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Voice over IP
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Soft Phones
SJphone
SJphone is a free VOIP softphone. It can be installed on:
- Windows (XP SP2, 2000 SP4, Vista, Vista x64), Windows CE (Pocket PC, Windows Mobile), Linux, MAC
It supports both SIP and H.323 standards, and has a brilliant logging feature if you are having trouble getting a call through.
Note for Vista x64 (64 bit version) users:
On my Dell Lattitude I couldn't make a call or run the Audio Wizard, it crashed the program. I found a workaround:
- Select: Menu -> To Advanced Mode
- Select: Menu -> Options
- Select: the Audio tab.
- Select: Advanced Settings...
- Uncheck: Enable Adaptive Echo Canceller
After that, I can make a call, and it is also possible to run the Audio Wizard....
I found this by trial and error, first I disabled everything in the 'Audio' tab, and reduced the sample rate, then I gradually enabled everything again.
In SJ Labs forum http://forum.sjphone.org/viewtopic.php?t=1589&highlight=aec you can read the following:
- Bugs and issues in 1.65
- 1. Although there is a Video tab on the Options panel, this build does not support Video.
- 2. AEC may work unstably if you use some old audio boards, or use different devices for input and output audio (for example, a USB mic and on-board audio system).
- 3. SJphone may deliver sound with less high frequency. In this case, reduce Driver sampling rate to 8000 (Options -> Audio -> Advanced Audio Settings).
This might explain my problem, but I was able to run it on the same Dell D620 without any problems, when it was running Windows XP.
Service Providers
Open source projects
SipX
The sipXtapi SDK is a C application programming interface for voice communications over IP. Specifically, sipXtapi provides a generalized telephony interface on top of the Session Initiation Protocol (SIP), RFC 3261, and the real-time Transport Protocol (RTP), RFC 1889. While the SIP and RTP protocols provide signaling and media transport infrastructure, sipXtapi also includes many other protocol and standards implementations needed for voice communications.
- sipXtapi: SDK Overview
- SipXtapi and sipXezPhone Build Environment for Windows - SIPfoundry sipx, The Open Source SIP PBX for Linux - Calivia
- SipXezPhone Introduction and Screenshot - SIPfoundry sipx, The Open Source SIP PBX for Linux - Calivia
- HowTo compile sipXezPhone - SIPfoundry sipx, The Open Source SIP PBX for Linux - Calivia
Help for compiling sipxmedialib.
- Re: sipxtapi-dev compiling sipxmedialib with ilbc msg00951
- Re: sipxtapi-dev compiling sipxmedialib with ilbc msg00953
Sofia-SIP Library
Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification (see the feature table). It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communication services. The primary target platform for Sofia-SIP is GNU/Linux. Sofia-SIP is based on a SIP stack developed at the Nokia Research Center. Sofia-SIP is licensed under the LGPL.
Streaming Audio from C#
- A low-level audio player in C#
- A full-duplex audio player in C# using the waveIn/waveOut APIs
- A C# Audio Recorder / Player Library
- MemoryStream and Media.SoundPlayer
- The waveInOpen function opens the given waveform-audio input device for recording
- The waveInClose function closes the given waveform-audio input device
- Audio Input/Output level
DirectSound
http://www.codeproject.com/KB/audio-video/DirectSound9p1/DirectSoundDemo.jpg
- The ultimate Managed DirectSound 9 Tutorial. Part 1: a full introduction to Playback
- Sound Experiments in Managed DirectX
- Sound Experiments in Managed DirectX
- WAVE PCM soundfile format
- How to use c# to write a sound recorder?
- DirectSound C# Generate White Noise / Single tone? - GameDev.Net Discussion Forums
- Tutorial under construction - VBForums
- Building a Drum Machine with DirectSound
- Coding4Fun
- Coding4Fun Beginning Game Development: Part VIII - DirectSound
- The Z Buffer \Managed DirectX Tutorials and Sample Code
.NET Compact Framework
Information about RTP, RTCP, SIP, RSTP
Department of Computer Science at San Diego State University
Their Communication Networks Laboratory has this nice presentation with a lot of demystification.
On this page http://medusa.sdsu.edu/network/CS596/Lectures/Lectures.htm?PAGE=Lectures you can find other network presentations from them. To navigate to this page from their main page http://medusa.sdsu.edu/network/ select CS596 -> Lectures.
Their presentations are based on Forouzan's book: TCP/IP Protocol Suite.
Henning Schulzrinne
Network Sorcery
The RFC Sourcebook is a great source for information on Internet protocols for the software professional. This guide is a reference for official networking standards and protocols.
Tools for testing and debugging VOIP calls
Wireshark
Network protocol analyzer for Windows and Unix that allows examination of data from a live network, or from a capture file on disk. It is free, easy to use and the best network analyzer you can get.
It recognizes VOIP calls and can play back the audio afterwards.
SIPp
SIPp is a free Open Source test tool / traffic generator for the SIP protocol.
Links
Tech-invite
- ABNF Grammar for SDP -- Session Description Protocol (RFC 4566)
- SIP Protocol Structure through an Example
Microsoft
Microsoft Real-Time Communications API - RTC
- Microsoft Real-Time Communications: Protocols and Technologies
- Integrating Rich Client Communications with the Microsoft Real-Time Communications API
- Enhancing Rich Client Communications with the Microsoft Real-Time Communications API
Misc
- MSDN Webcast: Visual C# Express 2005: Creating an Instant Messaging Application Part 2 of 3
- Ever heard a PBX talk!
- Microsoft Office Communicator 2007 SDK
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