webrtc multicast

In the next is WEBRTC development novice, at present encountered a problem, turned over to have not understood. Maybe English is not good, look at the document to see blindfolded, so has not found a solution.Development environment:node. JS Server builtI'm using Socket.io to do communications now.Development Purpose:A classmate to B students to initiate a request, B received after the two sides live video.If there is a clear classmate trouble tell me

The development of video conferencing based on the third party WEBRTC open source platform is not very difficult, mainly the business aspects. However, once involved in the core of the underlying issues need to read the source code, to find out the bug, the difficulty is not small.The project needs to analyze the creation process of peerconnection.assuming clienta,clientb is divided into offer and answer. Offer end PC =new rtcpeerconnec

write on the frontA: The purpose of writing a blog1. Self-study of the hard self-evident.2. All kinds of information on the Internet is a mixed bag, many are outdated.3. Based on the latest WEBRTC source to share some experience in their work.4. If you write a good people clap, write bad don't spray. Money to hold a field, no money ...Two: Compile compile or compile1. It is best to prepare a VPN, do not think of someone to copy the code to upload to t

Reprint Please specify source: http://www.cnblogs.com/fangkm/p/4401075.htmlThe first two blog posts complete the WEBRTC audio and video collection module, and the next step is to introduce the key audio and video coding modules. However, before introducing the audio and video coding module, we need to introduce the channel concept, and the transmission flow of each WEBRTC data is encapsulated into a channel

, the receiver side decoding good performance, no mosaic phenomenon.3.2, adding the QoS module will bring a certain delay and lag, because packet retransmission is time-required.3.3, the above plan is WEBRTC inside the nack concrete realization way.The above scheme is provided by Peng Zuyuan, a senior audio and video expert from the ring, with some adjustments, and Kelly for editing and finishing.Peng has many years of audio and video codec developmen

This article mainly introduces to help a programmer solve WEBRTC doubt process, the article from the blog Garden Rtc.blacker, support original, reprint please explain the source (www.rtc.help)This article mainly comes from the mail, why I will be specially organized into essays, mainly based on the following reasons:1, the author email me The purpose is to ask questions, but he asked questions in a way worthy of praise, asked very specific (if asked t

1, about WEBRTCWebRTC is a very popular project. The first problem encountered is the WEBRTC compilation problem.Fortunately, a company has helped compile and put it in Maven's repo.Address:Http://mvnrepository.com/artifact/io.pristine/libjingleThe update is very fast, and WEBRTC the official Basic sync update.2,android DemoThe project is also within the pristine project:Https://github.com/pristineio/apprtc

The combination of gstreamer and webrtc is a little breakthrough, gstreamerwebrtc Today, I found a fork killer in gstreamer, and quickly came up with a general framework and solution plan, using the gst-inspector to perform object introspection attribute detection first, then, the gst-launcher tool is used for Pipeline Test. Finally, the channel Logic Source Code is implemented using c to implement webrtc-

from a downhill racing race. Most of the video remains the same, except that the moving parts, i.e. the car and the audience, need to be encoded as P-frames without changing the video. The I frame is generated as a new reference point for P frames. Usually create an I-frame when the image changes very much, such as: panning, scene switching, a large number of actions, sudden disappearance and other scenes. error recovery mechanism:it is suitable for the error recovery mechanism of various packe

The previous article (WEBRTC Audio-related Neteq (a)) is an overview of Neteq, know that it is mainly used to solve the network delay jitter drops and other problems to improve the voice quality, but also know that it has two large units of MCU and DSP components. MCU is mainly received from the network of voice RTP packets into the packet buffer, but also based on the calculated network delay and jitter buffer delay and the feedback from the DSP unit

WebRTC (Web Real time communication) is not Google's original technology, in 2010, Google bought about $68.2 million for VoIP softwareDeveloper Global IP Solutions Company, open source WEBRTC real-time communication project.Voice engine is the gips of voice communication, it is mainly through a series of transmission control to achieve low bandwidth transmission of real-time voice, Gips speech engine hasa w

General Statement In the previous article, we explained how to integrate the OPENH264 codec into the WEBRTC, but OPENH264 can only encode baseline H264 video, and in terms of encoding quality, X264 is the best, This article will explain how to integrate the X264 encoder into the WEBRTC, in order to achieve decoding, at the same time to use the ffmpeg. The overall process, as before, is divided into the re-

. Publish-subscribe model (pub/Sub) [multicast-based communication mode]The publishing-subscription model uses a content hierarchy called a topic to replace the unique destination in the PTP model and send applications to publish their own messages, it indicates that the message describes a topic in the layered structure. Applications that want to receive these messages subscribe to this topic. Subscribe to a topic in the subtopic hierarchy. subscribe

As the sender of multicast data, adding arm_linux to multicast groups is not required. You can add or not. Arm_linux data sending program: (only functional functions are provided) Void multicast_thread (){Rtpsession sess;Int portbase = 6000;Int status;// Int length;Int loop;Bool iscast;Unsigned long destip;Unsigned long localip;Char * buff = "multicasting test ";Int length = strlen (buff );Destip = inet_add

The role of network layer and Internet multicast 1. Briefly describe the role of the OSI physical layer, data link layer, and network layer? Network Layer: provides the most suitable path for packets or groups through the Communication Subnet through the routing selection protocol (Routing Algorithm). Data Link Layer: various control protocols are used to convert erroneous physical channels into error-free data frames that can be reliably transmitted.

Windows NLB (Network Load Balancing) is a load balancing technology that Windows $ Server begins to provide on the operating system. NLB uses a distribution algorithm to distribute load balancing across multiple hosts, increasing the scalability and availability of key IP-based services such as the WEB, virtual private networks, streaming media, Terminal Services, proxies, and so on, while detecting host failures and automatically reallocating traffic to other operations masters. Thus providing

Inside the Ethernet there will be broadcast and multicast. Both of these packages are an STA contracted to multiple Sta.Broadcast and multicast between LAN ports are reliably forwarded when there is no wifi, but it is not the same if WiFi is present.Power considerationsAccording to the Protocol, broadcast and multicast are sent to STA at Dtim. The Dtim in the AP

A python script that can send multicast data on LINUX. In fact, it is not too difficult to compile. The specific code is as follows: first declare that this is not what I wrote, but what I found, but what others wrote is quite good. The code is as follows: # UDP multicast examples, Hugo Vincent, 2005-05-14.Import socketDef send (data, port = 50000, addr = '192.1.100 '):"Send (data [, port [, addr])-multi

memory leak:Calling the DeleteObject in the API[System.Runtime.InteropServices.DllImport ("gdi32.dll")] bool DeleteObject (INTPTR hobject );PrivateBitmapSource Copyscreen () {var handle =IntPtr.Zero; BitmapSource Source =Null;Try{var bitmap =New Bitmap ((intintvar g = Graphics.fromimage (bitmap); G.copyfromscreen ( Span style= "color: #800080;" >0, 0, 0, 0, bitmap. Size); Handle = bitmap. Gethbitmap (); SOURCE = Imaging.createbitmapsourcefromhbitmap (handle, IntPtr.Zero, Int32rect.empty,

Real-time video communication via WebRTC (I.) Real-time video communication via WEBRTC (II.) Real-time video communication via WEBRTC (iii) In the previous article, we explained WebRTC 's overview, history, security, and developer tools. The next step is to explain the process of building