点对点对等SIP呼叫与Android SIP协议栈？ [英] Peer-to-peer SIP call with Android SIP Stack?

问题描述

我一直在寻找一种方式来建立Android的SIP协议栈，以便能够建立一个SIP呼叫两个设备之间在同一网络中，在一个特设的方式。即不注册到SIP服务器。

I have been looking for a way to set up the Android SIP stack to be able to establish a SIP call between two devices on the same network, in an ad-hoc manner. i.e without REGISTERing to a SIP server.

我一直没能得到这个工作，作为SIP演示包括服务器注册，我不能让它拨打或接听电话没有这一步。

I have not been able to get this to work, as the SIP Demo includes server registration, and I cannot get it to make or receive a call without this step.

我甚至不知道这应该是可能的。在几乎没有提到这一点，我已经能够找到的是相互矛盾的（有人说这是可以做到与特定的设置，他们不说什么，有的说Android的SIP API是不是意味着这个）。

I am not even sure if this is supposed to be possible. The little mention of this I have been able to find is conflicting (some say it can be done with a specific set up which they do not say what is, and some say the Android SIP API is not meant for this).

我想知道是否有人已经得到了这个工作，或有任何线索，我怎么可能去配置API对于这一点，因为我想用在寻找第三方的人之前，内置的SIP API。

I was wondering if anyone has got this to work or has any clues as to how I could go about configuring the API for this, as I would like to use the built in SIP API before looking at third party ones.

我开发的应用程序是一个内部的，这将始终在相同的设备上运行，这样一个事实，即SIP API是不是在所有设备上present不会为我是一个问题。

The application I am developing is an internal one which will always be running on the same devices, so the fact that the SIP API is not present on all devices will not be an issue for me.

推荐答案

我一直停留在相同的问题。

I have been stuck on the same problematic.

如果你可以把它没有Android的SIP API，你可以看的 RTP API ，让你有点低级别的工具，使一个P2P的VoIP应用没有服务器的需要。

If you can make it without the android sip api, you can look at the rtp api which gives you a bit lower-level tools to make a P2P VOIP application without the need of a server.

要支持音频会议和类似用途，则需要   实例化两个班作为端点流：

To support audio conferencing and similar usages, you need to instantiate two classes as endpoints for the stream:

AudioStream指定远程端点，由网络   映射和配置的音频codeC。 AudioGroup重新presents地方   端点对一个或多个AudioStreams。该AudioGroup混合所有   AudioStreams并任选与设备扬声器和交互   麦克风同时

AudioStream specifies a remote endpoint and consists of network mapping and a configured AudioCodec. AudioGroup represents the local endpoint for one or more AudioStreams. The AudioGroup mixes all the AudioStreams and optionally interacts with the device speaker and the microphone at the same time.

相对应的是，你必须写自己的设备发现协议，以了解所使用的audiostream同行的端口在此的答案

The counterpart is that you have to write your own device discovery protocol in order to know the port used by the audiostream peer as explained in this answer

这个问题并不难，如果你只打算做一个一对一的谈话，但有点麻烦，如果你想使一对N的谈话。

The problem is not so hard if you only intend to make one-to-one conversation but is a little bit trickier if you want to make one-to-n conversation.

有关一对n的谈话，会议主持人有权实例化ñaudiostream因为他想打电话给每个远程设备。每个远程对只有一个audiostream连接到主机audiostream之一。

For a one-to-n conversation, the conference host has to instanciate n audiostream for each remote device he wants to call. Each remote peer has only one audiostream linked to one of the host audiostream.

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