sip

sip会话流程以及sip介绍

狂风中的少年 提交于 2019-12-05 14:52:51
参考连接 :https://www.2cto.com/kf/201609/546336.html https://www.w3cschool.cn/session_initiation_protocol/session_initiation_protocol_introduction.html https://www.ibm.com/developerworks/cn/opensource/os-cn-sip-intro/ 专业词汇: SDP(会话描述协议)    RTP(实时传输协议) SIP (会话发起协议) SIP 网络元素 (1)用户代理 它是端点和SIP网络的最重要的网络元件之一。端点可以启动,修改或终止会话。 用户代理是SIP网络中最智能的设备或网络元件。它可以是软电话,移动电话或笔记本电脑。用户代理在逻辑上分为两个部分 。 用户代理客户端(UAC) - 发送请求并接收响应的实体。 用户代理服务器(UAS) - 接收请求并发送响应的实体。 SIP基于客户端 - 服务器架构,其中呼叫者的电话充当发起呼叫的客户端,并且被叫者的电话充当响应呼叫的服务器。 (2)代理服务器 它是从用户代理接收请求并将其转发给另一个用户的网络元素。 基本上代理服务器的作用就像一个路由器。 它具有一些智能来理解SIP请求并且在URI的帮助下向前发送它。 代理服务器位于两个用户代理之间。

Android-ready JAIN-SIP library?

烂漫一生 提交于 2019-12-05 12:22:37
Greets! I am developing (trying to develop) a VoIP SIP application for Android, and after two weeks of bickering with mjsip, pjsip and the sdk's libraries, I have settled on JAIN-SIP. The libraries look great, in theory. No need to learn any android native code, lots of documentation (yet not enough, since I'm here), etc. My first attempt was using the Android SDK's SIP libs (yes, I know they're based on JSIP), and it failed after the SipManager.open() refused to open the profile for communication with the server, with no exceptions or any sort of output to help me diagnose the issue. (If you

Why does the native SIP stack included in Android 2.3 does not work over 3g?

三世轮回 提交于 2019-12-05 11:04:50
I was wondering why does the native SIP stack included in the Android framework(since 2.3) does not work over 3g? Could it have something to do with any laws or restrictions google may have with his partners? And furthermore, does anybody know if there is any plans to remove that restriction ? Thx In GingerBread , SipManager is set to work only on wifi. `<bool name="config_sip_wifi_only">true</bool>` But from 4.0 onwards , this config has been changed to false So Ideally native sip stack should work on 3G from Ice Cream Sandwich onwards. But again if the phone is sold by a service provider,

No ringing event on incoming calls

拈花ヽ惹草 提交于 2019-12-05 09:09:04
I don't know where is a problem. Listener doesn't catch onRinging event (so I can decide will I accept or reject incoming calls). in manifest is this: <uses-permission android:name="android.permission.USE_SIP" /> in main activity onCreate is this: IntentFilter filter = new IntentFilter(); filter.addAction("android.SipDemo.INCOMING_CALL"); callReceiver = new IncomingCallReceiver(); this.registerReceiver(callReceiver, filter); ... SipManager manager = SipManager.newInstance(this); Intent i = new Intent(); i.setAction("android.SipDemo.INCOMING_CALL"); PendingIntent pi = PendingIntent.getBroadcast

PJSIP Custom Registration Header

天涯浪子 提交于 2019-12-05 08:21:13
问题 I am attempting to setup SIP communication with an internal server (using the PJSIP library), however, this server requires a custom header field with a specified header value for the REGISTRATION call. For example's sake we'll call this required header MyHeader . From what I have found, the pjsua_acc_add() function will add an account and register it to the server using a config struct. The parameter reg_hdr_list of the config struct has the description: The optional custom SIP headers to be

SIP, asterisk, adhearson and VoIP

扶醉桌前 提交于 2019-12-05 08:07:21
问题 I'm trying to create a VoIP based IVR service that interacts with a web application. From what I understand, adhearson runs on top of asterisk. What else do I need to have on the server to satisfy the equation? I think I need a way for asterisk to connect to a voip account. I'd appreciate any help and/or phrases to google. 回答1: If you want to build your IVR quickly and easily you'd be better off looking at something like tropo.com (tropo's parent, voxeo, own adhearsion) or twilio.com, they've

Ubuntu 14.04下搭建Python3.4 + PyQt5.4.0 + Eric6.0开发平台

怎甘沉沦 提交于 2019-12-05 07:06:50
简要说明:QT是一个跨平台界面库和开发工具集,PyQt是其Python封装模块,其封装采用Sip完成(可以让C++库直接在Python中调用),Eric是基于Python的一个代码编辑器。 引言 找 了很多Python GUI工具集,还是觉得PyQt比较理想,功能强大跨平台,还支持界面设计器。花一天时间折腾了Ubuntu14.04(32位)+ Python3.4 + Qt5.4.0 + PyQt5.4 + Eric6.0 的完整开发平台的搭建,各种出错差点放弃了,好在终于一一解决了,记录下来以后搭建起来方便点。 安装Python3.4 Ubuntu14.04貌似默认安装了Python2.7 和Python3.4。故Python3.4的安装可略去。 安装Qt5 Ubuntu14.04自带Qt4的部分库文件,但是我们的平台要求Qt5,所以首先安装Qt5. Qt5.4.0下载: http://qt-project.org/downloads 安装过程: 下载到的如果是Qt5的在线安装程序,为其赋予可执行权限即可单击启动。由于我之前文件都安装在/opt目录下,故这里以root身份安装: sudo chmod a+x qt-opensource-linux-x86-1.6.0-5-online.run sudo ./qt-opensource-linux-x86-1.6.0-5-online

In SIP, why the Contact header field MUST be present in the Invite request

生来就可爱ヽ(ⅴ<●) 提交于 2019-12-05 05:11:31
Usually, the Contact header field in the Invite request is useless. For example, the UAC and the UAS are in different LANs. The Contact field may be: INVITE sip:bob@sipprovider SIP/2.0 Contact: Alice<alice@192.168.1.10> ..... There is no use of the Contact field while we can still build a dialog. Then, why the Contact header field is mandatory? The contact field contains the address at which the callee can reach the caller for future requests. For example, it's necessary so that the callee can send a BYE or a re-INVITE to the caller. The Contact header says where you are (or rather, where your

Connecting an OBi (OBi202) to Twilio SIP?

允我心安 提交于 2019-12-05 04:41:31
I am looking to connect Twilio SIP to an OBi202 endpoint. It seems like it would be easy considering the Twilio SIP support and the OBi is a SIP endpoint... I found the settings for the RTP port range, server, and credential information, but beyond that I am stuck and cannot connect. Can anyone provide some guidance for connecting an OBi to Twilio? Edit: I did manage to figure out the inbound routing, although as Devin said, it requires a third party. I setup a free account with iptel.org. They provide registration and allowed me to connect my OBi to their pure SIP service. Using the verbs I

SIP support for Cordova/Phonegap

我怕爱的太早我们不能终老 提交于 2019-12-05 03:23:59
Is there any possibility to use SIP in Cordova/Phonegap? I've not found any plugin, yet. Support for WebRTC is also not available (but planned in 2.x, according to their roadmap). Supporting SIP or WebRTC on Android has nothing to do with PhoneGap / Cordova , even though they could be added as some extension but naturally they come with the browser provided to PhoneGap by the OS. Right now you need to look for external libraries to do that for you. You may add an extension to your PhoneGap (written in Java) yourself to communicated to some SIP server. Or if you would like you can use a