I've loaded the Android SIP Demo into Android Studio. I registered an account on getonsip.com, added it to my phone accounts, and verified it works by calling the phone from the getonsip.com webapp...

In my application I want to have one Activity that enables user to add his SIP account parameters in fields. I don't want them to go Settings->Call->Internet Call Settings->Add Accounts->Add I hav...

I am looking for a good SIP library either written in C# or that provides a C# wrapper. Does not necessarily need to be free. Has anyone used anything good? To clarify, I am talking about th...

i'm trying to build a simple VoIP application using c# so i found that the Ozeki SDK is the simple way to do that but when i'm trying to registration SIP account using the SIPAccount class from the...

I'm trying to compile this lib: https://github.com/BelledonneCommunications/linphone-iphone During generation I get this: /linphone-iphone/submodules/build/..//externals/speex/libspeex/cb_search...

I've been doing some research on Phonegap and Cordova, it seems there is no support for creating apps that have SIP and VOIP functionality since HTML5 and Javascript have little or no support for t...

Can anyone describe each parameter in following media information m=audio 12548 RTP/AVP 0 8 101 Thanks

I have a Twilio account which has a number (let's say 8881231234), and I have Asterisk box. I'd like to use Twilio as an Asterisk trunk to be able to make calls at their rates and receive cal...

What are the available API or existing technologies on android that can help one make call using the internet, more like VOIP. I checked out documentation online including this . Kinda confused whe...

From documentation for setKeepAliveTimeout:handler: method of UIApplication: In iOS 8 and later, voice-over-IP (VoIP) apps register for UIRemoteNotificationTypeVoIP push notifications instead of...

I am new in SIP call using RTP, now I am trying to send and receive voice streams using RTP for sip call. I am done with connecting two emulators and able to send INVITE and INVITE-ACK using jain...

I am trying to implemented the VoIP application using the AudioGroup and AudioStream classes of the android.net.rtp package. But my application not function properly. After "Join" the "AudioGroup" ...

I am writing a custom Android application that allows the user to make VOIP calls using SIP. When the user presses a button, a voice call is initiated with another SIP user. That's it. The only oth...

I've been coding a SIP client using the Android SDK's native SIP libs. For some reason I cannot get my account to register with the server. Here are the testing grounds: Linux Mint 17 XFCE runni...

I wanna make SIP peer to peer VoIP call in android. This call should be made in local network without any need to access to internet. In fact I don't want to use any external server to register the...

I have a quick sample app I am trying to build, to demonstrate simple SIP calls to and from a Nexus 7. The device is currently running Android 4.2.1. LogCat shows the error: SipBroadcastRequest...

I'm developping a SIP application, and when i want to call someone(with its identifier configured in the server) i have a NullPointerException => "Error when trying to close manager." Here is the ...

I have been learning more about WebRTC, SIP and PSTN and how they work together especially the ability to receive phone calls in browser. I have spent some time on Twilio's website and I like...

Recently I am working on an application in which I have to implement application to application voice call and instant messaging like Viber and Line. I did some R&D and got two methods t...

So, I've been coding this Android Sip application using Androids Sip library for some time, but i can't get the registration to work. Currently it gives me the following error when i call SipManage...

I'm trying to diagnose a problem with the OpenSIPS ( a SIP proxy ) application. When sending two different UDP packets to the same IP and port, one call fails with -1 EPERM (Operation not permitte...

Is there a replacement technology for TAPI that supports third-party call control (3pcc)? I want to provide the following 3pcc functionalities in an application: Outgoing call: User clicks at...

I'm trying to implement a sip server for connecting to from an HTML sip client(made using sipml5). During my research into doing this I've come across sip over web-sockets which might be useful to ...

I'm working with the Jain Sip library, trying to create a connection to an asterisk server. I've been working with the textclient example app from this page: http://www.oracle.com/technetwork/artic...

I am developing a SIP client. I understand SIP requests and SIP responses but, in SIP messages, how are the call id and branch tags generated? RFC3261 does not specify this.