I\'m interested in programatically answering a VoIP call, preferably within the Skype application, using my app\'s background service. So I would need to be able to detect an incoming Skype call someh
I am trying to use gnugk and openmcu for a video conferencing application. As per my configuration, both gnugk as well as openmcu run in same machine ( 10.21.34.2).
I found that you can do that by making some calcu开发者_StackOverflow社区lations on the sequence numbers from the RTP header. But the sequence numbers are stored in 16bits and could easily exceed the
I am making a VOIP application for m开发者_运维百科obile platform. My question is what algorithms should be used to calculate whether the RTP package is \"expired\".
I need to create an application that has multiple SIP connections and uses multiple soundcards as input / output devices. My background is C#, so .Net is preferred, but I\'ll work with wrappers / IKVM
I did like this : 1. install red5,asterisk 2. setting environment variable, such as java, apache-ant 3. configure sip.conf at asterisk
I have to calculate time offset between packets in RTP streams. With video stream encoded with Theora codec i have timestamp field like
I am working on a project where I need to play audio files over VOIP channel. I am using OpenSource phone (SFLphone). I would like to know how to play an MP3 audio file over VOIP channel.
I have 2 UACs connected to FreeSWITCH. Party 1 calls party 2. Party 2 rejects a call (either with \'Decline\' or \'Busy here\'). But FreeSWITCH does not send \'Decline\' to party 1. Instead, it sends
The Speex docs say that it\'s \'mainly\' designed for 8/16/32kHz sampling rates. Most PC inputs seem to report sampling rates of 8kHz, 16kHz and 44.1kHz.
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