The realization of screen sharing function based on Licode demoThis article in Licodeof theBasicexamplebased on addingScreensharingfunction. The main trouble is that screen sharing must beHttpstransfer under protocol, need to modifyErizo Controllerthe code, andHTTPSThe certificate problem of the protocol is also troublesome, the current method is to manually add the certificate toChromethe. ChromeThe new version needs to use plug-insScreen Capture,lic
based on WebRTC of the MCU Open Source Projects Licode the environment to buildDue to the needs of the project, we need to build multi-person communication and investigate three common structures of multi-person communication:1. The previous blog has been based on Codelab for three people chatting, a multi-person system based on Mesh structure. Specifically, the fake has n+1 client, then for each client needs to establish peerconnection with other N o
file Sharing Remote monitoring cooperative customer relationship ... More Vmukti Information
Last updated: Video conferencing software Vmukti 1.1.3.1 Release posted 6 year ago
Distance Education Platform BigBlueButton
BigBlueButton is an online video conferencing system developed using ActionScript, or a distance education system, with features such as online PPT presentations, video communication and voice communication, as well as tex
Previous notes, finishingWEBRTC uses UDP transport by default, but it can also be transmitted over TCP.With TCP transport, servers such as Turnserver,licode,janus and servers are required.1. If you use Turnserver, you only need the client to keep the relaytcp type of candidate, the others are discarded.2. If you are using a server such as Licode,janus, TCP is not supported by default.Because they are used a
There are Licode and kurento in contact.Licode Flaw: Limited documentation support, Licode app client library only JSKurento Advantages: Complete Documentation, demo-ready, Packaging API is more complete. Its main features are:
Networked streaming protocols, including HTTP, RTP and WebRTC.
Group Communications (MCUs (Multipoint Conferencing Unit) and Sfus (Selective Forwarding unit.) functional
, familiar with the relevant components preferred, such as: Kafka,elk,flink,storm,flume, etc.;
5, monitoring system to build research and development experience is preferred.
Job Requirements
1, the actual development experience of audio and video engine, master audio and video streaming input, output method, the audio and video streaming between the equipment transfer efficiency and management have a deep understanding;
2, has a solid network technology foundation, the socket Communication, UDP
Which framework or library is the best for use WebRTCHttp://stackoverflow.com/questions/24857637/current-state-of-javascript-webrtc-librariesI want to know the which framework or library is the best for use WebRTC. Here are a small and incomplete list of libraries/sdk out there. Any lib that I forgot, feel free to let me know:Libraries:
Simplertc
Rtcmulticonnection
Crocodilertc
Lynckia/licode (This was more interesting for their s
large number of client video calls is to use the multipoint Control Unit, which is a server that is primarily used to publish streaming media data between clients.MCU can handle video conferencing in different resolutions, frame rate, encoding. Ability to process transcoding, do selective streaming, mix, audio and video data recording,For multiplayer video, there are a lot of problems to deal with, such as how to display multi-person video? How to deal with mixing?Cloud platforms like VLine are
right now, Asbundle,trickle ICE, new codecs the existing media servers would most likely not support and So on, not to Mentiondata Channelsandwebsocketsand the the-they could be used in a WebRTC environ ment to transport protocols like BFCP or MSRP, which SBCs or other legacy components would usually expect on TCP and/or UDP and negotiated theOld fashionedThe.Ok, we need a gateway ... what's now?Luckily for us all!, several people has worked on gateways since the first WebRTC browsers has been
Previous notes, finishingWEBRTC in the default open RTX for packet loss retransmission, the introduction of RTX can refer to Rfc4588,https://tools.ietf.org/html/rfc4588#section-4RTX uses an additional SSRC transmission, SSRC is identified in the SDP.↵a=rtpmap: rtx/90000↵a2736695910239189782Like this.A RTX packet, in Turnserver, is such that the raw UDP data->turn/stun protocol header->RTP Header1->RTP header2In RTP header1, according to payload type to distinguish RTP, RTX data, if it is rtx, yo
support multi-person interaction, but many people communicate at the same time, for the host side of the CPU pressure andThe network pressure is very big;Question 8: What encoding do you use for your video and audio separately?Answer 8: Universal coding scheme is: Video using H264, audio using AAC; If the end-to-end is controllable,It is recommended to use H265 for higher compression rate;Question 9: What is the recommended video conferencing system in the third scenario?Answer 9: If you are in
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