Asterisk
Asterisk1 is an open source telephony applications platform distributed under the GPLv2. In short, it is a server application for making, receiving, and performing custom processing of phone calls.
The project was started by Mark Spencer in 1999. Mark had a company called Linux Support Services and he needed a phone system to help operate his business. He did not have a lot of money to spend on buying one, so he just made his own. As the popularity of Asterisk grew, Linux Support Services shifted focus to Asterisk and changed its name to Digium, Inc.
The name Asterisk comes from the Unix wildcard character, *. The goal for the Asterisk project is to do everything telephony. Through pursuing this goal, Asterisk now supports a long list of technologies for making and receiving phone calls. This includes many VoIP (Voice over IP) protocols, as well as both analog and digital connectivity to the traditional telephone network, or the PSTN (Public Switched Telephone Network). This ability to get many different types of phone calls into and out of the system is one of Asterisk‘s main strengths.
Once phone calls are made to and from an Asterisk system, there are many additional features that can be used to customize the processing of the phone call. Some features are larger pre-built common applications, such as voicemail. There are other smaller features that can be combined together to create custom voice applications, such as playing back a sound file, reading digits, or speech recognition.
1.1. Critical Architectural Concepts
This section discusses some architectural concepts that are critical to all parts of Asterisk. These ideas are at the foundation of the Asterisk architecture.
1.1.1. Channels
A channel in Asterisk represents a connection between the Asterisk system and some telephony endpoint (Figure 1.1). The most common example is when a phone makes a call into an Asterisk system. This connection is represented by a single channel. In the Asterisk code, a channel exists as an instance of the ast_channel data structure. This call scenario could be a caller interacting with voicemail, for example.
Figure 1.1: A Single Call Leg, Represented by a Single Channel
1.1.2. Channel Bridging
Perhaps a more familiar call scenario would be a connection between two phones, where a person using phone A has called a person on phone B. In this call scenario, there are two telephony endpoints connected to the Asterisk system, so two channels exist for this call (Figure 1.2).
Figure 1.2: Two Call Legs Represented by Two Channels
When Asterisk channels are connected like this, it is referred to as a channel bridge. Channel bridging is the act of connecting channels together for the purpose of passing media between them. The media stream is most commonly an audio stream. However, there may also be a video or a text stream in the call. Even in the case where there is more than one media stream (such as both audio and video), it is still handled by a single channel for each end of the call in Asterisk. In Figure 1.2, where there are two channels for phones A and B, the bridge is responsible for passing the media coming from phone A to phone B, and similarly, for passing the media coming from phone B to phone A. All media streams are negotiated through Asterisk. Anything that Asterisk does not understand and have full control over is not allowed. This means that Asterisk can do recording, audio manipulation, and translation between different technologies.
When two channels are bridged together, there are two methods that may be used to accomplish this: generic bridging and native bridging. A generic bridge is one that works regardless of what channel technologies are in use. It passes all audio and signalling through the Asterisk abstract channel interfaces. While this is the most flexible bridging method, it is also the least efficient due to the levels of abstraction necessary to accomplish the task. Figure 1.2 illustrates a generic bridge.
A native bridge is a technology specific method of connecting channels together. If two channels are connected to Asterisk using the same media transport technology, there may be a way to connect them that is more efficient than going through the abstraction layers in Asterisk that exist for connecting different technologies together. For example, if specialized hardware is being used for connecting to the telephone network, it may be possible to bridge the channels on the hardware so that the media does not have to flow up through the application at all. In the case of some VoIP protocols, it is possible to have endpoints send their media streams to each other directly, such that only the call signalling information continues to flow through the server.
The decision between generic bridging and native bridging is done by comparing the two channels when it is time to bridge them. If both channels indicate that they support the same native bridging method, then that will be used. Otherwise, the generic bridging method will be used. To determine whether or not two channels support the same native bridging method, a simple C function pointer comparison is used. It‘s certainly not the most elegant method, but we have not yet hit any cases where this was not sufficient for our needs. Providing a native bridge function for a channel is discussed in more detail in Section 1.2. Figure 1.3 illustrates an example of a native bridge.
Figure 1.3: Example of a Native Bridge
总结：Asterisk使用了两种传输方式，一个是通过服务器中转的方式进行客户端交互；另外一个就是类似当初FBT的架构，如果客户端之间可以通过p2p连接，就使用p2p进行Media传输。