Friday, July 2, 2021

Scale up your WebRTC Application in 2021

You might often ask yourself, "How do I scale this application?" You can come up with questions like this all the time to build a successful web-based software app. 

The plan to scale a software app is never overplayed. But most of us leave for the end. Until then, something has already failed in the app. It has become very important for applications to incorporate real-time web communications or any other real-time tech. Real-time communications are the most important for apps with video components. 


So, how does one scale up their WebRTC applications? All it requires is the right pieces, and then you are done.


What is WebRTC development?


webrtc development

Real-time Web communication is an open-source network system. The specification of real-time media communications is possible with the help of this. For example, native voice notes, video notes, and data transfer between browsers and devices can be done. Thus, helping the users communicate easily from their web browser without needing the help of difficult plug-ins of hardware systems. 

Google created WebRTC in the year 2011. This meant a new way of developing common protocol sets that helps in creating high-quality web browsers and mobile platforms for loT devices. 


Before Web RYC came, flashes and plug-in methods were the only methods of offering communication between servers. With the help of WebRtc, the first-ever call between chrome and firefox browsers as possible. Ever since then, the support for WebRTC in the developer's community has grown. More organizations are adding WebRTC development to their support systems. One can use WebRTC as a native platform holder in any browser. WebRTC has also allowed android and iOS to have video calling features. 


WebRTC has a lot of advantages - 


1. Supports multiple platforms and devices in one go. 

Data in this can be transferred between two browsers and web apps for end-to-end communications. Helping in forming better video and voice connections for media servers.


2. Provides safe and secure video and voice calls. 

WebRTC has video and voice authentication encryption modes. This helps with secure connections. Moreover, no communication is recorded even if one uses open wifi networks.


3. High-definition video and voice call quality. 

WebRTC has a fidelity voice opus codec system. This helps with streaming videos of excellent quality. 


4. WebRTC can easily adapt to any network condition.

Since WebRTC application development is a smart technology, it can adapt to any network connection under any condition. This is possible due to the RTP protocols for controlling browser work. 


5. It has voice and video interoperability.

WebRTC has video systems that use SIP, PSTN, XMPP, and other standards, helping in supporting the device-based WebRTC protocols. 


Scale WebRTC applications


WebRTC application development

To start the process, you will need to know how your WebRTC application works and ask yourself some questions - 


1. What is the max number of users you need support for in one session? 

2. How many users can you expect in one session? 

3. Who will be sending videos in the session?

4. How many users will be using media in one session?

5. Where will the users be joining a session? Corporate network or residential?

6. Will there be high-security network connections?

7. What needs to be recorded?


Once you take a deep look at these questions and study them thoroughly, you will be able to study the technical architecture of your application, and it will help you make a decision. Once you know the answers to these questions, you can choose between the platforms - Mesh, Hybrid, Mixing, or Forwarding.


Forwarding architecture - Forwarding architecture depends on the units or SFU. This helps with the media relay between a session. Each client will connect to the SFU once media needs to be sent, then others will follow. This will result in every client managing their n unidirectional connections, where n will be the total number of clients in one session. 


For example, if a session has five clients, then there will be five connections, and each of the clients will have to manage five connections. One of these connections is used for sending media files, and the other four are for receiving media. 


The total number of connections in a session in architecture is n2. Mesh has more scalability. Hence you get a lot more things done with this. Mesh requires a more asymmetrical nature of the internet for clients to upload something. Since there is only one encryption allowed, it will decrease the pressure from the device, especially if you are using mobile. 


The best advantage of forwarding over mesh architecture is that it can be employed using different scaling techniques. This is because forwarding acts as an intermediary between senders and receivers. Thus, forwarding architecture can easily apply bandwidth capabilities of each leg and selectively apply temporal and spatial scaling as well. But, of course, this all happens when it moves through the server. 


Forwarding architecture requires the use of additional servers for handling infrastructure. However, adding servers makes it a highly efficient solution. 


Mixing architecture - Mixing architecture depends on the multipoint control unit system or the MCU. This allows the server to act as a high-power media mixer in the middle of a session. 


Each client in the session connects with MCU at least once. This means each of them will manage a single bidirectional connection system. In mixing architecture, the number of clients does not matter. Connections in this are used to send and receive media from the server. 


Mixing allows users to download and encrypt the data, unlike forwarding, where one only encrypts files. If we talk from the perspective of clients, this will be the most efficient for them. But if we talk from the server's point of view, this is the least effective way. Because it puts much burden on the packetizing, decoding systems, encoding, mixing, and packetizing. 


Many things can be done to optimize the pipeline. The operations that need to take out are still very high and require significant server resources to complete the real-time process. 


Mixing architecture can take full advantage of scaling techniques. All it requires is the application of temporal and spatial scaling to the audio outputs. This helps with adapting to the changing conditions on the network routes for each client's needs. Mixing architecture also maintains the quality of the session as high as it possibly can. 


For webRTC application services with many participants, where devices need to be bandwidth and resource contained, it is a good approach to start from even though it increases the server's expenses. 


Hybrid architecture - Hybrid architecture is a mixture of mesh and forwarding architecture. In hybrid architecture, participants can join a session at their convenience. Meaning they can join whatever makes the most sense for them. 


If it is only two parties, then the mesh setup will be simple and use fewer resources. If the session is a small group session for live events or broadcasts, forwarding architecture will mix your needs. But for bigger group sessions, the practical option would be to use mixed integrations. 


Hardware requirements for scaling up WebRTC apps


webrtc application development


The architecture selection depends on you, but WebRTC developers will need to keep in mind the hardware requirements with all of them. 

  • There should always be a signaling server present in the application. 

  • Turn server for network traversals. 


Turn servers can be managed automatically. With the help of a few signaling signals servers, you can use Turn servers with high availability. While choosing server hardware, the general-purpose should be for compute-optimization servers. For signaling servers, use network-optimized servers.


1 comment: