I found this article:
www.codeproject.com/Articles/1077937/Possible-ways-to-organize-interaction-between-co,
and I know that there exist a code for the flash player.
Can I use only code for managing connections (as in the articles examples) and free flash player code and therefore get rid from integration software?
You need to be more specific but in reality the idea of the integration software is the following:
Session management
Multi-codec/format support
Interface Resource
Scheduling
Normally IP PBX supports SIP only, hence you need the transcoding between the SIP world (Audio + Video) and the Webcast world (Web browser/client/camera). Integration software as the one defined do a pretty good job and some of them are open Source (Wooza), If you want to replace them, I would do it with an MCU which support RMTP/Flash. Take a look at McuWeb project. Otherwise you need to write SIP Client code as well to integrate it with the SIP world
Related
I want to setup a personal videoconferencing service for my family, friends and myself. The main problem I have with current options is that they are either closed-source and centralized (GG hangouts, skype) or open-source but not working in corporate environment or in hotels (due to strict firewalling rules and the "Skype is going through, if you want VOIP use that" kind of netadmin reaction).
I have two solutions then. Either setup a STUN/TURN relay server and use XMPP and SIP as I used to, but that would require my friends to setup that too. Or setup a whole VOIP server. 2 solutions come to mind: SIP and XMPP. Though to my knowledge, each of them ultimately uses the (S)RTP/RTCP protocol.
And that's the problem. Out of the specific signaling part used by the two of them, I really can't figure out the difference between them, their typical use case.
I think you're right in that as far as setting up a video conferencing system XMPP and SIP are equivalent. They both are signalling only protocols and the media sessions they set up typically use RTP (although they can both be used to set up any kind of session you want but RTP is the norm).
The biggest problem is also going to be the one you mention about getting video streams out of a corporate firewall. Skype overcomes this obstacle by sending it's media over an SSL connection and is thus able to get through firewalls. Theoretically you could do the same with RTP and in the past I once used openvpn connections with a SIP client to test some audio calls. My experience wasn't great as the audio was very choppy, assumedly as a result of all the extra packaging that is required to get the high volume of small audio packets from one end to the other. That was nearly a decade ago though so perhaps with the better CPU and bandwidth resources available now it would work better.
Personally I think I'd stick with Skype as it's going to be a big hassle to set up your own system. If you were to go ahead with your own the first option I would try would be Asterisk combined with openvpn so that if the clients were behind a firewall or had NAT issues they could connect over it.
I have a client company with a simple web application (Python Flask) and I need to add a phone notification functionality to it.
The main requirement is that the app should call users, play a certain sound file and accept some tone input ("Hello! This is an automated message from your WebApp account. You have a meeting with $John today at $5pm. Please press 1 to confirm").
The other requirement is that the solution should be relatively cheap and fast to market.
I have done some research already and it seems that there are a few consequent steps to achieve that:
Set up an Asterisk or a FreeSwitch server;
Set up a SIP account;
Write some business logic for the Asterisk server which allows to make calls and play sounds via a SIP account;
Write an API at the Asterisk server and expose it to the Python Flask web app.
Do I miss something here? Can any of the steps be omitted anyhow? Can I do it simpler?
the fastest way to get it working is to use one of the cloud voice services with speech synthesiser. Here's a short list to check out:
Twilio
Tropo
Plivo
Here I listed some details.
Those services charge you per minute, plus you may have to pay some monthly fee.
If you want to run an independent and standalone service, I would recommend FreeSWITCH instead of Asterisk. It's got reach integration possibilities and API. You will need to read the FreeSWITCH book in order to understand how it works and how to build your service.
I agree with Stanislav Sinyagin on the cloud based solutions, but I would add one more, Voxeo Prophecy. Tropo is from Voxeo, but they have offered Prophecy as a solution for a lot longer and it supports the open standards CCXML and VoiceXML. The advantage of CCXML for outbound notification applications is you have a lot more control of the notification process.
The Prophecy platform has excellent call progress analysis (CPA) which will allow you to determine whether a machine or a human answered and handle the call accordingly. For example, it does not make sense to ask a machine to "...press one to confirm". Instead you may want to leave a message that provides a call back number for the user to confirm with after they have listened to the voice message. The CPA can be used to leave a message on a machine at the correct time (when the greeting message has stopped) so that you do not get clipped messages in the voice mail. CPA will also allow you to provide detailed reports on who was notified and for those that did not it can tell you whether it was a bad number (received a SIT tone), a modem or fax answered, or ring-no-answer (pretty rare these days). These type of details can factor into your retry process for failed notifications.
The other advantage to using Prophecy and open standards is your application will be portable to other IVR systems that are VoiceXML/CCXML compatible if you ever want to migrate. Tropo, Twilio, and Plivo all use proprietary API's which does not allow you to move your applications to other services. Prophecy is also available as a software solution so that if you want to take it out of the cloud you can run it on premise. You can get a two port version for free to try it out.
There is excellent documentation on developing outbound notification systems on Voxeo's developer site. Take a look at the CCXML documentation in section F on Outbound Dialing.
Not sure which development languages you are familiar with, but if you are used to ASP.NET MVC there is an open source project called VoiceModel that makes it easier to develop VoiceXML applications. The other advantage of VoiceModel is that you develop your application once and it will run on any VoiceXML compatible platform and Tropo. They are currently working on adding outbound notification support in this project that will work for both Tropo and VoiceXML.
Third party solutions listed are your easy choice. Running your own asterisk is also suitable for what you want to do, but i think for only this much it would be overkill, from an operational perspective.
In asterisk, you can originate a call that has the 2 variables you need with an (basic-authenticated) HTTP request. You will also need some settings and a tiny dialplan. Setting up the SIP account is easier or more difficult, depending on the documentation from the provider. Most of them have detailed documentation for configuring asterisk (not so much so for freeswitch). Keeping the damn thing alive is what's gonna get to you :)
I am asking this question as a small part of my question series regarding game programming .
Refer this question as the main one .
Now suppose I want to develop a game on iphone - a simple online multiplayer board game.
Suppose its a casino table.
In the main question ChrisF has tell me of sort of Client - Server architecture for iphone online multiplayer game.
I want to know what sort network programming I have to do for this type of application .
What will be the responsibilities and activities carried out by client and server .
You can provide me link, tutorials or to the point answers , anything will be great help for me and will be really appreciable .
thanks
You'll want to write a socket application running on a server. When you have access to a wifi access point or edge/3g you can send data to it from your iphone application. This server can then handle the incoming data and send an appropriate reply to the people connected.
For server socket programming, take a look at this guide - http://beej.us/guide/bgnet/.
For iphone specific socket programming, take a look at the samples supplied with the Iphone SDK. This link also has some basic information.
A simple online multiplayer board game
Given that the iPhone isn't always connected to the internet, you might need a server to store state. Alternatively you could always stipulate that if one person loses their connection the game finishes.
Client to client would be the obvious choice for the latter. Both clients have a port they listen to, and send the other commands based on the board state. Like almost all online games the obvious choice would be to use UDP as it's fast and compact.
For the server architecture you will of course need some kind of server listening for commands and a game number. It would store your state in a datastore on the server, a mySQL database for example. UDP or even SOAP or JSON over HTTP would be the two obvious choices for this.
This second approach using JSON/SOAP route would be a lot easier for you to get started with, assuming the iPhone has a decent JSON or SOAP library which is not likely. I have no idea about UDP in Objective C, however in C it requires a certain level of knowledge which won't get you something to play with quickly.
As you already said, you will need a server, but you can have two kinds of design:
The server can serve only as a gateway between the players to connect one to each other: it's two uses are, first, to list the running games, and second, to list the IP addresses of the players so that each client will read the IP addresses and connect to them. This will require less bandwidth and processing power for the web server and more for the client which will host the game. In this configuration, one of the clients will act as a server and will send data to the others. If your game has only one of the players playing at a time, and not a huge lot a players, this is what you should use as what you pay is the server's power.
The server can also store all games' states: it might require far more processing power and/or bandwidth depending on your game.
Anyway, most of the time you will want only one machine (which can change during the game as in the first case) to do the processing and the others will only receive the computed data.
To program a networked game, you will need knowledge of sockets (deep knowledge in the first case because you will have to deal with NAT issues, routers blocking the way between clients). The guide to that is Beej's Guide to Network Programming, the number one ressource on this topic, although it doesn't focus on games.
If not too much processing is needed on the WWW server, you could deal with it with server scripting languages like PHP along with MySQL, but you're most likely to use your own server programmed in C++ (or in C). If you use C++, you might want to use an already existing library such as RakNet.
The client will obviously be programmed in Objective-C as it's on the iPhone. I believe there is a good networking framework looking at the number of online games, so you might as well not want to use an external server networking library.
It might be a bit too much for what you want to do, but you could also use the well known Torque Engine.
By looking at current Google wave APIs, I can't find a way to create an alternative client.
It's not a robot or gadget, and the embed API is very slim.
Nevertheless, I do see some clients out there - such as Waver and Waveboard.
How do they do it ? is it based on XMPP ?
Note that Waver and Waveboard aren't actual clients, rather single-application web-browsers wrapped around the official https://wave.google.com/wave/ URL.
The Wave Federation protocol comes with a Protocol Buffers based experimental client/server protocol. Some people are using that to make a client, but it's not yet interoperable with the existing wave infrastructure unless you set up your own server (it won't work with #googlewave.com users, you must set up your own wave federation server and have it communicate to that).
During Google I/O they announced the Google Wave Data API which allows a program to read and write to wave on behalf of a user using OAuth. I'm using it to create a true mobile client, but at the current state, it's still very limited and restricted to the actions of fetchWave, search and folderAction (markAsRead/Unread, mute and archive).
http://code.google.com/apis/wave/extensions/wavedataapi/index.html
Lars Rasmussen did mention the beginning of a public client/server protocol, but I can't yet find anything about it.
IVR Systems are so expensive and proprietary.
The vendor systems I've used are all pretty weak... What is a good alternative to Genesys, Intervoice, etc.?
Asterisk is open source and it includes the capability of defining and running IVR interactions.
They have proprietary hardware for interaction with the PSTN or you can build services on top of VoiceOverIP protocols such as SIP (i.e. IP only).
See the Asterisk Gateway Interface wiki page - this describes the primitives you would use in the IVR script.
I use Asterisk for IVR implementations. I don't think the learning curve was too steep, especially considering I'm not really into telecoms, and it's worked pretty well for what I've used it. I currently have an IVR with 8 E1 links (240 channels) with Asterisk and a Sangoma card, works pretty well. And it's easy to test new features, since I can run it in my machine with just VoIP, setting up SIP in Asterisk and using the X-Lite softphone.
I connect Asterisk with my Java application using FastAGI and JAGIServer/OrderlyCalls on the Java side.
I would recommend Katalina 'VoiceGuide' - a windows-based CTI, IVR and dialler system at a reasonable price, supporting both tradtional POTS/ISDN telephony cards and VoIP. see: www.voiceguide.com