I am new to Symbian development. I am developing a VOIP [SIP protocol] app for Nokia Symbian S60 3rd edition. I want to send DTMF signal through my app. But couldn't find any clue. So how can I send and catch DTMF signals in my app.
Thanks in advance.
in VoIP world you send SIP INFO messages with the DTMF data to the SIP server. For example to send the digit '1':
INFO sip:7007471000#example.com SIP/2.0
Via: SIP/2.0/UDP alice.uk.example.com:5060
From: <sip:7007471234#alice.uk.example.com>;tag=d3f423d
To: <sip:7007471000#example.com>;tag=8942
Call-ID: 312352#myphone
CSeq: 5 INFO
Content-Type: application/dtmf-relay
Content-Length: 22
Signal=1
Duration=160
See also this link.
If you want "traditional" DTMF messages over the RTP, the best way is to have the DTMF signals recorded (1.wav, 2.wav, etc) and play them it back during the call.
Related
I am looking for a service
which can take reminders through alexa, pc(debian, mac or windows)
and mobile (android or iOS).
should be able to dispatch notifications
should sync between devices
As of today there is not. However, I know that Urban Airship is working on that. Here is a link to their announcement about it.
https://www.urbanairship.com/blog/voice-notification-early-access-program
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...
I recently got the Xoom, for some reason I thought you could SMS message on it :-S
Does anyone know a way that from my android phone I can push a notification to my tablet to tell me I have a message or even if someone is calling.
Basically I want to mainly use my tablet and keep my phone in my laptop bag but be notified if I get a message.
Any help will be greatly received.
You will need to develop an application which can talk to another instance of the same application over Bluetooth (or Wi-fi). The application running on the phone can listen for the broadcast of an incoming SMS, and send the info to the paired device (Xoom) where it will throw a Notification.
If anyone is interested it seems HTC have solved this.
HTC have an online service for sense found at: https://www.htcsense.com
From here once you have linked your Google account you can read and send text messages pushing them through your phone.
This allows me to use the browser on my Xoom tablet to send SMS messages.
Are there any packet capture tools (like tcpdump and wireshark) for Nokia N8/ symbian ^3 phones! I am currently working on my thesis and have almost done with the application layer behavior using JavaScript but am required to study the network behavior of HTML5 web applications on symbian phones.
Any help would be highly appreciated.
There is an app available at betalabs.nokia.com called Nokia Connectivity Analyzer:
http://betalabs.nokia.com/apps/nokia-connectivity-analyzer
I have not tried this part of it, but the "Connection Observer" component claims it can do ip packet capture.
The Symbian browser is based on Webkit
I'm building a VoIP application using the TeamSpeak 3 SDK. Is there any way to view the UDP packets going in and out of the server/client applications?
Any packet sniffer such as Wireshark (formerly Ethereal) can do that for you.
Did you already try Wireshark?