Android IOS WebRTC Audio Video Development Summary (10)

Continue with the unfinished part of the previous article, including the following three sections:

1, extension: WEBRTC multiparty calls.

2,mcu Multipoint Control Unit.

2, Extension: VOIP, telephone, message communication.

Note: Translation is not verbatim, but in accordance with their own understanding of the translation, at the same time, in order to facilitate understanding, but also to join some of their own organization language.

Reprint please indicate source: Http://www.cnblogs.com/lingyunhu.

English from: http://www.html5rocks.com/en/tutorials/webrtc/infrastructure/

1 to 1 Extension: WEBRTC multiparty calls

Perhaps you can first look at the IETF standard proposed by Justin Uberti. REST API for access to TURN Services.

1 to 1 users outside of a call can easily be imagined, for example: Video conferencing between universities, or public event handling, where one person says hundreds of people listen. In a network of networks, WEBRTC clients are able to use

Multiple rtcpeerconnections and other clients to establish a connection between the Talky.io use this method, the client is not a 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 other

In 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 is then pushed to each client to relieve client pressure, especially on the mobile side, Webrt.org provides an example. (Sample client application)

Starting with chrome 31 and OPERA18, streaming data from one rtcpeerconnection can be used as an input source for other endpoints, a demo link (simpl.info/multi), which enables a more flexible architecture

Because he enables the client to choose which remote endpoint to establish a call link with.

Multi-point control unit

A better solution to a 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 also trying to optimize network routing.

If possible, you can also create your own routing service by buying an MCU hardware package.

Cisco MCU back view.

Open source MCU software is also available, for example: Licode,mantis.

Extensions: VOIP, phone and messaging

WEBRTC's standardized properties make it possible to communicate with other communication platforms through a browser, such as a phone, video conferencing.

SIP is the signaling protocol for VoIP and videoconferencing, and if the SIP client is to establish communication with the WEBRTC client, the preferred one must have a server to convert the signaling,

Of course, if your WEBRTC client with the SIP protocol will not have to be transferred, the communication is established, is the transfer of streaming media, because both sides of the audio and video coding is not the same,

So need to have a service to do code stream conversion, such as WEBRTC with the VP8 video encoding, general video conferencing is H264.

PSTN (Public switched telephone network), he is a common analog telephone circuit-switched networks, so if the WEBRTC client wants to communicate with the telephone, first of all through the PSTN gateway.

Similarly, the WEBRTC client should interoperate with IM clients that follow the jingle protocol, which must have a server to convert the signaling, which jingle developed by Google and extended on an XMPP basis to support audio and video, and Google talk uses jingle. Jingle, but want to communicate with QQ is a bit difficult, QQ with their own definition of format, but they also do not want to also outside the product directly interoperability, as for the reasons you understand.

There are already many products that communicate with external products by making full use of WEBRTC, such as:

SipML5: Open source JavaScript SIP client.

Jssip:javascript SIP Library.

Phono: Open source JavaScript phone library.

Zingaya: An embedded mobile phone component

Twilio: Voice and message.

UberConference: Meeting.

SIPML5 developers have created the Webrtc2sip gateway, and WEBRTC communicates between the phone and the computer by using Openbts cell.

 More...

WebRTC Codelab: Step-by-step instructions tell you how to create a text and video chat app that uses the Socket.io signaling service that runs on node.

Google I/O WebRTC presentation with WebRTC Tech leads, Justin Uberti.

Chris Wilson ' s SFHTML5 presentation:introduction to WebRTC Apps.

This book (WebRTC books) gives details about data and signaling channels, as well as network topology diagrams.

WebRTC and Signaling:what The years has taught Us:tokbox's blog describes why it is not a good idea to define a signaling in the specification.

Ben Strong's presentation A Practical Guide to Building WebRTC Apps offers a number of WebRTC topologies and infrastructure architectures.

The WebRTC chapter in Ilya Grigorik's high performance Browser Networkinggoes deep to WebRTC architecture, use Cases and performance.

Above if errors and omissions, welcome to correct and supplement, greatly appreciated.

Android IOS WebRTC Audio Video Development Summary (10)