voip | 易学教程

What's your Interactive Voice Response Platform? [closed]

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问题 Closed. This question is off-topic. It is not currently accepting answers. Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed 3 years ago . For those of you working in the voice space, what are you using as your IVR platform? I am using Microsoft Speech Server 2007. What are some equivalent packages? Is anyone using open source software for handling inbound or outbound calls? Note that I'm not just talking about speech recognition, which is one

Are there parallels to Asterisk AMI and AGI in FreeSWITCH?

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问题 Asterisk has Asterisk Manager Interface (AMI) and Asterisk Gateway Interface (AGI), using which one can trigger PHP scripts at certain events from Asterisk. Using the same PHP scripts can also instruct Asterisk what to do next to a call/conference. Are there parallels to AMI and AGI in FreeSWITCH? 回答1: These are good questions. I just wanted to add a few things to what @dkwiebe said (which is correct, BTW). The AMI equivalent in FreeSWITCH is the event socket. (Technically it's not the "Event

iOS - best practice to send incoming call notifications on VoIP app

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问题 The first solution I can think of for the incoming call notification is Apple's Push Notification service. However, it is not guaranteed.. there's a relatively high chance it may get lost.. and in a VoIP app, the incoming call notification is so important that I can't afford to miss it too often... Thus, I followed the tips and enabled the Background Mode to keep the app alive and listening to any incoming call invite. By right, I should just show local notification when the app gets the

How does WebRTC handle many-to-many connections?

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问题 if I am in a room with other 7 users, I am wondering if WebRTC force every user to establish a connection to each one of other participants. Obviously it would consume something like 7kb/s*7 download and even upload, and many connection cannot handle this if their connection is already busy. Instead with some kind of media relay the bandwidth usage would be only 7kb/s but you would lose bandwidth adaptation between peers. Do you know any media relay, or way to solve this problem? is TURN

Even with “voip” present in “UIBackgroundModes” in “plist”, iOS App does not auto start after device reboot in iOS10

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问题 I need my VoIP App to auto start after rebooting the device. Apple docs clearly mention that :- (=========EDIT: This is from official Apple docs please have a look at this before commenting or answering that the App cannot be launched without user interaction or silent push notification. Also have a look at Github project below, people have verified this behaviour) Values for the UIBackgroundModes array Value : voip Description : The app provides Voice-over-IP services. Apps with this key are

VOIP on Android libraries or sample code

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问题 Thanks in advance. I am working on a VOIP application on Android. Are there any open-source code, libraries or sample applications, for VOIP software developers to quickly create VOIP apps? Thanks in advance. actually i don't know the process for using voip in android app.. so how to use voip in app. 回答1: I think you probably want the SIP package. 来源： https://stackoverflow.com/questions/10734519/voip-on-android-libraries-or-sample-code

关于这几天使用IOS的ASYNCSOCKET完成无限后台的过程

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【推荐】2019 Java 开发者跳槽指南.pdf(吐血整理) >>> 　　这几天用了下ASyncSocket完成前后台即时通讯，当时有想过用消息推送的技术实现的，可是后来想到消息推送的不可靠性还是算了。于是使用了tcp/ip实现后台主动发送数据给前台的功能。最开始设计后台的时候，我有考虑到数据量比较大的问题，所以数据大的时候我会使用分包和组包的功能去实现。TCP/IP在传输数据的时候，一般不会大于1500字节，所以我每512字节分了一个包。然后当一次性数据包接收太多的时候，就出现了粘包的问题。因为我在数据传输的时候使用的是json，每一个分包都是由{}括起来的，所以我就想着在包头上加上一段基本不会重复的分割字符串，然后服务器接收到分包的时候每次都根据这个字符串分割一下，第一次分割的时候第一行绝对是空字符串例如：@Hinagiku{“Name”=“桂雏菊”}，我分割出来结果是： “”，“桂雏菊”，所以说第一行我就可以直接跳过，每次取分包的时候从第二行开始取。然后后台根据包的ID号，序号进行组包。如果当前分包在5分钟内没有接收完毕，就代表当前分包接收失败了，要求客户端或服务器重新发送。粘包问题解决完毕之后，我开始实现心跳包功能，当时想的是，每隔1分钟发一次心跳包，服务器放一个线程。每隔几秒钟判断一次，当前的所有TCP连接的最后一次访问时间是多少号

iOS实用技巧

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【推荐】2019 Java 开发者跳槽指南.pdf(吐血整理) >>> 　　在开发企业app的时候，有的时候基于安全性的考虑，不允许设备连接外网。这就出现了一个问题，就是iOS的推送功能没法工作了，因为iOS的推送功能是固化在系统里，必须连接苹果的APNS服务器才能工作的，为了能让这类只能工作在内网里的app也能拥有推送功能，就需要我们自己来实现推送功能了。　　自主推送的实现方法是利用iOS的voip类app可以驻留在后台的功能。这类app，系统在设备开机时即被启动，app可以将自己的一个socket委托给系统，在socket有数据到达时，系统就会唤醒app，给它一段很短的cpu时间来处理数据，再加上UIApplication的keepAliveTimeout handler（最小10分钟间隔一次），可以每隔一段时间就重新建立一次连接，来达到保持socket长连接的需求。注：由于voip类app可以使app常驻在后台并维持socket长连接，因此苹果对这类应用有及其严格的审查，所有不是真正的voip的app都会被拒绝！切记！ 1.设置应用为voip应用打开<app>-info.plist文件，加入如下的key Required background modes，追加App provides Voice over IP services 2.将socket设置成异步模式

私有推送实现方法

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SIP Trunk / SIP 中继服务

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【推荐】2019 Java 开发者跳槽指南.pdf(吐血整理) >>> CTS SIP Trunk/SIP中继可以让你现有的IPPBX连接到CTS的平台从而使用VOIP外线落地。享受拨打全国统一资费0.09元，多路中继号码，可接听电话。如果你已经有了IPPBX系统，使用CTS SIP Trunk企业级VOIP中继/SIP中继话务服务，可以让你在外呼电话时节省更多的钱，让您的VOIP系统对内对外都实现真正的 Over IP。使用CTS SIP Trunk/SIP中继无需硬件语音板卡，无需电话线路，只需一条网线，架设从内部分机到外呼电话的完全 VOIP 系统。不限企业地域，全国均可使用。如需办理请电话咨询：4008290998. 备注：使用CTS SIP Trunk服务，几乎任何的现有IPPBX均可以直接配置使用，无需任何硬件设备，仅需支持SIP协议即可。例如：FreePBX，Asterisk，FreeSwitch，Trixbox，PBX IN A Flash，Elastix，3CX，AVAYA，Lync等等。来源： oschina 链接： https://my.oschina.net/u/1452491/blog/200550

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