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• How to best decide what VM to use on google cloud ? Any best practices ? [closed]

  2 juillet 2024, par Prabhjot Kaur

  I have a script that reads google sheet for urls and then records those url videos, then merges it with my "test" video. both videos are about 3 minutes long. I am using e2-standard-8 Instance with ubuntu on it. Then running my script in node using puppeteer for recording and ffmpeg for merging videos. It takes 5 minutes for every video.

  


  My question is that should I run concurrent processed and use a stronger VM that will complete it in lesser time, or should i use a slow one ? It doesnt have to run 24/7, because I only have to generate certain amount of videos every week.

  


  Please provide the guidance that I need. Thanks in advance.

  


  I tried creating instance with more CPUs with free credits and ran out with them fairly quickly. I wonder if there is some other service i could use that will make the process faster ?

  


• WebRTC predictions for 2016

  17 février 2016, par silvia

  I wrote these predictions in the first week of January and meant to publish them as encouragement to think about where WebRTC still needs some work. I’d like to be able to compare the state of WebRTC in the browser a year from now. Therefore, without further ado, here are my thoughts.

  WebRTC Browser support

  I’m quite optimistic when it comes to browser support for WebRTC. We have seen Edge bring in initial support last year and Apple looking to hire engineers to implement WebRTC. My prediction is that we will see the following developments in 2016 :

  • Edge will become interoperable with Chrome and Firefox, i.e. it will publish VP8/VP9 and H.264/H.265 support
  • Firefox of course continues to support both VP8/VP9 and H.264/H.265
  • Chrome will follow the spec and implement H.264/H.265 support (to add to their already existing VP8/VP9 support)
  • Safari will enter the WebRTC space but only with H.264/H.265 support

  Codec Observations

  With Edge and Safari entering the WebRTC space, there will be a larger focus on H.264/H.265. It will help with creating interoperability between the browsers.

  However, since there are so many flavours of H.264/H.265, I expect that when different browsers are used at different endpoints, we will get poor quality video calls because of having to negotiate a common denominator. Certainly, baseline will work interoperably, but better encoding quality and lower bandwidth will only be achieved if all endpoints use the same browser.

  Thus, we will get to the funny situation where we buy ourselves interoperability at the cost of video quality and bandwidth. I’d call that a “degree of interoperability” and not the best possible outcome.

  I’m going to go out on a limb and say that at this stage, Google is going to consider strongly to improve the case of VP8/VP9 by improving its bandwidth adaptability : I think they will buy themselves some SVC capability and make VP9 the best quality codec for live video conferencing. Thus, when Safari eventually follows the standard and also implements VP8/VP9 support, the interoperability win of H.264/H.265 will become only temporary overshadowed by a vastly better video quality when using VP9.

  The Enterprise Boundary

  Like all video conferencing technology, WebRTC is having a hard time dealing with the corporate boundary : firewalls and proxies get in the way of setting up video connections from within an enterprise to people outside.

  The telco world has come up with the concept of SBCs (session border controller). SBCs come packed with functionality to deal with security, signalling protocol translation, Quality of Service policing, regulatory requirements, statistics, billing, and even media service like transcoding.

  SBCs are a total overkill for a world where a large number of Web applications simply want to add a WebRTC feature – probably mostly to provide a video or audio customer support service, but it could be a live training session with call-in, or an interest group conference all.

  We cannot install a custom SBC solution for every WebRTC service provider in every enterprise. That’s like saying we need a custom Web proxy for every Web server. It doesn’t scale.

  Cloud services thrive on their ability to sell directly to an individual in an organisation on their credit card without that individual having to ask their IT department to put special rules in place. WebRTC will not make progress in the corporate environment unless this is fixed.

  We need a solution that allows all WebRTC services to get through an enterprise firewall and enterprise proxy. I think the WebRTC standards have done pretty well with firewalls and connecting to a TURN server on port 443 will do the trick most of the time. But enterprise proxies are the next frontier.

  What it takes is some kind of media packet forwarding service that sits on the firewall or in a proxy and allows WebRTC media packets through – maybe with some configuration that is necessary in the browsers or the Web app to add this service as another type of TURN server.

  I don’t have a full understanding of the problems involved, but I think such a solution is vital before WebRTC can go mainstream. I expect that this year we will see some clever people coming up with a solution for this and a new type of product will be born and rolled out to enterprises around the world.

  Summary

  So these are my predictions. In summary, they address the key areas where I think WebRTC still has to make progress : interoperability between browsers, video quality at low bitrates, and the enterprise boundary. I’m really curious to see where we stand with these a year from now.

  —

  It’s worth mentioning Philipp Hancke’s tweet reply to my post :

  https://datatracker.ietf.org/doc/draft-ietf-rtcweb-return/ … — we saw some clever people come up with a solution already. Now it needs to be implemented

  The post WebRTC predictions for 2016 first appeared on ginger’s thoughts.

• ffmpeg can't recognize an UDP stream

  30 décembre 2014, par yaapelsinko

  When executing

  ffmpeg -i udp://239.192.1.2:3456

  kind of command, ffmpeg seems not being able to read such stream. No metadata info, and no transcoding if appropriate commands given.

  My network layout is the following :

  Ubuntu Server (ffmpeg) <---> Windows Server (Wowza) <---> Multicast subnet

  Stream must come from Multicast subnet through Window Server. Windows is configured to route IGMP via RRAS service. When I launching ffmpeg on Ubuntu, I can monitor that appropriate reports are received by RRAS and UDP stream starts to flow from Windows-to-Multicast network interface. I wasn’t able to monitor Ubuntu-to-Windows network interface, though, because Ubuntu is actually a Hyper-V VM on that Windows Server. Something is preventing Wireshark from listening on virtual NICs. Windows Server also has third NIC to the Internet, but it doesn’t matter here. Stream itself is okay, it can be successfully played with VLC or transcoded by Wowza (all on Windows Server). It is encoded with MPEG2/MP3 codecs.

  If I restream the stream through Wowza (passing through or transcoding), then ffmpeg is able to ingest it from rstp ://windows-server-ip:1935/LiveApp/myStream.stream so that I see metadata report and can transcode it. But I want to get it directly from multicast.

  Is it ffmpeg can’t read directly from udp ? Or maybe I missed something in configuration ? How can I investigate it further and localize the problem ?

  Update : Well, when restreaming the stream via VLC right into Ubuntu server NIC, ffmpeg can grab it. There are another problems, though, but at least I see that ffmpeg receives something. So, IGMP routing is not working correctly.

  Here is what I’ve done when configuring it : Enabled RRAS service. Added IGMP protocol to IPv4 routing. Added pNIC and vNIC as interfaces. pNIC is in Proxy mode, vNIC is in Router mode.

  That way I can at least see : 1) new records in IGMP group table when someone is requesting IGMP membership, 2) UDP packets flooding pNIC multicast interface when request from vNIC is received. However, I can’t listen vNIC interface with Wireshark from guest or host by some reason so I don’t know if packets are actually reaching the player on VM. I assume they aren’t, because I can’t play it with VLC or ingest the stream by ffmpeg (but who knows, maybe it just can’t be played in Hyper-V ?).

  If both interfaces are in IGMP router mode, no UDP traffic can be detected.