Webrtc wowza example


  • Wowza Community
  • Using #webrtc as a replacement for rtmp
  • Streaming Media Buyers Guide features CoSMo and Millicast
  • Wowza Community

    While it is a very interesting question, I believe the blog post is conveying the cliche the streaming ecosystem as been carrying about WebRTC and that are not longer true. I do not believe wowza to be knowingly deceiving people, i see their point, I just believe that recent advances in the webrtc protocols make most of their statements inaccurate.

    This post is an attempt to document the statements that can be proven wrong today. Fact-checking in some way. As a global comment, while the title states the question as a pure rtmp vs webrtc, lots of the statement deals with global systems which involve recording, storage, and HLS, which is quite different. In this case you might even be able to pre-process your media, doing encoding, chunking, and uploading to the CDN way before it is streamed to an actual viewer.

    Anybody saw that Batman recently? In that case, all the time-sensitive magic happen between the storage CDN and the player on receiving side. Most of the difficult problems are dealt with in HLS by delegating the problem to the transport HTTP , or to the buffer, in all cases, adding delay.

    Moreover, this is not a protocol limitation, as webrtc includes all timestamps and a client web app can use them to sync if need be. There is no real difference there with HLS, a lot depends on the features of the client app, more than the protocol itself. WebRTC Does not scale beyond 1,? Because WebRTC utilizes peering networks, there still has to be a nearby node to help distribute the stream to other local hosts—and peering across a global network can be incredibly difficult.

    From the start, it was about real-time. While the Video conferencing industry and the telcos had experience in real-time media, they did not in dealing with large audience.

    If we take jitsi, one of the best open source webrtc media server out there, as an example: they presented results of their benchmark that showed that once you reach 30 users in the same conference, a normal server would become saturated. Scalability has never been a bottleneck for those who created webrtc in the first place, as a single server would always be plenty enough to support their worse case scenario.

    The scalability limit of one server roughly 1, streams , then became the scalability limit of the protocol. Practically there was nothing more stable, because nobody had ever tried. Now this is not true anymore. Vidyo is most famous for their cascading technology video presentation in , jitsi is rumoured to be about to release a big-conference cluster solution, and some others like red5 pro cluster, liveSwitch, claim to have equivalent solutions.

    The difficulty here is not to cascade the media, but to make sure that the mechanisms that handle bad network bandwidth fluctuations, jitter, packet loss a.

    Contact Xirsys to set up a demo and check both latency and scalability by yourself. Note that RTMP has exactly the same limitation, i. The only difference is that, it s been a long time that people have cracked the problem of having ingress servers, and egress servers with possibly a lot of other stuff in between to deal with that single server limit.

    WebRTC not broadcast-quality? In many ways. Second, VP8 and H. Ericsson had a webrtc stack with H. Moreover, with webrtc or rtmp, the encoding is done on the sender side. The problem being addressed here is the following: if I have different viewers with different playback capacity, in terms of bandwidth, hardware, and display size, how can I manage to serve them all with the best quality stream they can handle with a single source. In file-based systems, you will encode different chunks for different resolutions to be served.

    Depending on the available bandwidth, display size, and hardware, the player will chose which chunk to download. WebRTC has the equivalent mechanism, just real-time. The simple version of this is achieved with simulcast multiple separate encoder with different settings , the best version is achieved with SVC codecs. Conclusion WebRTC has evolved very quickly in the past years. It has the capacity today to replace Flash with an even lower latency.

    Given the extremely fast update rate of the browsers 6 weeks , it will get next codecs faster than other protocols AV1 support in Firefox since version Share this:.

    It turns out it is a relatively simple protocol but there are few gotchas which did trip us up however we have come up with an application module that should meet most needs along with a server listener to start applications when the server starts so applications can listen for SAP packets. If you are not completely familiar with SAP, do have a Google, however it is possible that multiple addresses can be announced to, thus the simplest method was to provide properties at the application level so each application can listen for different addresses if needed.

    This code is extremely basic but by default publishes VP8. This presents an opportunity to change the SDP data. If you look through the SDP data, it is conveniently logged to the console in your browser, then you can see where the video codec is We did a little digging and it appears Wowza Streaming Engine provides access to frames after they have been decoded. There are several posts such as Stack OverFlow and a somewhat more complete example Wowza Code Example This module demonstrates how to perform motion detection on an incoming stream.

    It does require the use of the Wowza Streaming Engine transcoder as it does require access to the decoded frames. It is only a proof of concept module so there is no doubt room for improvement. You will need to configure the module in the Application. This can be done and is extremely simple to do. If you use the Wowza Streaming Engine Manager then this is not possible as it will prompt for a server restart.

    It does require the use of the Wowza Streaming Engine transcoder as it does require access to the decoded frames. It is only a proof of concept module so there is no doubt room for improvement. You will need to configure the module in the Application. This can be done and is extremely simple to do. WebRTC not broadcast-quality? In many ways. Second, VP8 and H. Ericsson had a webrtc stack with H. Moreover, with webrtc or rtmp, the encoding is done on the sender side.

    The problem being addressed here is the following: if I have different viewers with different playback capacity, in terms of bandwidth, hardware, and display size, how can I manage to serve them all with the best quality stream they can handle with a single source.

    Using #webrtc as a replacement for rtmp

    In file-based systems, you will encode different chunks for different resolutions to be served. Depending on the available bandwidth, display size, and hardware, the player will chose which chunk to download. WebRTC has the equivalent mechanism, just real-time. The simple version of this is achieved with simulcast multiple separate encoder with different settingsthe best version is achieved with SVC codecs. Conclusion WebRTC has evolved very quickly in the past years.

    It has the capacity today to replace Flash with an even lower latency.

    Streaming Media Buyers Guide features CoSMo and Millicast

    Given the extremely fast update rate of the browsers 6 weeksit will get next codecs faster than other protocols AV1 support in Firefox since version We are currently scaling up my project with Ant Media. Vendor support is fantastic!

    We are using Ant Media Enterprise Edition 2. Read More The AntMedia team is very responsive and supportive. We highly recommend them. Best of all, I do not have to worry about the technical details of streaming. It is also very easy to monitor and setup streams using their user friendly dashboard. Their support is also worth mentioning. They will never fail to help out with your difficulties. Whenever needed, you can also book a live consultation with them.


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