WebRTC Integration with SIP, RTP, and HTTP to Enhance Communication

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Integration of WebRTC with SIP, RTP, and HTTP to Enhance Communication

 

Explore how WebRTC integration with SIP, RTP, and HTTP to Enhance Communication changes the whole communication market. 

WebRTC is one of the technologies used to provide a platform to build real time communication applications that can support voice, video, and multimedia. If observed carefully, WebRTC application development is focused on a specific type of communication solution. It is a browser based communication technology. Thus, it is not wrong to say that WebRTC development is here to uplift communication standards and make it easier to enhance business communication in real time using the browser. It is definitely not here to replace any other communication solution development technologies. 

WebRTC is quite flexible and it can be integrated with any other communication technology or platform to build a comprehensive communication solution. It can also be integrated with already existing solutions such as a call center solution, IP PBX solution, class 5 Softswitch solution, etc. to support real time communication. 

Three major integrations with WebRTC are with three popular protocols: 

  • SIP: Session Initiation Protocol 
  • RTP: Real Time Transport Protocol and 
  • HTTP based streaming protocols 

Let’s discover how WebRTC integration with these three protocols changes the whole communication market. 

WebRTC integration with SIP 

Integration of WebRTC with SIP is very popular among WebRTC developers, which is why, you can also find a dedicated WebRTC SIP.js developer as well as a WebRTC SIP.js development company. 

In general, WebRTC application development includes both WebRTC and SIP because WebRTC is used to transmit audio, video, and data in real time and SIP is the most powerful signaling protocol, which enhances audio and video communication and makes it easier to build this kind of business communication applications. 

SIP has several advantages over other available protocols, which makes it the first choice of WebRTC SIP.js development company to build robust and scalable communication apps. 

SIP is highly scalable in nature and it comes with built-in security features. It also supports augmented reality to build more engaging apps that stand apart from basic streaming solutions. It is also compatible with a majority of operating systems, and devices such as desktops, laptops, tablets, smartphones, and more. 

A WebRTC SIP.js developer would integrate WebRTC and SIP using JavaScript to build more robust and secure communication solutions that support real time communication and collaboration. 

In general, WebRTC integration with SIP aims to provide a building block for WebRTC app development, so real time audio and video communication can be achieved, but there are several more applications of WebRTC and SIP integrated solutions. With this integration, the developers can build versatile apps for desktops and smartphones, so users with any device can be served/ 

Some of the solutions that use SIP, plus, WebRTC are Facebook Messenger, Google Hangout, and Discord. Discord handles around 14,000,000 callers every single day using WebRTC and SIP. Google Hangout, which is now Google Meet, handles calls, video calls, chat messaging, etc. Likewise, Facebook Messenger provides many more features and functionalities than just instant messaging. 

In general, WebRTC development companies use SIP as a signaling stack for WebRTC solutions to enable real time communication. There are several SIP gateways and codecs necessary to use the full potential of WebRTC. 

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WebRTC integration with RTP 

It is one of the most important protocols in the real time communication solution industry. It helps in sending and receiving media in real time. RTP is very efficient in identifying communication quality related issues like out of order communication packets and jitter in voice because RTP packets will always include timestamps, as well as, the sequence of each packet. This makes it easier to debug and identify issues in the packets. However, it actually increases the size of the packet by adding the overhead of the packet header that includes the sequencing and timestamp. SRTP is a securer version of RTP. It is used along with RTCP by VoIP development companies to build different VoIP solutions. 

The benefit of RTP in VoIP based communication solution development is that it supports an extensive range of media and it can transmit media in real time. This finds an application with WebRTC because this technology is originally formed to support real time communication by exchanging different types of media like voice, video, images, etc. As SRTP is the most secure version of RTP, WebRTC developers use SRTP integration with WebRTC to build different types of communication solutions. 

A WebRTC development company can build different features by using different APIs of RTP and integrating them with WebRTC. Some of the commonly added features are listed hereunder: 

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  • Mute and unmute at the transceiver level and track level 
  • Replace track at the sender side 
  • Remove or add tracking at the receiver side
  • Building a cross browser solution that works on multiple browsers 

WebRTC integration with HTTP 

HTTP is commonly used for streaming online and reducing latency. WebRTC is more efficient in reducing the latency to milliseconds, which makes WebRTC development preferred over all other options. 

HTTP has multiple protocol support for web streaming, which includes the following: 

  • Microsoft Smooth Streaming
  • Adobe HDS 
  • MPEG-DASH 
  • Apple’s HTTP Live Streaming 

The beauty of HTTP and its associated protocol is that it reduces load by cutting down a long media file into smaller pieces and delivering each piece in sequence at a rapid rate, so consumers can enjoy the live streaming the same as if it is a complete video or multimedia file. HTPP is more of a client server oriented streaming. 

On the contrary, WebRTC is web based real time streaming of voice, video, and other data files, which makes it stand apart from HTTP. It is more complicated and advanced in streaming real time files compared to its counterpart. WebRTC can also be used for traditional server-client architecture for streaming if required. As WebRTC is completely focused on real time communication, if the internet bandwidth of the client side is low, then latency would significantly go down, which is a big disadvantage of WebRTC. 

So technically, HTTP and WebRTC are two options available for live streaming, and VoIP development companies can choose any one of these options based on their requirements. There are two driving factors that help in deciding one over another: 

  • Audience size: Both WebRTC and HTTP can cater to a large audience, but HTTP is cheaper when you need to cater to a larger audience. WebRTC used CDNs, which make it pricey to live stream to a larger audience size. With the integration of HTTP and WebRTC, large audiences can be catered to without increasing the cost or latency. 
  • Latency: This is another factor that can define the selection between HTTP and WebRTC. If you have to handle latency around 1 second or less than one second such as 200 milliseconds, then WebRTC is definitely the best choice. But, if your latency is 8 seconds or more or if your audience size is huge, then HTTP is a better choice. With the integration of both of these technologies, you can serve any sized audience with any latency without increasing expenses. 

This is how HTTP and WebRTC can be used together to build more robust live streaming solutions. 

Concluding notes 

Before the emergence of WebRTC, the VoIP world already had different signaling and real time communication protocols: from UDP to RTP. The three most commonly used protocols are definitely: SIP, RTP, and HTTP. Each of these has its own role to play with some key functionalities that make them a better choice to get implemented in VoIP projects by different VoIP companies. 

The launch and popularity of WebRTC switched the game somewhat and made WebRTC more popular than other signaling and communication protocols. WebRTC has emerged as the best technology to develop real time communication solutions that use web browsers to establish and carry out communication. There are already several WebRTC solutions available in the market and more and more are emerging. All thanks to WebRTC development companies. Ideally, WebRTC is a complete platform to build any live streaming or real time communication platform and it does not need any other signaling protocol to establish a connection between sender and receiver. It also does not need any other protocol to transmit packets for real time communication. Moreover, it does not need any technology for live streaming. WebRTC can handle all different aspects of communication. 

Even if WebRTC is a complete solution to build any type of real time communication app, it has some limitations or drawbacks compared to popular legacy protocols and technologies, namely, SIP, RTP, and HTTP. Therefore, WebRTC app development includes the usage of SIP, RTP, and HTTP to build more scalable, robust, and cost effective communication solutions. 

To choose the right technologies to integrate with WebRTC or to choose the correct pathway for building a sustainable communication solution to support real time transmission of voice, video, and data, it is necessary to have in-depth knowledge of SRTP, RTP, SIP, HTTP, and WebRTC. Thus, it is necessary to contact an expert when you are embarking on a venture using the power of WebRTC. We are renowned as the top WebRTC development company. We have developed several apps using WebRTC, plus, the integration of other APIs with WebRTC. For any type of WebRTC development project, contact us. 

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