voip

Android - supress the native In Call Screen and replace with 3rd party?

馋奶兔 提交于 2019-12-06 12:07:50
问题 Currently creating a VOIP application for Android and I was wondering is there any possible way to supress the In Call Dialog when sending a call over GSM and use our own 3rd party In Call Dialog or screen instead? At the minute when we switch from our VOIP call and send it over GSM via the ACTION_CALL intent it does everything we want except it brings up the native In Call screen whereas we want our own In Call screen to display here. So I was wondering what kind of options we would have on

Detect incoming VoIP call Android

天涯浪子 提交于 2019-12-06 11:47:27
问题 I want to close my app, when user get Viber/Skype/WhatsApp call. I am tried to use AudioFocus listener and Notification parsing, both of them not really good. May be there is other method that can help me? 回答1: Checking for MODE_IN_COMMUNICATION in audio manager should achieve this. https://developer.android.com/reference/android/media/AudioManager.html#MODE_IN_COMMUNICATION public static boolean isVoip(Context context){ AudioManager manager = (AudioManager)context.getSystemService(Context

ios10 iphone5s voip siphon pjsip2.5.5 Error opening sound device

陌路散爱 提交于 2019-12-06 10:52:22
Someone could tell me why this happen? iOS10, iPhone5s, VOIP Siphon pjsip2.5.5. These issue was not appear every time,it came sometimes when you want to make call or answer call (add at:20161226). Error opening sound device: 14:06:03.137742 pjsua_aud.c ....Set sound device: capture=-1, playback=-2 14:06:03.137775 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@16000/1/20ms 14:06:03.137837 coreaudio_dev. ......Using VoiceProcessingIO audio unit 14:06:03.147404 acquire_call enter,call_id:2 14:06:03.233365 level:4,msg:14:06:03.233 pjsua_aud.c .....Opening sound device (speaker + mic)

Why asterisk not properly working with android sip client?

为君一笑 提交于 2019-12-06 07:18:27
问题 Asterisk= 1.8.11.0 Android= 2.3/4.0.3 Android Sip client=Native Android sip client/sipdemo When i call from my pc using zoiper/xlite to android (native android sip client) now i can hear audio from both sides but when i make call from android to pc (zoiper/xlite) i cannot hear anything on android. On the other hand i have tested this scenario on elastix (which also uses asterisk 1.8.11.0) with no problem in audio. pc(zoiper) ip 192.168.15.27 android ip 192.168.15.71 asterisk server ip 192.168

how to implement voip in android

孤街醉人 提交于 2019-12-06 05:41:19
问题 I want to use voip functionality in android. What can I use to implement voip functionality in android?. It is required that same application is install on receive side mobile. Please post links or source so that I can implement this functionality. 回答1: I am using this link that describe how to build a VoIP client for Android. I think you will also find it useful. 回答2: Have you tried sipDroid . Its open source client for Voip calls. Follow the link i shared you will get all the details you

Android VOIP SipException: Failed to create SipSession

久未见 提交于 2019-12-06 03:47:17
问题 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

How to build PJSUA2 (swig) with Visual Studio 2015

社会主义新天地 提交于 2019-12-05 22:05:18
I try to compile PJSUA2 for Python. I could manage to compile the source with Visual Studio 2015 Community edition. My question is, how can I install the Python module now? The guide suggests to use make and make install. I tried to install also minwg, but it is not working as the project was compiled with VS. I have also tried to compile with minwg, but I never could succeed due to undefined requirements. Also official PJSUA guide recommend to use VS for Windows. Following settings have been applied to build from source: Swig location has been added to path JAVA_home system variable has been

speex splitted audio data - WebAudio - VOIP

只谈情不闲聊 提交于 2019-12-05 20:03:22
Im running a little app that encodes and decodes an audio array with the speex codec in javascript: https://github.com/dbieber/audiorecorder with a small array filled with a sin waveform for(var i=0;i<16384;i++) data.push(Math.sin(i/10)); this works. But I want to build a VOIP application and have more than one array. So if I split my array up in 2 parts encode>decode>merge, it doesn't sound the same as before. Take a look at this: fiddle: http://jsfiddle.net/exh63zqL/ Both buttons should give the same audio output. How can i get the same output in both ways ? Is their a special mode in speex

Failed to push call stats, status code: 403 | ERROR TCMetricsPublisher | Twilio

陌路散爱 提交于 2019-12-05 19:58:27
I am integrating the voip , thus integrated twilio . I also setup the server code on heroku successfully. I create the token successfully, and getting the callback on handler as well. When I run the application on device, then callback in delegate comes Device: <TCPresenceEvent 0x17de3840 name=jenny, available=YES> didReceivePresenceUpdate After some time, below error occurred: [ERROR TCMetricsPublisher] Failed to push call stats, status code: 403 Delegates callback: I got the callback in connectionDidConnect After some time, I got the callback in connectionDidDisconnect automatically. My code

Parsing SIP message body

落爺英雄遲暮 提交于 2019-12-05 18:42:31
I'm implementing SIP protocol and I'm stuck while parsing SIP message. I'm using the oSIP library. My code is like that: #include <stdio.h> #include <stdlib.h> #include <osip2/osip.h> #include <osipparser2/osip_parser.h> #include <string.h> void main() { int i,error; osip_message_t *message; char text[]="INVITE sip:jarsku@feanor.pc.lut.fi SIP/2.0\nCall-ID: 123456789@aradan\nVia: SIP/2.0/UDP 157.24.25.137:5060\nFrom: Arto <sip:athamala@feanor.pc.lut.fi>\nTo: Jari <sip:jarsku@feanor.pc.lut.fi>\nCSeq: 1 INVITE\nContent-Type: application/sdp\n\nv=0\na=3333aaa333"; char *p=(char *)&text; i = strlen