesses to transmit voice, video, and other multimedia data over an IP network, enabling more efficient and cost-effective communication.

A SIP trunk enables a business to make long-distance calls over the Internet without using traditional phone lines. Your business can leverage SIP for adding or modifying new phone lines without any hassles or the need to invest in new hardware. Hence, SIP trunking saves costs and lets your businesses benefit from cloud services. Why? Because a cloud system is simple to monitor and track critical activities like call frequency, length, and quality.

Here are some notable benefits of SIP trunking:

1. Cost savings: SIP trunks eradicate separate voice and data connections, hence reducing infrastructure and maintenance costs. You might pay more for long-distance and international calls with traditional phone lines. However, with SIP trunk, you save a lot because it’s changed on a per-user basis, allowing you to predict your recurring costs. Plus, you can also choose from metered or non-metered plans.

2. Scalability and flexibility: SIP trunks can quickly scale up or down to accommodate changing communication needs without requiring additional physical lines. They allow for integrating various communication channels into a unified system, such as voice, video, instant messaging, and presence. Since it creates a unified virtual system, they also allow business and individuals to choose their availability. Your phone calls can be directed to the next immediate person or routed through a smartphone. It’s a centralized communication system that grants a single address to all users.

3. Reliable and Recoverable: Since SIP trunking operates over an IP network, businesses can establish virtual local phone numbers in different geographic locations without physical presence. It enables quick and flexible rerouting of calls in case of network or system failures, ensuring business continuity. Traditional systems can face phone failures through damaged phone lines or bad weather. SIP trunking and VoIP are highly reliable combinations because you can route your calls even when there’s a failure.

How Does SIP Trunking Come into Play with WebRTC Applications?

SIP and WebRTC are two crucial components for enabling real-time communications. On the one hand, SIP is a protocol for initiating and managing communications, and the latter enables real-time audio, video, and data communication within web browsers. Businesses can leverage the highest potential for signaling and call control when combined. They both enable a versatile and interoperable environment. Here’s an overview of some significant technologies used in WebRTC:

Signaling helps in modifying, terminating, and establishing communications. It also helps set up calls and negotiate media and session controls.

In a scenario with SIP and WebRTC, the WebRTC browser acts as a user agent to handle media streaming, rendering, and APIs to access input and output functions.

SIP servers also perform call routing and address resolution functions by negotiating communication between parties to determine optimal codecs and transport protocols.

After establishing a session, WebRTC uses its in-built mechanisms to create a secure transport between parties with three main protocols:

• Real-time Transport Protocol (RTP) for media streaming
• Secure Real-time Transport Protocol  (SRTP) for encryption
• Interactive Connectivity Establishment (ICE) for firewall traversal

Using SIP in WebRTC SDK Across Multiple Platforms

SIP in WebRTC SDK applications is a powerful way to revolutionize browser-to-browser communication without extra plugins. It facilitates enhanced voice and video communication. It’s an open-source platform to facilitate communication on web pages. With the SIP protocol in place, it empowers more than audio and video functions. Some notable examples of using SIP with WebRTC include:

Google Hangouts for video, audio, and messaging across multiple platforms for websites and applications.

Discord’s 14,000,000 callers are supported by WebRTC in-app messaging and unlimited calls.

FB Messenger offers calling functions better than VoIP by integrating WebRTC peer-to-peer technology for multimedia connectivity.

Integrating SIP in WebRTC SDKs for Android, iOS, Mac, and Windows empowers applications by enabling real-time communication capabilities within these platforms. Here’s how SIP in WebRTC SDKs enhances applications:

1. Voice and Video Calling: By incorporating SIP in the WebRTC SDK, applications can offer voice and video calling functionality. Users can make and receive audio and video calls directly within the application, leveraging WebRTC technology for high-quality real-time communication.

2. Instant Messaging: SIP integration allows applications to include instant messaging capabilities. Users can exchange text messages, multimedia content, and other data in real time, enabling seamless communication within the application.

3. Presence and Availability: SIP in WebRTC SDKs enables applications to provide presence and availability information. Users can see the online/offline status of their contacts and their availability for communication, enhancing the overall user experience.

4. Call Routing and Transfer: Applications can leverage SIP to implement advanced call routing and call transfer features. Users can route calls to different destinations based on specific rules or user preferences, such as other users, external numbers, or voicemail.

5. Call Conferencing: SIP integration allows applications to support multi-party audio and video conferencing. Users can initiate and join conference calls with multiple participants, facilitating collaboration and remote communication.

6. Integration with Existing SIP Infrastructure: Many organizations already have SIP-based telephony systems. By incorporating SIP in the WebRTC SDK, applications can seamlessly integrate with the existing infrastructure and provide a unified communication experience for users across different platforms.

7. Interoperability with SIP-based Systems: SIP in WebRTC SDKs enables interoperability with other systems and services. Applications can communicate with SIP-based PBX systems, gateways, and telephony networks, expanding their reach and compatibility.

8. Customization and Extensibility: SIP in WebRTC SDKs allows developers to customize and extend the communication capabilities of their applications. They can tailor the user interface, add features, and integrate with other APIs and services to create unique communication experiences.

By combining the power of SIP and WebRTC, SDKs for Android, iOS, Mac, and Windows empower applications to offer real-time communication features, enhancing user engagement, collaboration, and overall functionality.

Choose Moon Technolabs to Leverage SIP with WebRTC SDKs and Develop Robust Applications

Are you looking to enhance broadband connectivity and skyrocket your business communications system with real-time connectivity?

Moon Technolabs brings the power of WebRTC SDKs for all your Android, iOS, MacOS, and Windows applications. Our SDKs enable any business application to work seamlessly with your legacy SIP server. Our developers can integrate our mobile SDKs with your applications even if you don’t have existing infrastructure. We use an extensive range of libraries like WebRTC + PJSIP and self-built SDKs.

We follow MVVM architecture with the MYSQL database and enhance your communication for video streaming, webinars, and conferences with XCode and VCode.

Embellish your websites and applications with cost-effective digital solutions and gain a competitive edge.