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How Voice And Video Call Works in Android?

  • Last Updated : 25 Aug, 2021

Voice over Internet Protocol (VoIP) is a common technology for voice and video communication over the internet. This Geeks for Geeks article will go through how voice and video calls function on a high level. We all utilize audio and video on services such as WhatsApp, Skype, Messenger, Facebook, and so on.

Essentially, both voice and video calls are dependent on how we transmit material between two clients that are linked to each other. As a result, there must be something capable of media streaming from one client to another. WebRTC is a free and open project that delivers Real-Time Communications (RTC) features to browsers and mobile applications via simple APIs. The WebRTC components have been tuned to suit this function to the greatest extent possible. However, there are many more things that must be done because WebRTC alone would not enough for full deployment.

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Other Items to Consider Are:

  1. Signaling
  2. STUN Server
  3. TURN Server


What Exactly is Signaling?

To establish a call between two clients, each must conform to the other by exchanging crucial data, messages, and media metadata. We do these things through over-signaling.

GeekTip: Signaling may be accomplished via WebSocket.

It is just used to indicate that these two clients want to connect for the call.

Peer-to-Peer Communication

Following signaling, we must link both the client peer to peer. And we’ll need the public IP addresses of both clients to join. So, we utilize the STUN Server to obtain the public IP address. STUN Server is used to obtain the public IP address.

Why is a Public IP Address Required?

A public IP address is an Internet address that is globally unique. A public IP address can only be assigned to one device. A private IP address is one that is not globally unique and can reside on many devices at the same time. A private IP address is never linked to the Internet directly. Devices with a private IP address will have their own IP space (e.g. different companies or domains).

The NAT (Network Address Translation) gives the device’s local IP address, which cannot be used publicly to join peers. In order to use WebRTC, we must have a public IP address. STUN Server is capable of doing so.

If everything is in order, we obtain the public IP addresses of both clients and then link both clients through WebRTC to begin the connection. WebRTC can handle any type of media.

If we are unable to obtain the public IP addresses of both clients. Then we won’t be able to connect peer to peer. In that situation, the TURN server is required.

TURN Server

TURN Server is used as a mediator to link both clients if peer-to-peer fails. Essentially, it takes data from one client and transmits it to another. So its role is to convey information from the media.

As a result, the two clients begin to converse with one another. Other minor data that is not connected to media, such as a client cutting the call, setting changes, messages, and so on, are delivered through the signaling process. The following thought occurs to us.

Why Can’t WebRTC Do Signaling?

Answer: In order to reduce redundancy and promote compatibility with existing technologies, the WebRTC Standards do not specify signaling techniques or protocols. WebRTC is designed with media in mind. As a result, the voice and video calls function properly.


Voice over Internet Protocol (VoIP) is a technology that enables you to make voice calls via a broadband Internet connection rather than a traditional (or analog) phone line. Some VoIP services may only enable you to contact other VoIP users, whilst others may allow you to call anybody with a phone number, including local, long-distance, mobile, and international lines. Furthermore, while some VoIP services require you to utilize a computer or a specific VoIP phone, others enable you to use a regular phone linked to a VoIP adaptor.

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