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How to use SIP with WebRTC Application?

Session Initiation Protocol is basically a text-based protocol which controls and signals both the voice and video chats/calls that is used in the internet telephony through VoIP technology. The main purpose to develop SIP protocol in the first place is to create and collaborate the real-time communications in an efficient way. SIP facilitates a unified communications to their end-users over IP-PBX platforms.

Before looking at the technical details of how SIP protocol works let us understand the SIP and WebRTC concepts.

Understanding SIP and WebRTC Technologies

What is WebRTC & SIP

WebRTC signaling provides an easy browser to browser communication platform without using any separate plugin that provides excellent voice and video communications in a seamless way. Also, WebRTC signaling is an open source platform which provides the media communication to work within the website pages. In 2016 it was estimated that the number of web applications that embedded WebRTC into their browsers is around 2 billion which is a significant number. Though WebRTC integrates SIP protocol for audio/video communications it can be used to do much more functionality.

A SIP user typically accesses these SIP services usually through a VoIP which is accessed either through a mobile application or a PC. In WebRTC, the users access the WebRTC services like the WebRTC text chat for android or any other services in a traditional browser.

Whereas SIP is a signaling protocol which is mainly used for voice and video calling, WebRTC provides a more versatile option to the end-user which offers SDKs to build powerful mobile applications as well as web applications so the users can literally implement it anywhere. Apart from WebRTC video call in android phone or WebRTC voice chats in an iOS phone is made possible by the portable source code of WebRTC and it also provides webinars no matter where the client and the user are geographically put up!

Why SIP Needed?

Though there are several signaling methods, SIP has several benefits over their counterparts. Let us briefly look into some of the advantages of SIP protocol.

Compatibility

– As SIP is an open standard it is compatible with most of the devices including but not limited to desk phones, tablets, laptops, and much more devices.

Augmented Efficiency

– SIP facilitates the augmented reality, which is gaining popularity in recent times. Augmented reality successfully implements the virtual image over the real world object that receives the input either through smart glasses or camera.

Scalability

- According to the “Journal of Computer and System Services” from Elsevier, SIP protocol is accepted as one of the promising signaling protocols which offer great flexibility, scalability which has built-in security features that increases the overall performance of the real-time communication irrespective of the n number of users.

Provides Easy Readability

- SIP packets are easily readable and it is simple to debug as well which efficiently controls the new services in a better way.

Cost-Effective solution

- The SIP setup fees with new phone lines and porting fees is comparatively low when compared to other signaling protocols. This makes the SIP protocol a more affordable solution. Also with cloud SIP trunking, there is no upfront investment is necessary where it does not require any legacy telephone lines in order to connect any public or private network.

How SIP Protocol Works?

Basically, SIP is the backbone of any VoIP technology which became the recent household name for all kinds of telephony devices right from desktop phones, softphones to smartphones as well. SIP was not only used for audio/video calls but also designed to streamline any other kind of communications like configuring a gaming session or operating a coffee vending machine and so on.
SIP basically contains three types of components for any call flow.

User Agents

When a user calls through any VoIP applications either through a software application or VoIP phone, the users communicate with the help of VoIP getaways through an application server or through any public switched network(PSTN)

Proxies

Next, the role of a proxy is to perform a certain logic where these proxies may either forward or reject a request according to the user’s profile.

Registrar Servers

The sole purpose of the registrar server is to combine the current IP address to that of the user’s VoIP address and also helps to maintain the location database.
Also, apart from these components, the three most common type of SIP requests are,

  • REGISTER
  • INVITE
  • BYE

As the name indicates the functionality of these requests are pretty straightforward where the REGISTER requests indicates the SIP server, the SIP’s phone location address so that it can easily forward the request to the appropriate location. The INVITE request indicates the dialogue initiation between two users and finally, BYE request is the termination of this dialogue.
sip integration works

Role of SIP in WebRTC

The Signalling Plane

Depending upon the existing infrastructure, the SIP can be either added as the signaling stack for the WebRTC application or it can be added in a signaling gateway if the WebRTC application uses any other signaling protocol.

The Media Plane

In order to integrate the SIP protocol into the WebRTC applications , if there is an already existing SIP infrastructure then we must add an additional media gateway known as Session Border Controller that enacts as a gateway between WebRTC and VoIP endpoints or if there is no SIP infrastructure then choosing a WebRTC compatible SIP technology which has many SIP gateways and SIP trunking services is an optimal solution.

Conclusion

Hence integrating SIP into WebRTC web application provides a robust platform which prevents a lot of complexity by providing a simple solution. Though WebRTC can be implemented with any other signaling protocol, several benefits of SIP-like greater scalability, compatibility and cost-effective features and so much more makes it a great combination in providing the user with a reliable media communication platform. WebRTC has successfully incorporated SIP protocol which does not require any additional plugin to utilize the audio/video or file sharing. Thus investing in top-notch technologies increases the overall revenue of the product.
WebRTC & SIP

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3 Comments on "How to use SIP with WebRTC Application?"


Guest
Farooq
7 months 10 days ago

What are the applications that demonstrate the power of SIP & WebRTC?

Guest
David
7 months 10 days ago

How to use WebRTC to create a direct webcam communication application?