Voice over IP - Revision history

WikiSysop: /* C# DirectX Libraries */

2012-09-14T11:03:59Z

C# DirectX Libraries

← Older revision		Revision as of 11:03, 14 September 2012
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	SlimDX is very mature, and fully featured. Some larger scale, commercial games have been written and published using SlimDX. It also provides more of a "framework" to use, and has more feature-complete documentation.		SlimDX is very mature, and fully featured. Some larger scale, commercial games have been written and published using SlimDX. It also provides more of a "framework" to use, and has more feature-complete documentation.

	SharpDX promises slightly better performance in certain scenarios (see benchmarks). It's generated directly from the DirectX headers, so is more of a thin DirectX wrapper.		SharpDX promises slightly better performance in certain scenarios ([http://code4k.blogspot.com/2011/03/benchmarking-cnet-direct3d-11-apis-vs.html see benchmarks]). It's generated directly from the DirectX headers, so is more of a thin DirectX wrapper.

	* [http://slimdx.org/ SlimDX is a free open source framework that enables developers to easily build DirectX applications using .NET technologies]		* [http://slimdx.org/ SlimDX is a free open source framework that enables developers to easily build DirectX applications using .NET technologies]

2012-09-14T11:03:06Z

Streaming Audio from C#

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 Kamailio and SER has now [http://sip-router.org/2008/11/04/the-sip-router-project-launched/ joined] in [http://sip-router.org/ The SIP Router Project].
 * [http://sip-router.org/ The SIP Router Project - Homepage]
 ==Streaming Audio from C#==

2010-03-04T14:05:46Z

Jitter

← Older revision		Revision as of 14:05, 4 March 2010
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	====Jitter====		====Jitter====
	* [http://toncar.cz/Tutorials/VoIP/VoIP_Basics_Jitter.html VoIP Basics: About Jitter]		* [http://toncar.cz/Tutorials/VoIP/VoIP_Basics_Jitter.html VoIP Basics: About Jitter]
			* [http://www.rfc-archive.org/getrfc.php?rfc=3550 RTP: A Transport Protocol for Real-Time Applications (A.8)]

	====Department of Computer Science at San Diego State University====		====Department of Computer Science at San Diego State University====

2010-03-04T13:57:45Z

← Older revision		Revision as of 13:57, 4 March 2010
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	==Information about RTP, RTCP, SIP, RSTP, IAX2==		==Information about RTP, RTCP, SIP, RSTP, IAX2==
	==Jitter==		====Jitter====
	* [http://toncar.cz/Tutorials/VoIP/VoIP_Basics_Jitter.html VoIP Basics: About Jitter]		* [http://toncar.cz/Tutorials/VoIP/VoIP_Basics_Jitter.html VoIP Basics: About Jitter]

2010-03-04T13:56:23Z

Information about RTP, RTCP, SIP, RSTP, IAX2

← Older revision		Revision as of 13:56, 4 March 2010
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	==Information about RTP, RTCP, SIP, RSTP, IAX2==		==Information about RTP, RTCP, SIP, RSTP, IAX2==
			==Jitter==
			* [http://toncar.cz/Tutorials/VoIP/VoIP_Basics_Jitter.html VoIP Basics: About Jitter]

	====Department of Computer Science at San Diego State University====		====Department of Computer Science at San Diego State University====
	Their Communication Networks Laboratory has this nice presentation with a lot of demystification.		Their Communication Networks Laboratory has this nice presentation with a lot of demystification.

2009-06-08T10:22:05Z

← Older revision		Revision as of 10:22, 8 June 2009
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	* [http://www.microsoft.com/downloads/details.aspx?familyid=ED1CCE45-CC22-46E1-BD50-660FE6D2C98C&displaylang=en Microsoft Office Communicator 2007 SDK]		* [http://www.microsoft.com/downloads/details.aspx?familyid=ED1CCE45-CC22-46E1-BD50-660FE6D2C98C&displaylang=en Microsoft Office Communicator 2007 SDK]
	* [http://blogs.microsoft.co.il/blogs/tamir/archive/2008/02/17/sound-tone-and-dtmf-generation-by-using-managed-directsound-and-c-and-sine-tone-detection-with-pure-managed-goertzel-algorithm-implementation.aspx Sound, tone and DTMF generation by using managed DirectSound and C# and sine tone detection with pure managed Goertzel algorithm implementation ]		* [http://blogs.microsoft.co.il/blogs/tamir/archive/2008/02/17/sound-tone-and-dtmf-generation-by-using-managed-directsound-and-c-and-sine-tone-detection-with-pure-managed-goertzel-algorithm-implementation.aspx Sound, tone and DTMF generation by using managed DirectSound and C# and sine tone detection with pure managed Goertzel algorithm implementation ]
	~~<google>ENGELSK</google>~~

	=====And some fun :-)=====		=====And some fun :-)=====
	* [http://tools.ietf.org/html/draft-kaplan-sip-four-oh-00 Session Initiation Protocol (SIP) Version 4.0: P2P2PSIP]		* [http://tools.ietf.org/html/draft-kaplan-sip-four-oh-00 Session Initiation Protocol (SIP) Version 4.0: P2P2PSIP]

			<google>ENGELSK</google>

2009-06-08T10:21:19Z

Misc

← Older revision		Revision as of 10:21, 8 June 2009
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	* [http://blogs.microsoft.co.il/blogs/tamir/archive/2008/02/17/sound-tone-and-dtmf-generation-by-using-managed-directsound-and-c-and-sine-tone-detection-with-pure-managed-goertzel-algorithm-implementation.aspx Sound, tone and DTMF generation by using managed DirectSound and C# and sine tone detection with pure managed Goertzel algorithm implementation ]		* [http://blogs.microsoft.co.il/blogs/tamir/archive/2008/02/17/sound-tone-and-dtmf-generation-by-using-managed-directsound-and-c-and-sine-tone-detection-with-pure-managed-goertzel-algorithm-implementation.aspx Sound, tone and DTMF generation by using managed DirectSound and C# and sine tone detection with pure managed Goertzel algorithm implementation ]
	<google>ENGELSK</google>		<google>ENGELSK</google>

			=====And some fun :-)=====
			* [http://tools.ietf.org/html/draft-kaplan-sip-four-oh-00 Session Initiation Protocol (SIP) Version 4.0: P2P2PSIP]

2009-05-21T05:47:50Z

SIPp

← Older revision		Revision as of 05:47, 21 May 2009
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	:'''sipp -sn uac -i <local ip> <local ip>'''<br>		:'''sipp -sn uac -i <local ip> <local ip>'''<br>
	Where '''<local IP>''' is your IP address on your local LAN network, and '''-i''' sets the local IP address for 'Contact:','Via:', and 'From:' headers.<br>		Where '''<local IP>''' is your IP address on your local LAN network, and '''-i''' sets the local IP address for 'Contact:','Via:', and 'From:' headers.<br>
	~~It looks like 127.0.0.1 can't be used as IP, at least on Vista.<br>~~
	If you start SIPp without parameters you get the help.<br>		If you start SIPp without parameters you get the help.<br>
	The last tho lines is an example.<br>		The last tho lines is an example.<br>

2009-05-21T05:25:08Z

SIPp

← Older revision		Revision as of 05:25, 21 May 2009
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	To call the server on the default IPv6 address which is ::1, You must use:		To call the server on the default IPv6 address which is ::1, You must use:
	:''sipp -sn uac ::1''		:''sipp -sn uac ::1''
			For the same reason always specify th -i switch. It will also default to ::1 if not specified.

	==Dictionary==		==Dictionary==

2009-05-21T05:23:39Z

SIPp

← Older revision		Revision as of 05:23, 21 May 2009
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	In the second windows start the client with:		In the second windows start the client with:
	:'''sipp -sn uac -i <local ip> <local ip>'''<br>		:'''sipp -sn uac -i <local ip> <local ip>'''<br>
	Where '''<local IP>''' is your IP address on your local LAN network, and '''-i''' sets the local IP address for 'Contact:','Via:', and 'From:' headers~~. Default is primary host IP address~~.<br>		Where '''<local IP>''' is your IP address on your local LAN network, and '''-i''' sets the local IP address for 'Contact:','Via:', and 'From:' headers.<br>
	It looks like 127.0.0.1 can't be used as IP, at least on Vista.<br>		It looks like 127.0.0.1 can't be used as IP, at least on Vista.<br>
	If you start SIPp without parameters you get the help.<br>		If you start SIPp without parameters you get the help.<br>