webrtc streaming

Transferred from: Http://www.oschina.net/p/kurentoKurento is a WebRTC streaming media server and some client APIs, which makes it easier to develop advanced video applications for WWW and smart phone platforms. The types of applications that can be developed using Kurento include video conferencing, audio and video broadcasting, audio and video recording, transcoding, and more.kurento/kurento-media-serverwa

Federico II, where Lorenzo is currently also a Ph.D student. He has been involved in real-time multimedia applications over the Internet for years, especially from a standardisation p Oint of view. Within the IETF, in particular, he especially worked in XCON on centralized conferencing and Mediactrl on the interactions Between application Servers and Media Servers. He is currently working on webrtc-related applications, in particular on conferencing

direct-to-peer connection provides a low latency that enables game operation and video streaming. Faster, more real-time interactions, such as sensor feedback. A secure peer-to- peer connection allows you to implement a private exchange of information without the intermediate server records and management. This reduces the need for large service providers. At the same time, it offers opportunities for people to create new services and applications. I

This paper mainly introduces the RTP/RTCP protocol in WEBRTC, Weizhenwei, the earliest published articles in the Wind network, ID:BEFOIOSupport original, reprint must indicate the source, welcome attention to my public number blacker (Id:blackerteam or WEBRTCORGCN).a prefaceThe RTP/RTCP protocol is the cornerstone of streaming media communications. The RTP protocol defines the packet format for

ability to record and store media streams for any purpose, which is hard to do on the mesh architecture.Integration with other communication technologiesAnother advantage of using media servers is the ability to communicate with other systems outside the web system, such as the PSTN via SIP trunking or a service streamed over rtmp (like Fackbook live or YouTube live streaming).You can see an example of a previous blog in which the Kurento media serve

jitter and video packet loss), Image enhancements ( Image quality enhancement). The transport contains SRTP (secure real-time transport protocol for audio and video streaming), multiplexing (multiplexing), P2p,stun+turn+ice (for NAT network and firewall traversal). In addition, secure transport may also use DTLS (datagram safe transport) for encrypted transport and key negotiation. The entire WEBRTC commun

of code. In the following article, we will elaborate on the main differences between WEBRTC native development and hybrid development.WebRTC native DevelopmentWEBRTC code is developed in C + +, and if native development is used, there must be someone on the team who is proficient in C + +. And if you want to be able to understand and modify the WEBRTC code, just C + + is far from enough, but also to b

from end to end. If it fails, it is forwarded through the Mediation Server Relay service. Streaming is the work of rtcpeerconnection.RtcpeerconnectionRtcpeerconnection is part of the WebRTC, which is a stable and efficient handle to the end-to-end transfer of data.Below is a WebRTC architecture that shows the role of Rtcpeerconnection, as you can see, the green

This article mainly introduces WEBRTC (we translate and collation, translator: Weizhenwei, check: Blacker), the earliest published in the "Weaving wind net"Support original, reprint must indicate the source, welcome attention to my public number blacker (Id:blackerteam or WEBRTCORGCN).Technically speaking, using a webcam for online broadcasting does not require WEBRTC. The camera itself is a server that can

ability to create and manage sessions. This layer protocol is left to the application developer to customize the implementation. (5)VoiceengineThe audio engine is a framework that includes a range of audio multimedia processing, ranging from video capture cards to network-based transmission solutions. Ps:voiceengine is one of WEBRTC's most valuable technologies and is open source for Google's acquisition of Gips company. On VoIP, the technology industry is leading, and later articles will learn

and ultra-wideband audio codec for VoIP and streaming audio, ISAC with a sampling frequency of up to khz or + khz and a variable bit rate of 12-52 Kbps. The ilbc--is a narrowband speech codec for VoIP and streaming audio, with a sampling frequency of 8 khz, a 20 millisecond frame bitrate of 15.2 kbps,30 mm frames with a bit rate of 13.33 Kbps, and the standard defined by IETF RFC 3951 and 3952. Neteq

Original link: Introduction to WebRTC on Android Original Author: Dag-inge Aas Translated by: appear.in Translator: Dorisminmin Status: Complete WebRTC is regarded as a New of web long-term open source development, and is the most important innovation in web development in recent years. WEBRTC allows web developers to add video chats or point

lot of time this way the effect is very good,This only takes up a lot of bandwidth and CPU, especially for mobile phones.Complete mesh topology: Everyone is connected to each otherIn addition, the WEBRTC client can select a client to send streaming data directly to other clients, in this star network structure, you can directly do a publishing service side, the client will stream to the server,The server i

the device's camera and microphone* Rtcpeerconnection:rtcpeerconnection is a component that WEBRTC uses to build stable, efficient streaming between point-to-point* Rtcdatachannel:rtcdatachannel enables a high-throughput, low-latency channel between browsers (Point-to-point) for transmitting arbitrary data Here's a general introduction to these three APIs MediaStream (Getusermedia)The MediaStream API provi

"Note" This series of articles, as well as the use of the installation package/test data can be in the "big gift –spark Getting Started Combat series" get1 Spark Streaming Introduction1.1 OverviewSpark Streaming is an extension of the Spark core API that enables the processing of high-throughput, fault-tolerant real-time streaming data. Support for obtaining data

this This paper mainly introduces the realization of WEBRTC in Nack, Weizhenwei, the article was first published in the Wind network , Id:befoioSupport original, reprint must indicate the source, welcome attention to my public number blacker (Id:blackerteam or WEBRTCORGCN).In WEBRTC, forward error correction (FEC) and packet loss retransmission (NACK) are important methods to resist network errors. FEC adds

first, the network topology structureWEBRTC can also be used as multiparty calls, such as video conferencing, in addition to peer-to-peer communication. When it comes to multi-party calls, we need to select a schema for our application. This is a very important decision, because how to organize users is related to the scale of the conference system. Corresponding to WEBRTC, there are two common network topologies: Mesh networks and star-shaped netwo

WEBRTC reply content: I am in development and have a basic understanding of the WebRTC source code stack. It mainly consists of two key technologies: 1. webRTC Video/Voice Engine, including camera microphone operations, Video preprocessing, VP8 coding/decoding, and streaming media transmission (RTP/RTCP); 2. implement

the MediaStream API can get video, audio synchronization flow through the device's camera and microphone* Rtcpeerconnection:rtcpeerconnection is a component that WEBRTC uses to build stable, efficient streaming between point-to-point* Rtcdatachannel:rtcdatachannel enables a high-throughput, low-latency channel between browsers (Point-to-point) for transmitting arbitrary dataHere's a general introduction to

stackReal Time ProtocolB. Stun/iceCall connections between different types of networks can be established through stun and ice components.c. Session ManagementAn abstract session layer that provides session building and management capabilities. This layer protocol is left to the application developer to customize the implementation. (5) VoiceengineThe audio engine is a framework that includes a range of audio multimedia processing, ranging from video capture cards to network-based transmission