With the whole world shifting towards video in various applications, your business too will encounter WebRTC sooner or later. The choice that you must make here is whether you should build the native WebRTC services yourself (inhouse) or have a Cloud Communication platform-as-a-Service (CPaaS) provider to manage that for you. The native WebRTC API can meet the basic requirements like one-to-one communications in video calls etc. However, adding on various features or making it practically more feasible can be a challenge.
Here are 20 reasons why native WebRTC without the support of a CPaaS provider might suffer from some roadblocks-
Challenges in Supporting Various Browsers
While the native API might not work well with all browsers, a WebRTC platform provider handles all the browser differences and ensures that your WebRTC application continues to work consistently across all possible browser combinations. For example, different browsers handle the session negotiation (i.e. SDP) differently, have a different ramp up/ramp down behaviour for WebRTC sessions and may have different API for dealing with muting/unmuting streams etc. Your CPaaS provider will take care of this for you wherein the native WebRTC API may struggle.
Limitations Because of Peer-to-Peer Technology
WebRTC is built on a peer-to-peer (mesh) topology and to enjoy a multiparty call (i.e. 6 participants and above), a server-based topology such as MCU (Multipoint Conferencing Unit) and SFU (Selective Forwarding Unit) is required. A server-based topology receives and sends video streams to an intermediate server, thus reducing the bandwidth requirement. Even if the number of participants is then 6, we can’t assume that all the participants are on a high-speed and uncongested network. Often, the participants will have their video call degraded (in the form of frozen video and sound problems) when one faces limited bandwidth. Only a server-based solution can ensure that video quality degradation happens only to the endpoint, which has a bad network.
Read this article to know more – 5 considerations for developer to create video with WebRTC
May Struggle with Bandwidth Adaptability (Simulcast)
Further to the above, this ability to forward the best quality stream to the recipient is called Simulcast. It makes video encoding and forwarding more flexible by generating different versions of the same stream (based on the resolution) and then using SFU’s intelligence to decide which stream is forwarded to which user depending on the end-user bandwidth. Hence, it is critical to choose a service provider with simulcast capabilities to avoid an unnecessary surge in bandwidth requirements.
May Not Have the Best Mobile Experience
It is highly recommended that you go for native WebRTC app for better video quality and easy navigation. However, integrating WebRTC into your mobile application requires you to have an in-depth understanding of WebRTC. Leveraging on CPaaS eliminates these challenges. In addition, a CPaaS service provider can handle the different interfaces required for Android and iOS.
Innovation within the Mobile can be Tedious
Further to the above point, a platform provider can build on more innovative features like Bluetooth-based audio/video connectivity on mobile devices which for a native WebRTC API to do will require a lot of expertise.
Time to market might be slow
CPaaS providers pre-designed APIs for different use cases – large scale broadcasting, group video conferencing or one-to-one call – shorten the need for complex coding and as a result, simplifies your integration and improve time to market.
Building from scratch can be complex
Building ground up using WebRTC can be complex. Apart from ensuring scalability and interoperability, one has to validate signalling logic, media connectivity and more; a CPaaS provider provides you with the carrier-grade platform and the necessary building blocks to pick and choose the features and communication functionalities required. This will also mean that your initial development cost will come significantly down at the time of the production of the app.
There is a limited scope of innovation
A platform provider can help you add on more useful and innovative features to your stack like Active Talker. Active Talker automatically sends the video streams of the most recent speakers to the display window. It allows for a customisable layout and switches to auto-mode when required. It also provides scalability to your video sessions. If you were to develop this capability on the native WebRTC API, it would require immense expertise and resources, hence, costing you money.
Latency could be an issue
Most platform providers offer a globally distributed cloud-based solution. With a global footprint of data centres, endpoints can be automatically connected to the nearest data centre for best performance – best video and audio quality and lowest latency possible. It will not be feasible and economically possible to have a globally distributed network if you would develop real-time communication on your own.
The requirement of hardware
While WebRTC is an open-source and free technology, you may still need to invest in hardware such as media servers for scalability. On the other hand, the ‘pay as you go’ pricing feature of platform providers ensures that you pay for only what you use and does away with the need for any upfront investment.
Recording function not native to WebRTC
Recording is not natively available in the WebRTC stack and can be a challenge to add the functionality into your application. For an application that needs recording, you may either set up a client-side or server-side recording. Given the complexity of building a recording solution, many CPaaS providers do not offer such features. Do carefully evaluate your CPaaS provider if the recording is a “must-have” feature for your application.
Read more about Recording here – Recording WebRTC-enabled session: Server-side or Client-side?
Transcoding requirement for single-stream recording
WebRTC is a peer-to-peer technology using VP8 as an underlying codec. For recording a session between two or more participants, you’d need a transcoding solution. This solution mixes all the participant media streams and transcodes them to a more common format such as Mp4 which can be played in various devices with ease. A platform provider can provide you with this solution as a native capability to your WebRTC solution.
The requirement of the TURN server
When your application is trying to communicate with different clients, there may be firewalls blocking the communication. When a WebRTC session is trying to interact from behind a restricted firewall, they often needs a TURN server. A platform provider usually hosts its own TURN server to ensure that such endpoints can connect with such restricted networks.
Auto Fall-back is required for a better video experience
Often, communication gets disconnected when the network falls below the required bandwidth. Leveraging a CPaaS provider could address this issue. Some CPaaS providers offer the auto-reconnect/ fall-back functionality, which intelligently monitors each participant’s available bandwidth and automatically switches to audio when it falls below a minimum threshold. The video stream is re-established when the network improves, ensuring uninterrupted communication.
The requirement for Call insights
Platform providers offer a dashboarding facility that allows you to analyse the pre and post-call insights in order to troubleshoot and rectify errors. This dashboarding in real-time ensures efficient monitoring of sessions.
The need for SIP interconnection to reach PSTN users
While web browsers and native apps tend to be the primary focus of modern communication, we cannot ignore a large handful of users who will be using a regular mobile or landline. Having a SIP interconnection is necessary to connect to these groups of users. Only a few CPaaS providers offer SIP and PSTN connectivity. If you need to reach a wider audience who are on PSTN, you need to work with Carriers for interconnection or look for a CPaaS provider who has this functionality.
Ensure Security and Compliance
WebRTC is inherently safer in several ways. Because it does not use plug-ins, it eliminates one possible vector of attack through malicious plug-ins or malware. In addition, make sure you’re aware of relevant regulations and compliance requirements when developing your apps. For example, healthcare organizations require Health Insurance Portability and Accountability Act (HIPAA) compliance for patient personal data privacy.
Updates and patches are required
WebRTC platform providers can provide you with the latest and most optimized WebRTC stack, under their API. This is especially important for native mobile applications built on the WebRTC stack. If you integrate the WebRTC stack on your own in your native application, you need to continuously optimize, upgrade and merge the latest WebRTC stack to stay up to date with the latest fixes.
Domain Expertise is key
CPaaS providers obviously come with immense domain expertise which can help you in many ways. Their deep understanding of the WebRTC stack and WebRTC applications make for the most optimized WebRTC based solutions. If you were to build your own solutions, the same expertise might cost you considerable money and resources.
The need for redundancy friendly architecture
Most CPaaS providers build their platform on a High-Availability, redundant, clustered architecture keeping the application downtime to a minimum. This might be resource intense and uneconomical if you would host WebRTC on your own.
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