Latency, the Fault in Our WebRTC Communication
Latency, literally, means the delay between the cause and effect. For web technology, it means the delay between a user’s action and an application’s response. In networking terms – the total round trip time it takes for a data packet to travel from user one to user two and vice-versa.
In telecommunications, latency is due to the laws of physics: the length of the pathway that the information or data needs to traverse through the networks.
What does that feel like? Latency causes discomfort to both users. It breaks the rhythm of work. It makes you weary, as the user expects smooth and effortless communication. Even the slightest of lag is noticeable and needs consideration. When there is lag, you speak and then wait while the other user “rogers”.
You would have seen latency at work on television when a field journalist reports to the studio:
– ‘Hey Larry, what is the situation?’
– 2 seconds later… ‘will the protest see an end?’
– 3 seconds later… ‘Am I audible Lary?’
– Hey Stephany, the situation here is worsening, since people are gathering in large numbers and… I can hear you,
– Why is it worsening, Lary? oh, people are gathering
– 2 seconds later…because people are gathering, Stephany?
It is weird, right? Latency in real-time communication just spoils the party for all.
Take another example of virtual reality (VR). While having a VR experience through VR goggles, you need smooth backing from remote serves. A shorter delay will facilitate a more realistic user experience. The same goes for live streaming, online training, and conferencing. The future demands zero latency and technology has to cater to that need.
It’s a no brainer that you want your business communication solution to be future-ready. Real-time Communication is the future! In these fast-paced and connected times, telecommunication solutions must be as smooth as silk.
Latency can be caused by many factors that may or may not be network-related.
There are other non-network related causes of latency like a misconfigured or faulty DNS server, poorly optimized or over-utilized backend database, low memory, CPU cycles, etc. That said, the main culprits are network components that move data from point A to point B. The list of factors includes – the distance between source and destination, QoS policies are not properly configured, network device CPU/Memory spikes, suboptimal routing, unstable wired connections, problems with in-line infrastructure components (network load balancers, firewalls, and intrusion prevention systems (IPS).
No matter how latency forms, the result is the same – an unhappy user. Talking of the delivery of live-streams, WebRTC is the technology making the rounds of late.
WebRTC is the best way to achieve ultra-low latency. RTCweb.in’s WebRTC implementation results in sub 500 ms of latency, which is as good as real-time. WebRTC enables fully interactive live-streaming making real-time communication possible.
Why is WebRTC technology the best solution for low-latency or real-time streaming? Here are some reasons:
- Supported On All Major Browsers
WebRTC enjoys high consistency and stability. It is supported by major internet browsers: Chrome, Safari, Firefox, Edge, and Opera. Therefore, no plugins are needed. WebRTC empowered applications to work directly inside web pages. No matter the device, every browser works.
- No Plugin Required – Flash is not required anymore
The live-streaming technology sector has since evolved and expanded, outgrowing the Flash and RTMP. With the growing demand for live video, Flash’s limitations proved a block to live-streaming.
WebRTC technology came into existence in anticipation of this only. WebRTC creates peer-to-peer, real-time communications directly between browsers via simple APIs. Major players like Apple, Google, Microsoft, Mozilla, and Opera support WebRTC. The technology remains up to date and functional for the foreseeable future.
While RTMP and Flash were used to deliver streams at low latency, WebRTC is capable of providing high-quality performance in real-time.
- Secure Connection – Peer-to-Peer
A peer-to-peer connection creates a secure connection. WebRTC uses DTLS (Datagram Transport Layer Security) to exchange encryption keys and SRTP (Secure Real-Time Protocol) as the transport protocol to send and receive encrypted data.
- WebRTC is UDP based
WebRTC is the only protocol providing sub-500 milliseconds of real-time latency. Unlike TCP based HTTP live streaming, WebRTC is UDP based. UDP does not care about the order of the data, rather it delivers data packets to the application, the moment they arrive. Unlike TCP, WebRTC focuses on the dropped packets instead of queuing packets and waiting for them to load.
- RTP Efficiency
WebRTC transfers videos over the Internet and IP networks using the streaming protocol RTP. RTP sends data in small chunks, each one of which is preceded by an RTP header. The data packets are organized in sequence suitable for transmission between the servers and clients. RTP streams carry the media payload encoded by an audio or video code.
RTP streamlines media delivery by consolidating essential information. When RTP is layered on top of UDP it makes it much faster compared to streaming solutions like HLS or DASH. RTP also reduces latency by sending out the stream by using a push method. We shall discuss this in detail in some other article.
- No buffering
Our implementation of WebRTC does not perform buffering or caching. WebRTC technologies allow the users to exchange information without waiting for a queue of packets and they can just be sent as soon as they are ready. Our quality control methods ensure that the stream performs with good quality even in poor network conditions.
I hope this helped and showed how the right tools can help you keep up with the future. Are you looking for expert services in WebRTC? Contact us today!