When should I allocate DirectSound buffers?

Please help me make up my mind. Allocate at the start of the application, free at the exit. Allocate when streaming starts, free as soon as streaming stops. Which one? Why? In case it matters:...

Reading *.wav files in Python

I need to analyze sound written in a .wav file. For that I need to transform this file into set of numbers (arrays, for example). I think I need to use the wave package. However, I do not know how...

What is returned by wave.readframes?

I assign a value to a variable x in the following way: import wave w = wave.open('/usr/share/sounds/ekiga/voicemail.wav', 'r') x = w.readframes(1) When I type x I get: '\x1e\x00' So x got a...

What is H.450.2? What happens when it doesn't exist?

What is H.450.2? I noticed that many applications such as Ekiga for linux/windows allowes this. But many others does not, as a result i see some problems which confusing me and having no answer to...

Open source video encoders for an embedded system

I recently designed an H.323/SIP compliant video server (in code at least) fully equipped with a sockets based API which a .NET SDK would use, and a web server, you know ... all of that stuff....

SIP: Wait for ACK packet on Callee site to start RTP session

The Situation: I have a question concerning the ACK message (yellow) which is send from the Asterisk to the Callee (Tel B) after the Callee has send its 200 OK + SDP message (purple). The Asterisk...

Lib OPAL compilation error

I would like to get some help about compiling OPAL lib in order to build Ekiga from source. I've installed Ptlib, all right. But when I try installing OPAL, first I do ./configure...

openmcu+gnugk+ekiga

I've got openmcu server installed and wanna to use gnugk with it, but actually can't find any manuals of how can I configure it. Could you, please, point to any or tell how can gnugk can be...

Makefile : add a shared library compilation

I am having some troubles trying to compile a shared library in Ekiga. In fact, I would like to make a .so library to get information about packets. But I have to compile this library in the...

How to work with freeswitch

I should setup a VOIP network. I installed Virtualbox and install ubuntu server on a virtual machine and installed freeswitch on it. I also installed Ekiga softphone on my ubuntu desktop. Now , I...

Capturing incoming audio from Ekiga softphone/Pulseaudio JACK sink

I want to do speech recognition using Sphinx. I'm looking to capture the output/incoming audio of the Ekiga VOIP softphone using Java or Python and pass it on to Sphinx. Right now, the output is...

speex codec - packets per second

how many packets per second if we use Speex Codec - 16kHz - H.323,SIP in Ekiga Softphone? and how to calculate it?

how to change the default welcome message in asterisk

I have installed asterisk 1.8.9.0. and i have installed ekiga soft phone too. When tried to call using ekiga, the caller can hear the default message , "Congrats, you have successfully installed...

debian apt packages hash sum mismatch

From the Debian command line, I'm getting a hash sum mismatch after executing aptitude update; aptitude upgrade. Below is the command line output. I've tried an aptitude clean, but this does not...

Asterisk instantly terminates WebRTC (JSSIP) call

I'm running Asterisk 11.2.2 with SRTP and STUN support under Calculate Linux (Gentoo-based distribution). When I try to call from one WebRTC instance to another, using JSSIP, the call passes, but...

WebRtc2SIP: No video is been received/transmitted when made call between chrome and a SIP client

I am a newbie to webrtc2sip. I have setup my webrtc2sip gateway and registered to sip2sip.info as my domain. The problem is when I make video calls from chrome to any SIP client(ekiga/jitsi) the...

Get SPS and PPS from h264 encoded video in JAVA

I'm a bit stuck on a question actually and i reaaly hope that someone can help me with this issue. My problem is as follows : I have a live usb camera with which i'm encoding only the video in...

dvswitch, v4l2loopback and gst-launch to v4l2sink give too high framerate

I'm testing two webcams with dvswitch: dvswitch -h localhost -p 2000 using avconv to generate dv streams, piping them to dvsource-file: avconv -y -f video4linux2 -s 640x480 -r 15 -i /dev/video0...

Network Steganography StegoSip Tool

I am trying to use StegoSip tool coupled with Ekiga softphone. Finally Ekiga works, but when I run StegoSip, it gives me the warning that cb() takes exactly 3 arguments (2 given). I found the...

SIP Session Establishment Through Two Proxies with Kamailio

I want to implement a session establishment SIP call through two SIP proxies. For that I am using Kamailio Server, but I do not know how to change kamailio.cfg (/etc/kamailio/kamailio.cfg) config...

Pjsip multiple calls using few outgoing lines fails

I'm using pjsip high-level api PJSUA for doing multiple calls at one time. My sip operator support up to 3 outgoing lines for calls, so in theory it should be possible to call 3 persons at one...

Register for remote VOIP server with PJSIP in IOS

I want to make app that let user to make calls over ip (VOIP). I was following this tutorial http://www.xianwenchen.com/blog/2014/06/09/how-to-make-an-ios-voip-app-with-pjsip-part-1/ but now I...

Asterisk reached but can't register

I'm new at asterisk and following asterisk...

Asterisk gives "Strict RTP learning" message and no audio for Chrome WebRTC but works in Firefox

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...

How to make a H.323 trunk in Asterisk 15

I'm trying to make a H.323 trunk in asterisk 15 (in a remote server with Ubuntu server 16 installed) with ooh323 addon, to test if works I've the softphone ekiga on my local machine. But when I...