voip

Service for sending SMS and making Voice Calls from Web site and Desktop application

We plan to integrate sending of SMS and making calls to our desktop and web applications. Both are written in Java. As for only sending SMS we know about great gateway from Clickatell. But ideally, we would like to use one service similar to it, but which supports Voice Calls and SMS. What service/gateway could you recommend? Here are...

Which iPhone API should I be using for streaming audio?

I'm looking to build a little toy app that is very similar to a voip application. One person would hold one iphone and talk to the other iphone. I don't want to use gamekit because it forces a p2p connection and does not work over 3g. I'm worrying about the server side of this later but just wanted to get started with the iPhone side of ...

Most important feature in VoIP honeypot today

I'm writing a VoIP honeypot. Right now, it's listening on a specific port (SIP) for incoming connections. What would you suggest are the most important features it should have in terms of scanning/attack detection and analyzing? I don't think there are many sophisticated attacks out there (yet), so implementing anything beyond DoS/floodi...

Problem initiating SIP session/ getClientTransaction(request) throws NullPointerException

Hi, I have a small issue, I had my SIP client working, and I changed the structure of the code. I kept the creation process of the SIP objects as it was before, but now it does not work. I keep getting: java.lang.NullPointerException at gov.nist.javax.sip.SipProviderImpl.getNewClientTransaction(SipProviderImpl.java:285) and ther...

average voip compression rates?

I am running some tests on audio compression and trying out Skype's Silk. In their test application I am seeing compression rates of 94%. This seems high, is this a typical rate on Silk? Is this comparable to other audio compression codecs? ...

How would a SIP request would look like when inviting more clients to conversation

Hi, I've finished my SIP client and it works, as long as it comes to one on one, or x on x if the participants are defined in the beginning of the call. I would like to attach a caller or a callee in the middle of the call, I understand the main issue is in the media session joining implementation, but I do need to tell the new partici...

PHP-Based intranet application Call center

Hi guys, As in title, I am making a PHP-Based intranet application Call center, I've finished the DB design/UI. But now I need to automatically intercept calls and send the caller ID to the PHP webserver so the request is routed to the next free agent while his webpage refreshes with the caller's card(and previous questions maybe). Now,...

What is the most standards compliant way to send a fax from an application?

From what I understand fax over VoIP is very unreliable due to the real-time requirements of the fax protocol. Simply using VoIP to send fax message to POTS-connected fax machines is hence not possible. Besides from proprietary fax API:s (such as sending fax messages via a company specific SOAP API), what standards based protocols are t...

SIP Callee does not get notification that call ended

Hey, I have deleted my previous question and post this updated: I have a an issue with my SIP UAC, once I received a ringing from the B2BUA on both the caller and callee, and the caller hangs up the call while the call is ringing (I send cancel request and receive "request terminated" on the caller side), the callee does not get any no...

Compiling pjsip for iOS 4.0

G'day guys, I've been having issues with compiling pjsip for iOS 4.0. I'm using the latest trunk version from SVN and keep getting a portaudio error. When using the piedmontwireless guide: http://www.piemontewireless.net/PJSip155_and_iPhoneSDK312 I get a missing separator error in my build.mak file, which would indicate a whitespace/ta...

Programming Asterisk PBX using PHP?

Ok I installed asterisk, now I would like to know how do I program asterisk PBX using PHP? Does Asterisk have an API that I can work with? Please provide basic examples how I could perform the following scenario send phone number to asterisk asterisk dials phone number ...

Use SIP in iPhone app...

Hi, I want to build an iPhone app which has the option to make calls over SIP (VoIP) but at this moment I have no idea how to start. Does anyone have some information about this topic, or maybe a demo project which I can use to implement the SIP functionality? Thanks in advance! ...

how do i deploy mobicents to glassfish ?

Possible Duplicate: how do i deploy mobicents to glassfish v3 Mobicents with Glassfish in place of JBoss? how do i deploy mobicents to glassfish v3? wating your answers.. ...

Off the shelf or roll my own?

I want to take an Anroid based tablet - not a phone, I need a large screen and I don't need 3G. The guy with the tablet will attach a web cam to it and a s/w application in the Adnroid tablet will stream the cameras feed to a web page (there may later be a need to stream video back to the Android tablet - tbd). Additionally, I need 2 w...

Blackberry VOIP client...

I want to start to implement the VOIP client for blackberry, but seeing the Blackberry api I feel that its not possible to build the VOIP client for Blackberyy. So I think there must be a different approach in implementing the client...... So any one have idea about it.... thanks in advance.... ...

H.225 User Information Packet Parsing

I'm writing some code using PacketDotNet and SharpPCap to parse H.225 packets for a VOIP phone system. I've been using Wireshark to look at the structure, but I'm stuck. I've been using This as a reference. Most of the H.225 packets I see are user information type with an empty message body and the actual information apparently shows up...

RTP packet combining

I have a bunch of RTP packets that I'd like to re-assemble into an audio stream. For each packet, I have the sequence number, SSRC, timestamp, and a byte array representing the data itself. Currently I'm taking each subset of packets by their SSRC, then ordering them by timestamp and combining the byte arrays in that order. Afterwards,...

How do I duplicate certificate authentication (Mumble (c/c++)) in Python?

Alright so, before I really get into this post, I am going to have to warn you that this might not be an easy fix. Whoever reads and is able to reply to this post must know a lot of c/c++, and at least some python to be able to answer the question I have above. Basically, I have a connection method from Mumble (a VOIP client), that con...

VoIP application in windows CE

Hi all, I want to develop VoIP application in Windows CE based device and included .NET compact framework 3.5 with C#,but I am newbie where do I start? Any API,samples? Thanks for your answers.. ...

SIP, asterisk, adhearson and voip

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. ...