sip

I implement simple SIP client app for receiving calls. I go through official manual and get code from it. I noticed that SipManager.newInstance(getApplicationContext()); returns null. Documentation says that it happens when SIP API is not supported by device. However I use LG G6 with Android 7.0 and I successfully test third-party SIP clients from Google Play. So I doubt that API is not supported really. How could I check that? My manifest has all permissions ( INTERNET and USE_SIP ) Permission for USE_SIP is granted by user The Problem is that Android SDK is not supported over all the devices

Im trying to run a VOIP call using built in SIP on android 3.1. I have physical tablet device (galaxy Tab 10.1). For testing purpose, I have created a project from SipDemo example - it works fine! (meaning my credentials are working and my device/network is fine). my Manifest.xml <?xml version="1.0" encoding="utf-8"?> <manifest xmlns:android="http://schemas.android.com/apk/res/android" package="modera.com.doorcontroller" android:versionCode="1" android:versionName="1.0"> <application android:icon="@drawable/logoeditedsmall" android:label="@string/app_name" android:debuggable="true"> <activity

我这篇文章是有所指和所限的，他就是限制在由Mobicents和Apache两位老大所出的mss-x.x.x-apache-tomcat-x.x.x，x.x.x当然是指的版本号了，在这里就不一一指定了。我所配置的是1.3.2的mss是基于tomcat6.0.20，其他的我也没有功夫去一一配置和实验，抱歉！ 一、Eclipse配置 1、使用Eclipse JEE版本 因为SIP Servlet开发要使用到和网页相关的部分，因此要使用Eclipse JEE版本(从同事那儿了解到的，因为本人原来从不做Web相关的东西，这次也只是为了将工程最终打成war包，才用到这个JEE版本) 2、下载与SIP Servlet开发相关的插件 插件的下载地址如下： http://mobicents.googlecode.com/svn/downloads/sip-servlets-eclipse-update-site/ Eclipse下怎么下载安装插件不用我废话了吧，大牛们，肯定都比我懂的多：） 3、创建基于SIP Servlet的工程 如下图所示，中创建工程时，选择Dynamic Web Project 然后在“下一步”中如下图所示，分别在"Target runtime”中选择"Apache Tomcat V6.0"，在"Dynamic web module version"中选择"2.4"，在

Is it possible to use an Android phone as a simple GSM gateway? The phone would receive a VoiP call using (preferably) Android built-in SIP stack, initiate a GSM call, and bridge audio both ways. After one call is terminated, the other one ends, too. How could I approach the problem? My earlier attempts failed at bridging audio between connections. Is there a SDK supported way of doing this, that I missed? Or do I need to implement some sort of a workaround? There are two problems with what you are asking: How to get at the incoming audio stream of the cellular call. How to get at the outgoing

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. 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 done a lot of the grunt work for you and setting up Asterisk is not for the faint hearted. If you want

To call phone number from notebook through Twilio I created ASP.NET-MVC 5.2 application. I can call a number and answer the phone but I don't know how to achieve live voice(to be able to talk) connection instead of just playing music. I created an action method inside HomeController : public ActionResult Call(string to) { client = new TwilioRestClient(Settings.AccountSid, Settings.AuthToken); var result = client.InitiateOutboundCall(Settings.TwilioNumber, to, "http://twimlets.com/message?Message%5B0%5D=http://demo.kevinwhinnery.com/audio/zelda.mp3"); //it causes to play zelda theme when call

I get this error when I try to establish a new call from pjsip: pjsua_aud.c ..Error retrieving default audio device parameters: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) [status=420006] Exception: Object: {Account <sip:192.168.0.2:54496>}, operation=make_call(), error=Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) I installed libasound2-dev : sudo apt-get install libasound2-dev , and recompiled pjsip, but still, it gives me the same result .. Am I missing something here? N.B. I use Ubuntu 11.10 and the sound works fine with no problems, so please don't tell me I

We have got into couple of interop issues where, The video mode that is required by couple of endpoints in market are little different and only understands H.264 packetization modes (FUA type) (i.e) FU -A NAL unit type.(while others do not play the video on receiving a fu-a nal type payload) Does anyone know what is this FUA type of packetization mode? How is it different from packetization modes 0,1,2 as defined in RFC3984? Is the video encoder/decoder supports it, how can it be appropriately signalled in SIP SDP session wherein the attributes do not get changed even when traversing through