Voice over IP: Difference between revisions

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==Information about RTP, RTCP, SIP, RSTP==
==Information about RTP, RTCP, SIP, RSTP==
====Department of Computer Science at San Diego State University====
====Department of Computer Science at San Diego State University====
* [http://medusa.sdsu.edu/network/ Communication Networks Laboratory]
Their Communication Networks Laboratory has this nice presentation with a lot of demystification.
Has this nice presentation with a lot of demystification.
* [http://medusa.sdsu.edu/network/CS596/Lectures/ch28_RT.pdf Multimedia over IP (RTP, RTCP, SIP, RSTP)]
* [http://medusa.sdsu.edu/network/CS596/Lectures/ch28_RT.pdf Multimedia over IP (RTP, RTCP, SIP, RSTP)]
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'',


==Links==
==Links==

Revision as of 07:29, 8 April 2008

<google>ENGELSK</google>

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.

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.

Help for compiling sipxmedialib.

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#


DirectSound

http://www.codeproject.com/KB/audio-video/DirectSound9p1/DirectSoundDemo.jpg

.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,

Links

Tech-invite

Microsoft

Microsoft Real-Time Communications API - RTC
Misc

<google>ENGELSK</google>

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