one of my client want following functionality in his site.
Person 1 will call person 2 from our website
When person 2 answers person 1 will hear his voice on website
person 1 will click on any mp3 file on website while call is going on and person 2 will listen to that mp3 file
and what person 2 speak will be played on our website so tha...
Hi,
I have successfully installed the Openfire XMPP server and built an Android app to text chat,
How can I implement VOIP functionality to my application?
Can this be done by connecting openfire with a SIP server?
Can someone guide me?
...
Hi all,
I need to test an implementation of the SIP/IMS application which is behind a NAT. FOr the same, I am looking for a public TURN server that can be used for a limited purpose to test the application.
I did look at numb.viagenie.ca, but it seems to be based on older version of the specs. Are there any other public TURN servers ...
I'm looking for best way to initiate call from a web page, and am trying to choose between the callto: and the tel: URL protocol.
The link will be a normal telephone number, not a Skype account, and I'm trying to support as many platforms as possible, including Skype and other similar VOIP applications, as well as Smartphones like the ...
Hello there.
I was wondering if it was possible to have a chatting feature on my site(VOICE NOT TEXT)
is this possible on a php enabled server(GoDaddy) or do i need like a flash type server.
Also if it will work how can i start.
Thanks So Much!!!
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Hello there
i wanted to know if it was possible to make a voip script in python and then if it works integrate it online on my web site
thanks a bunch
...
I assume that it is possible to invoke skype or another VOIP solution by starting the appropriate activity. My question is; are there any VOIP solutions that can be used as services in some way, making it possible to write a separate app with its own user interface that just uses the VOIP functionality of the original VOIP app?
This que...
Hi friends,
i have a scenario where i have to use the automatic calling functionality using VOIP (Diagrammatically).
I need a system or dialler that will automate the process of dialling and get response through the system. We need a tool that could be able to dial by itself (we don’t have any agent interaction) the given number and wi...
Hi,
I want to build a SIP client in java. A java class that will register with a SIP registrar and will be able to call another sip client.
Any suggestions?
...
libjingle's developer guide quite vaguely mentions incompatibilities between libjingle's implementation of Jingle and XEP-0166's specification. It also mentions it is incompatible with XEP-0167, XEP-0176 and XEP-0177.
It never goes into detail what these differences are; in fact, it is unclear if perhaps these differences have been res...
Does anyone know how to use Android device's built-in acoustic echo cancellation? It is located somewhere in silicon and is used for GSM/CDMA speakerphone calls. We'd really like to tap into it for a VoIP application instead of rolling our own.
Ben
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I want to encrypt voice calls on the fly. so which programming language should be preferred for symbian os. Are any API available for this purpose. which channel could be preferably used?
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Hi!
I want to mix to multiple ULAW samples together in Java. Is there a way to do so without converting to PCM? Im trying to do a 8000Hz 8bit VoIP app. Im testing with AU files (created in goldwave) since they use the ULAW encoding. My current implementation is:
[code]
AudioFormat f = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED...
OK! Im trying to setup a Mu-Law based Voip server and client in Java. Im using Javasound wrapped with DataSource and PushBufferStream for audio capture. I'm using JMF for Transcoding the LINEAR audio to ULAW and a DataSink wrapped around a UDP socket for data transport. I'm trying to save bandwidth so the implementatioon is a Push-To-Tal...
Iam developing a VOIP platform which would allow users to make 100s of calls concurrently using my service.
Asterisk stores all call detail records in the CDR table. I would like to know where is best place to keep this table for the best possible architecture of my system.
Should I keep it on the Asterisk Box and run a cron to sync it...
I have been @ this for hours and keep comming around to dead ends. There as another article here and it was some help but no real answer that could help me out in my particular application given the depth of the answer. I want to use the "Trick" method that the above post (VIEW POST) but I am always getting the unauthorized error. Throug...
Hi,
I am looking for good SIP, telecom and VOIP resources. In particular videos, talks and presentations that are considered "a must see" from the community.
Do you suggest anything?
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Are there any HOWTO's?
I want to send and recieve faxes in my rails app. So I'm looking for books/articles/libraries to read
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