Hi good people and i am needing to make use of telscale smsc. i have been studying about it for almost 2weeks and doubt about how i can meet my needs. And just need an smpp connection that works with (kannel & playsms)
i just need someone so kind to direct to right path combining telestax to work in a standard mode for its smsc capabilities with out no third party operator and building it from the community version source code available on github.com/restcomm/restcomm-connect
i just run into confusion tryna combine this things to work for me, using its smpp capabilities.
first i need various intallation to make, i know of restcomm smsc, sip serverlet, metaswitch(clearwater), jboss,
please i need steps to follow realizing my aim as i have information and no clew on how to combine them. below are the straight question that i need answers to.
1: can i build commuinity version of restcomm smsc to relay real message to a real mobile carrier eg VERIZON mobile carrier relying only on opensource available resources using it for real with out and no third party mobile operator to negotiate with over sms termination
2: if yes, what are the necessary installation and configuration needed to be made by me to make the smpp connection found arround telscale smsc to send real message to a mobile phone connecting with ESME like KANNEL.
Related
I want to develop a VoIP app with Xamarin and Xmpp server.
So far the only things that I have found is the openfire and "jitsi meet" for the server side and matrix for the client side. But the matrix has nothing to do with voice streaming and is just for text messaging and "jitsi meet" doesn’t have any sdk for .net client side.
I also have found the red5pro but this has client sdks just for native android and ios development platform and has nothing for Mono.
So what Should I look for?!
First, let's clarify some basics:
openfire is a XMPP server. Basically, this is all you need on the server side for basic VoIP support.
Alternatives include ejabberd and Prosody.
jitsi meet essentialy already is a VoIP app, so if you want to develop your own, you don't really need that.
"Jitsi Videobridge" on the other hand can be used to provide a relay server for video conferences. For your first steps with a simple VoIP app, you wont need that either, but if you want your users to be able to create video conferences with many participants, then this helps.
(Explanation: Normally, when you create a P2P-Video conference, you
have two options: First, all users send their video data to all
participants (everybody needs lots of bandwidth), or you pick one
participant ("host") that receives the video streams of every
participant end sends them to every other participant. In the second
case, a normal participant only has to upload his stream once and
download n streams, whereas the host does most of the work - so only
that one user needs high bandwidth.
Jitsi Videobridge can run on a server and act as this conference host (usually a server has a much better bandwidth than a home user), so that none of the participants has to act as a host.
In simple VoIP applications (without video), this may not be neccessary, as audio streams are usually much smaller than video streams.)
Now, as I said above, in order to write a VoIP app, you basically only need a XMPP server (openfire, prosody and ejabberd should all be sufficient for this use case), a client library that supports Jingle and client libraries for the RTP media streams (transfer and display).
Jingle is the name of a XMPP protocol extension that enables the negotiation of P2P data streams as they are needed for a VoIP call.
The relevant protocol specifications:
XEP 0166: Jingle
XEP-0167: Jingle RTP Sessions
So what you need to find is a XMPP library with support for the jingle protocol. The C# Matrix XMPP SDK (not to be confused with the "Matrix protocol", which is a different protocol and has nothing to do with XMPP except for having a common goal) is one example of such a library. According to their web site, there is support for Jingle, but I couldn't find any documentation about it.
However, as I mentioned above, Jingle is only about how to negotiate data streams, not the data streams and VoIP itself.
So what that library probably helps you with is parsing of the Jingle XMPP messages that are needed to set up a RTP data stream.
For displaying and transfering the RTP stream, however, you need additional libraries. For that, have a look at the following SO questions and answers:
Open Source .net C# library for Real Time transport Protocol
Streaming Avi files from C# using RTP
I hope I could give you some useful hints...
Is WebRTC going to be free for web developers to set up video calls on web pages?
why does Twilio having pricing 25c per mins for video calls,
is it going to be too expensive for the small guy to mange video calls on web hosting servers?
any advice from anyone deep into WebRTC already?
Some of the comments above are not well informed.
Someone wrote, since the bandwidth needed in case of media relay is higher as well. This is not entirely true, transmission happens between Peers(Browsers), servers are used just for signalling(relaying IP addresses of connecting peers and some more info), you can ROUTE your transmission from central server(for fail overs), but can surely do without it for free.
WebRTC is Free and you can setup the whole thing on your own without having to shell out even a penny. It is a bit hard and mitigating fail-overs is really difficult, but you can certainly do it for free.
Tokbox or Twilio charge money because these tools abstract some very rigid complexities of setting up, running and managing fail-overs in a WebRTC application.
In TokBox's Case:
You don't need to setup STUN, TURN servers, you don't have to worry
about integration with android or IOS clients, they provide a plugin
for IE too, so out of box you get everything and you just have to
concentrate on your application logic rather than WebRTC nuances.
This is a big plus.
Both RELAY and ROUTED schemes came in the box hence you can write
fail-over scenarios if RELAY communication fails. Although there are
some good JavaScript based frameworks that do this in a much cleaner
manner.
It adds slew of other goodies which help in building android and IOS
clients without any pain.
STUN or TURN Servers are used only for Signalling Purpose, and this signalling happens before any actual transmission. This signal is very small and carries the IP address of both the browsers(machines running browsers). For Transmission the communication is done between Browsers(Peer to Peer) themselves, so no server is involved.
Your relay is not happening from a central server so you don't have
to pay for the outgoing bandwidth cost.
To Setup Turn Server,
Use this server, build it and put it into a Rackspace/Amazon Web
Services instance and you are Good with your TURN
Server. That is It, setup your application and have fun with WebRTC
for FREE.
rfc5766-turn-server
If you wish to Use some more free framework to ease yourself more, check out: EasyRTC and PeerJS
Enjoy Developing with WebRTC....
Twilio developer evangelist here.
Your link at the end of your question points to our WebRTC page, which currently talks about the product Twilio Client. Twilio Client briefly is a way that, using WebRTC within browsers and mobile applications you can make phone calls to real phone numbers. This product does not allow you to conduct video calls.
Twilio Client has a cost because of the ability to call out from a browser to a telephone number. The cost is not in the WebRTC portion, but delivering those minutes to the other leg of the call.
Notably, it's not 25 cents ($0.25) a minute, instead it is just a quarter of a cent ($0.0025) a minute.
With regards to video calls with WebRTC, you can now access the public beta of Twilio Video, a platform to make setting up WebRTC calls much easier.
Twilio Video costs for the signalling infrastructure and you can see the prices here. If a WebRTC connection requires a TURN server to relay the media, that also costs per gigabyte of transfer. Usage of the STUN server is free, the costs for the TURN relay are available here.
Please get in touch with me at philnash#twilio.com if you have any other questions about WebRTC.
WebRTC is a technology placed in a browser. It requires backend infrastructure to support it - specifically, STUN and TURN servers as well as signaling servers.
This boils down to the fact that you pay for WebRTC - same as you pay for hosting your website on a server. The price is higher, since the bandwidth needed in case of media relay is higher as well.
To understand more about WebRTC and how it works (as well as why there's a price tag associated with services such as Twilio for it), you can check this free report: https://bloggeek.me/webrtc-business-people/
WebRTC is already free for developers to use. When we added WebRTC to our product, we used this example code, which made it very simple to build a WebRTC client:
https://shanetully.com/2014/09/a-dead-simple-webrtc-example/
Google and Mozilla provide free STUN servers, and it is easy to set up a TURN server. Most clients will be able to connect via STUN, so you won't end up using too much bandwidth on your TURN server.
To set up your own TURN server, coturn seems to be the easiest to set up:
https://github.com/coturn/coturn
Make sure you read the "WEBRTC USAGE" section in the README.turnserver file.
"STUN or TURN Servers are used only for Signalling Purpose, and this signalling happens before any actual transmission. This signal is very small and carries the IP address of both the browsers(machines running browsers). For Transmission the communication is done between Browsers(Peer to Peer) themselves, so no server is involved."
if that is the case, then you should be able to do this on a standard web server using Java/php. PHP will get the IP address of the guys connected to it. Then its just a matter of storing them in MySQL, then making a javascript that would run when the user go to that page in the site.
I've been looking for a solution around using a VPS because running a dedicated server for signaling is like golfing with a Ferrari instead of a golf cart. I still don't think node is efficient. Its single threaded. so node's fararri can only go 5mph.
Since they went to the web site, the php service already can get their ip address what else does it need? All of the above solutions so far require you to pay for a dedicated app to run on a server connected to the web separately for what 5k of data? What a waste of electrons.
But I'm going to start a new thread that is going to be based on getting webrtc without the buy a "VPS" because we want a VPS-less solution.
I have a local installation of Voxeo's Prophecy platform, and a voice xml application that runs on the voicexml browser of the platform. How can i call the application to listen to the message, without having a sip phone, and without using the voxeo's hosting capability? I just want to call the application at the local installation without using a phone. When i try this from the browser, i just get the xml file containing the dialog.
Why are you trying to start the application without a SIP phone? The developer version of Prophecy comes with a SIP phone that works great for testing and debugging. Are you trying to access the application from a regular telephone or POTS. If that is the case you will need additional hardware, such as a Cisco VoIP Gateway to translate the land-line signal to SIP. There are a number of VoIP Gateways available. Another option is to put Dialogic cards in your server and use a software that translates between SIP and the Dialogic API. If you purchase Prophecy from Voxeo as an appliance there are options they can sell you to achieve this. The advantage of getting this from Voxeo is that they will help configure it, which is not trivial depending upon the type type of telephone lines you are using and whether they are behind a PBX or not.
The telephony/communications interface to Prophecy is SIP so that is the only way to communicate with it. You could use any open source SIP stacks to develop something yourself, but the easiest thing to do is to use a software based SIP phone. It is not like you could just start talking and the VXML app would know what you intended, or if the if the voice input was really intended for the VXML app. You still need to maintain things like sessions and routing to the correct application which is easily handled by a SIP phone. Prophecy is designed for a telephony environment. If you are trying to use it for something like a home automation system, which some people have, I would think you would have to provide some type of SIP front end that is voice activated.
I am planning a software application where the user will be able to select a given media channel from a list of RTMP streams available on one or more media servers on the internet. The list should ideally be dynamically created through some kind of service that knows about the available and active channels.
My question is: Would this be possible through some kind of protocol between the service and the media server. I understand that RTMP by itself doesn't allow this. A therefore assume that some outbound mechanism will be required.
No...
...there is no native application discovery in RTMP. If you'd like this kind of functionality you'll need to program some sort of discovery service for which ever streaming server you are running.
I am working with an electronics appliance manufacturer to embed LAN based control systems into the products. The idea is to serve up a system configuration/control interface through a web browser so clients never need to install software. We can communicate with the appliance by sending and receiving serial data through the embedded module. Since the appliance can also be controlled from a front panel UI, it creates a challenge to keep a remote web interface in sync with very low latency. It seems like websockets or some sort of Push is what we need for handling real time events from the server to clients.
I am using a Lantronix Mathport AR embedded device server. Out of the box the unit will serve up any custom HTML and java servlets/applets. We have the option to install a lightweight Linux distro if we need more flexibility. I am not sure how to implement any server side apps since the device is not running standard Apache. I believe it is using Boa.
Can anyone guide me in the right direction of how to do this?
Some general info...The WebSocket protocol (draft spec here) is a simple layer on top of TCP. What this means is that, if you already have a TCP server for your platform, implementing the WebSocket is just a matter of hours. The protocol specifies a handshake and two ways of sending data frames.
I strongly suggest you start by reading the 39 pages spec.
As Tihauan already mentioned, start by reading the spec, and also note that there are still some changes ongoing, although websockets is now more stable than it was 1 year ago.
Key point for me was the requirement that websocket data is entirely UTF-8 text, which lends itself nicely to JSON based message definitions.
Our system uses a form of embedded linux, so we then added and made use of the following libraries:
"libwebsockets" from:
http://git.warmcat.com/cgi-bin/cgit/libwebsockets/
"jansson" from:
http://www.digip.org/jansson/
Using the above as support libraries, we created an internal lightweight "client/server" that allowed our other software modules to register for certain, applicable, websocket messages, and respond as needed. Worked great.
Good luck and best regards,
I'm a bit late, but Mozilla posted a guide entitled "Writing WebSocket servers", which literally guides you through writing a websocket server.
You will need to already know how HTTP works and have medium programming experience. Depending on language support, knowledge of TCP sockets may be required. The scope of this guide is to present the minimum knowledge you need to write a WebSocket server.
https://developer.mozilla.org/en-US/docs/Web/API/WebSockets_API/Writing_WebSocket_servers