blocks|key|5041481|text|lipphone提供Voip呼叫尝试this+link|type|unstyled|depth|inlineStyleRanges|entityRanges|offset|length|data|5041482|entityMap|0|LINK|mutability|MUTABLE|url|http://www.linphone.org/^0|I|9|0|0^^$0|@$1|2|3|4|5|6|7|L|8|@]|9|@$A|M|B|N|1|O]]|C|$]]|$1|D|3|-4|5|6|7|P|8|@]|9|@]|C|$]]]|E|$F|$5|G|H|I|C|$J|K]]]]

lipphone provide Voip calls try <a href="http://www.linphone.org/" rel="nofollow">this link</a>

blocks|key|1753875|text|从SIPLevel9开始，原生APi支持被添加到促进VoIP的框架中。http://developer.android.com/reference/android/net/sip/package-summary.html|type|unstyled|depth|inlineStyleRanges|entityRanges|offset|length|data|1753876|这是一个演示应用程序：http://developer.android.com/resources/samples/SipDemo/index.html|1753877|entityMap|0|LINK|mutability|MUTABLE|url|http://developer.android.com/reference/android/net/sip/package-summary.html|1|http://developer.android.com/resources/samples/SipDemo/index.html^0|Z|23|0|0|B|1T|1|0^^$0|@$1|2|3|4|5|6|7|P|8|@]|9|@$A|Q|B|R|1|S]]|C|$]]|$1|D|3|E|5|6|7|T|8|@]|9|@$A|U|B|V|1|W]]|C|$]]|$1|F|3|-4|5|6|7|X|8|@]|9|@]|C|$]]]|G|$H|$5|I|J|K|C|$L|M]]|N|$5|I|J|K|C|$L|O]]]]

Starting APi Level 9, native SIP support was added to the framework which facilitates VoIP.
<a href="http://developer.android.com/reference/android/net/sip/package-summary.html">http://developer.android.com/reference/android/net/sip/package-summary.html</a>

Here is a demo app:
<a href="http://developer.android.com/resources/samples/SipDemo/index.html">http://developer.android.com/resources/samples/SipDemo/index.html</a>

blocks|key|732758|text|编辑:我推荐使用Apache+web服务器http://www.openvbx.org/的这个解决方案。|type|unstyled|depth|inlineStyleRanges|entityRanges|offset|length|data|732759|对于原生sip项目，您可能需要检查http://code.google.com/p/csipsimple/以及this，这可能是一个很好的起点。|732760|为了进一步阅读，我推荐这篇文章，它可能对如何在更大的规模上进行start+with+VoIP和它的成本有用。|732761|对于简单的解决方案|732762|entityMap|0|LINK|mutability|MUTABLE|url|http://www.openvbx.org/|1|http://code.google.com/p/csipsimple/|2|http://www.denphone.com/14-google-code-projects-for-android-iphone-sip-and-asterisk|3|http://www.zdnet.com/blog/ou/voip-and-open-source-the-next-great-frontier/25^0|L|N|0|0|H|10|1|1J|4|2|0|V|F|3|0|0^^$0|@$1|2|3|4|5|6|7|X|8|@]|9|@$A|Y|B|Z|1|10]]|C|$]]|$1|D|3|E|5|6|7|11|8|@]|9|@$A|12|B|13|1|14]|$A|15|B|16|1|17]]|C|$]]|$1|F|3|G|5|6|7|18|8|@]|9|@$A|19|B|1A|1|1B]]|C|$]]|$1|H|3|I|5|6|7|1C|8|@]|9|@]|C|$]]|$1|J|3|-4|5|6|7|1D|8|@]|9|@]|C|$]]]|K|$L|$5|M|N|O|C|$P|Q]]|R|$5|M|N|O|C|$P|S]]|T|$5|M|N|O|C|$P|U]]|V|$5|M|N|O|C|$P|W]]]]

Edit: I would recommend this solution with Apache web server <a href="http://www.openvbx.org/" rel="nofollow">http://www.openvbx.org/</a> .

For native sip project you may wanna check <a href="http://code.google.com/p/csipsimple/" rel="nofollow">http://code.google.com/p/csipsimple/</a> as well as <a href="http://www.denphone.com/14-google-code-projects-for-android-iphone-sip-and-asterisk" rel="nofollow">this</a> may be good starting point.

For further reading I would suggest this article, which may be useful for how to <a href="http://www.zdnet.com/blog/ou/voip-and-open-source-the-next-great-frontier/25" rel="nofollow">start with VoIP</a> on bigger scale and the costs of it.

For simple solution

blocks|key|1748428|text|后端有两个选项：|type|unstyled|depth|inlineStyleRanges|entityRanges|data|1748429|a)家庭服务器-更难的方法，取决于你的技术规格。您需要从服务器获得哪些功能？例如，Asterisk有用于电话会议的音频混音器，而它缺乏用户在线状态(您需要与Openfire集成)。Kamailio/Openser通过SIMPLE支持在线状态，但它们缺乏音频混音器支持(您需要与Asterisk或SEMS集成)。您也可以考虑像Sipwise这样的"all-in“设备，但如果您想要配置除默认配置之外的其他内容，则需要广泛的知识。|offset|length|style|BOLD|1748430|好的部分是，大多数服务器只需要很少的工作就可以让sip音频呼叫运行，所以如果你只需要这样做，最好是设置自己的服务器。|1748431|这种方法的优点--你可以完全控制服务，你可以查看后端日志，你可以测试用例，这在公共服务中是做不到的。缺点-它需要大量的资源来设置、配置和支持。这是主观的，取决于您需要从服务器获得的功能。|1748432|b)公共服务-这是最简单的方式。根据服务能力的不同，它可能是免费的，也可能是月费的。大多数公共SIP服务允许sip音频呼叫，因为它依赖于服务的任何其他内容(到PSTN的呼叫、视频呼叫、会议音频/视频呼叫等)。我会推荐sip2sip.info，但你也可以很容易地找到其他的。优点是你可以立即开始使用它(在你注册之后)，并且你不关心服务的管理。缺点-你不能控制服务，你不能看到后端日志(如果你正在开发功能丰富的SIP软件客户端，这是至关重要的)。|1748433|entityMap|0|LINK|mutability|MUTABLE|url|http://www.asterisk.org/|1|http://www.igniterealtime.org/projects/openfire/|2|http://www.kamailio.org/w/|3|http://en.wikipedia.org/wiki/SIMPLE|4|http://www.iptel.org/sems/|5|http://www.sipwise.com/|6|http://wiki.sip2sip.info/wiki/SipDeviceConfiguration^0|0|0|7|15|8|0|26|8|1|2I|G|2|30|6|3|43|4|4|4I|7|5|0|0|0|0|6|30|C|6|0^^$0|@$1|2|3|4|5|6|7|17|8|@]|9|@]|A|$]]|$1|B|3|C|5|6|7|18|8|@$D|19|E|1A|F|G]]|9|@$D|1B|E|1C|1|1D]|$D|1E|E|1F|1|1G]|$D|1H|E|1I|1|1J]|$D|1K|E|1L|1|1M]|$D|1N|E|1O|1|1P]|$D|1Q|E|1R|1|1S]]|A|$]]|$1|H|3|I|5|6|7|1T|8|@]|9|@]|A|$]]|$1|J|3|K|5|6|7|1U|8|@]|9|@]|A|$]]|$1|L|3|M|5|6|7|1V|8|@$D|1W|E|1X|F|G]]|9|@$D|1Y|E|1Z|1|20]]|A|$]]|$1|N|3|-4|5|6|7|21|8|@]|9|@]|A|$]]]|O|$P|$5|Q|R|S|A|$T|U]]|V|$5|Q|R|S|A|$T|W]]|X|$5|Q|R|S|A|$T|Y]]|Z|$5|Q|R|S|A|$T|10]]|11|$5|Q|R|S|A|$T|12]]|13|$5|Q|R|S|A|$T|14]]|15|$5|Q|R|S|A|$T|16]]]]

You have 2 options for the backend:

a) home server - the harder way, depends on your technical specification. What functionality do you need from the server? For example <a href="http://www.asterisk.org/" rel="nofollow">Asterisk</a> has audio-mixer for conference call, while it lacks user-presence (you need to integrate with <a href="http://www.igniterealtime.org/projects/openfire/" rel="nofollow">Openfire</a>).
<a href="http://www.kamailio.org/w/" rel="nofollow">Kamailio/Openser</a> support presence through <a href="http://en.wikipedia.org/wiki/SIMPLE" rel="nofollow">SIMPLE</a>, but they lack audio-mixer support (you need to integrate with Asterisk or <a href="http://www.iptel.org/sems/" rel="nofollow">SEMS</a>). 
You can also consider "all-in" appliance like <a href="http://www.sipwise.com/" rel="nofollow">Sipwise</a>, but it requires extensive knowledge if you want to configure something aside from the default configuration. 

The good part is, most of the servers require very little effort to get the sip audio calls running, so if you only need that, better is to setup your own server.

The pros of this approach - you have total control over the service, you can view backend logs, you can test use-cases, which you can't with public service. The cons - it requires significant resource to setup, configure and support. This is subjective and depends on the functionality you need from the server.

b) public service - it's the easiest way. Depending on the service capabilities it maybe free or have a monthly fee. Most public SIP services allow sip audio calls, for everything else it depends on the service (calls to PSTN, video calls, conference audio/video calls etc). I would recommend <a href="http://wiki.sip2sip.info/wiki/SipDeviceConfiguration" rel="nofollow">sip2sip.info</a>, but you can easily find others as well. The pros are you can start using it immediately (after you register) and you don't care about the administration of the service. The cons - you don't have control over the service, you cannot see backend logs (which are vital if you're developing feature-rich SIP software client).

Are there any available api's in android for voice chat? What do we need to do exactly in the backend? Every where i could see information about the front end? No idea how it works in the backend? Do I need a SIP server? How to configure it? What will be its price?

Get started with VoIP

翻译质量差，导致语言生硬或混乱。

没有提供实际的解决方法或示例。

解答不清晰，无法理解或解决问题。

页面排版不美观，阅读体验差。

文章

问答

视频

学习中心

腾讯云实验室

直播

竞赛

腾讯云代码分析专区

腾讯iOA零信任安全管理系统专区

腾讯云架构师技术同盟交流圈

腾讯云数据库专区

腾讯云顾问专区

腾讯云原生专区

腾讯混元专区

腾讯云TCE专区

腾讯云Lighthouse专区

腾讯云HAI专区

腾讯云Edgeone专区

腾讯云存储专区

腾讯云智能专区

腾讯轻联专区 

腾讯云开发专区

TAPD专区

腾讯轻量云游戏服专区

腾讯云最具价值专家

腾讯云架构师技术同盟

腾讯云创作之星

腾讯云开发者先锋

腾讯云代码助手

云原生构建

TAPD 敏捷项目管理

Cloud Studio

SDK中心

API中心

命令行工具

涵盖代码开发、场景应用、自动测试全流程，助你从零构建专属AI助手

一站式MCP教程库，解锁AI应用新玩法

android有没有支持实时语音的api？我们到底需要在后端做些什么？我可以在任何地方看到关于前端的信息？不知道它在后端是如何工作的？我是否需要SIP服务器？如何配置？它的价格是多少？

问VoIP入门
EN

Stack Overflow用户

社区

活动

圈层

关于

腾讯云开发者

热门产品

热门推荐

更多推荐

问VoIP入门EN