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:

1. Select: Menu -> To Advanced Mode
2. Select: Menu -> Options
3. Select: the Audio tab.
4. Select: Advanced Settings...
5. 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.

• SJ Labs Homepage
• SJ Labs Download page

Service Providers

• Redspot Danish service provider
• everlove another Danish service provider

Open source projects

SipX

• SipXezPhone Introduction and Screenshot - SIPfoundry sipx, The Open Source SIP PBX for Linux - Calivia
• SipXtapi and sipXezPhone Build Environment for Windows - SIPfoundry sipx, The Open Source SIP PBX for Linux - Calivia
• HowTo compile sipXezPhone - SIPfoundry sipx, The Open Source SIP PBX for Linux - Calivia

Links

• How to use the managed RTP API classes in .NET to create your multicasting systems

• Microsoft Real-Time Communications: Protocols and Technologies
• MSDN Webcast: Visual C# Express 2005: Creating an Instant Messaging Application Part 2 of 3
• SIP stack with SIP proxy - (VOIP)

• STUN (Simple Traversal of UDP (User Datagram Protocol) through NATs (Network Address Translators))

Tech-invite

• ABNF Grammar for SDP -- Session Description Protocol (RFC 4566)
• SIP Protocol Structure through an Example

Microsoft

• Ever heard a PBX talk!
• Microsoft Office Communicator 2007 SDK

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