Soon I could use a SIP to GSM gateway, because I'll need to make phone calls from Germany to a mobile phone in Czech Republik. Hence, I thought about implementing one. Now, the questions are: Do...

For an WebRTC application, I want to signal, that there are no more remote candidates while using Trickle ICE. The W3C WebRTC spec says: This method can also be used to indicate the end of remot...

Is there a way to integrate SIP.js (both audio and video calling are needed) with React Native? Importing the library itself is easy enough, but the issues I'm running into are: WebRTC support: i...

A combination of local and remote tags along with Call-ID is used to identify a dialog. It is said that Call-ID is a unique value for a call. Why then is Call-ID not used solely to identify the dia...

I'm trying to migrate a voip service using freeswitch on GKE (google cloud managed kubernetes cluster) in order to make the service scalable. I have managed to migrate freeswitch to docker and get...

I've been experimenting with WebRTC with an Asterisk server (v13.18) on the same LAN as my computer. I configured the Asterisk extension 6003 to automatically answer and play a certain notorious so...

I have two SIP endpoint and wanted to send the media over DTLS-SRTP. I am little bit confuse in below points Is DTLS-SRTP and SRTP packets send over DTLS connections are same or different ? If d...

I want to develop a VoIP application for iPhone . But I don't know the basics of VoIP concepts and also if there are any sources or library available that can I use in my application . So i...

Consider this sample SIP dialog A-->--INVITE-->--B CSeq 101 A--<--TRYING--<--B CSeq 101 A--<--200 OK--<--B CSeq 101 A-->-- ACK -->--B CSeq 101 A-->-- INFO -->--...

Good evening Stack Overflow! I really need help for a project of mine where I'm using sip.js and a VoIP to make real calls to a phone number. The Goal I want to allow the user to record the audio...

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 ...

I see above log message when initiating SIP call from Android app using Android SIP SDK. Here's the line which causes it: mSipManager.makeAudioCall(mSipProfileLocal, mSipProfilePeer, listener, 20)...

The title is as clear as it gets. I'm searching for the jain-sip documentation but I'm unable to find it. There is a a link to documentation at the jsip.java.net website but I get send to the...

I have been looking for a way to set up the Android SIP stack to be able to establish a SIP call between two devices on the same network, in an ad-hoc manner. i.e without REGISTERing to a SIP serve...

I am using android SIP(android.net.sip) for VOIP calling. Trying to send DTMF. In SipAudioCall Class of android two method has been given void sendDtmf(int code, Message result) Sends a DTMF code....

This question was asked by somebody on Quora here but there was no answer. Is there any work around solution for this, other than changing the source code of jsSip? The issue is that most JS SIP l...

I have to develop a mobile app with audio and video functionality. I browsed the web, I found out I need a SIP server. An SIP server does it works like Apache for web apps? I also found this : p...

I am developing an Android application, in which I want to make a calls using internet using SIP in android. So I need to maintain my own SIP sever for my app users, how can I create my own SIP ser...

I have Followed the below path http://clearwater.readthedocs.org/en/stable/Manual_Install/ for installation of Clearwater IMS node-specific modules. But when i was trying to install Homer & Hom...

I'm little bit confused about sip proxy and sip outbound proxy? What is the basic difference between sip proxy and sip outbound proxy? Is it mandatory to use sip outbound proxy along with sip proxy...

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).

I have a requirement, i have a web site in which i want to implement outgoing call and incoming call functionality. I am using freeswitch on windows as sip server, currently i am able to initiate c...

I want to use G.729 audio codec on my android application. I have did a lot research on this and came to know that pjsip is most promising solution for this. But I have not much idea about all this...

By looking into another SIPp related question I learned that it is now possible to play WAV files using the rtp_stream action. I've tried several different WAV files with no success. All I get is ...

Is there any way to make call by just dialing a local IP address? Simply an IP to IP call. How can I do this? What changes should I make in pjsip code? I don't want to register in any server or VOI...