If you’re looking to create the ultimate real-time engagement your app needs, look no further. Talk to us and we’ll gladly deliver our optimal RTMP streaming services with our case-oriented solutions, best practices, and commitment to low latency, data security and user privacy.
Real-time engagement
Make use of our video, voice, chat SDKs to create the communication features that you want your users to enjoy. With this, they can easily interact, share and connect with one another anytime, anywhere.
Monitoring
We understand that having a consistent knowledge of how our services are benefiting you is crucial. That is why we prioritize monitoring the quality of communications happening in real time. By doing this, we gain important insights into your user experience with our powerful monitoring and analytics tools, so we can quickly resolve any issues that pop up.
Quality extras
With us, you get to create better experiences and amp up the interactions for your users with a wide array of add-ons, such as video effects, whiteboards, and many more.
High availability and scalability
Our team at Nexoyatech provides 99.99% uptime and the capacity to scale high within seconds to handle even enormous numbers of concurrent stream subscriptions.
RTMP, which stands for Real-Time Messaging Protocol, is a type of streaming protocol designed to transfer video, audio, and data over the internet in real-time and in small chunks.
It works with a dedicated streaming server and the Adobe Flash Player. It also usually uses the H.264 codec, a video coding format to record and distribute high-definition videos.
They are especially beneficial for enterprises that make use of platforms such as Customer Retention Management (CRM) and Enterprise Resource Planning (ERP). Such platforms utilize huge amounts of memory. If they are to function at a speed that can support business uses, then a database hosting provider should ideally be capable of delivering high availability and consistent speed.
RTMP is commonly used with various types of media such as live television broadcasts, video streaming, and internet phone services like Skype. When it comes to video data, especially the high-definition ones, the files are huge. But with RTMP, the huge data is divided into smaller chunks and then delivered to a viewer’s screen as a full picture.
Macromedia, today known as Adobe Systems, initially developed the RTMP specification for high-performance transmission of video and audio data. RTMP works in such a manner that it maintains constant connection between the player client and server, allowing the protocol to act as a pipe and rapidly transfer video data to the viewer.
As RTMP is on top of the Transmission Control Protocol (TCP), it involves a three-step process to transfer data. First, the initiator, which is the client, asks the accepter- the server to initiate a connection, the accepter responds, and then the initiator acknowledges the response and maintains a session between both ends. Because of this, RTMP is highly reliable.
To set up an RTMP stream, you’ll first need an RTMP-compatible camera or encoder. You may either own an IP camera, hardware encoder or software encoder. For anyone just getting started, our recommendation is the free OBS software.
Once you choose the source encoder for your RTMP stream, you can select a solution for video transcoding and delivery. The process of setting up an RTMP stream depends on a lot of different factors. For instance, if you’re:
Then, the setup will involve more steps.
However, if you’re looking for an easy and effective process, you’ll need to fulfill two requirements:
Once you fulfill these two requirements, you can easily set up a custom RTMP stream with the help of any software encoder and stream to one or multiple destinations that support RTMP.
If you’re looking for the best protocol for your workflow, it will depend on your use case. It is best that you compare your options keeping in mind the considerations listed below:
It’s a common practice to transcode RTMP live streams into adaptive HLS and DASH formats. So, this may be the best place to start. Additionally, WebRTC is more suited if you require sub-500ms latency, and the good thing is it is suitable for use for both egress and ingest.