RTP - Revision history

WikiSysop: Reverted edits by Yhorebapemy (Talk) to last version by Brain

2010-11-18T10:11:03Z

Reverted edits by Yhorebapemy (Talk) to last version by Brain

@@ Line 1: / Line 1: @@
->=== RTP Protocol===
+=== RTP Protocol===
 [[wikipedia:Real-time Transport Protocol|RTP (Real-time Transport Protocol)]], as a standard application protocol of the [[wikipedia:Internet Protocol Suite|Internet Protocol Suite]], has been designed for transport of multimedia streams in IP networks.
@@ Line 6: / Line 6: @@
 In multimedia applications low latency is essential, whereas an occasional  packet loss can be tolerated. A packet typically carries a fraction of a second worth of audio, which makes the lost packet almost inaudible. RTP adds a sequence number and a time stamp to each packet to allow detection and recovery from network errors.
 ===Payload Type===

Yhorebapemy at 23:51, 17 November 2010

2010-11-17T23:51:39Z

← Older revision		Revision as of 23:51, 17 November 2010
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Yhorebapemy at 22:57, 14 November 2010

2010-11-14T22:57:19Z

← Older revision		Revision as of 22:57, 14 November 2010
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	=== RTP Protocol===		>=== RTP Protocol===

	[[wikipedia:Real-time Transport Protocol\|RTP (Real-time Transport Protocol)]], as a standard application protocol of the [[wikipedia:Internet Protocol Suite\|Internet Protocol Suite]], has been designed for transport of multimedia streams in IP networks.		[[wikipedia:Real-time Transport Protocol\|RTP (Real-time Transport Protocol)]], as a standard application protocol of the [[wikipedia:Internet Protocol Suite\|Internet Protocol Suite]], has been designed for transport of multimedia streams in IP networks.
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	In multimedia applications low latency is essential, whereas an occasional packet loss can be tolerated. A packet typically carries a fraction of a second worth of audio, which makes the lost packet almost inaudible. RTP adds a sequence number and a time stamp to each packet to allow detection and recovery from network errors.		In multimedia applications low latency is essential, whereas an occasional packet loss can be tolerated. A packet typically carries a fraction of a second worth of audio, which makes the lost packet almost inaudible. RTP adds a sequence number and a time stamp to each packet to allow detection and recovery from network errors.

			----
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	===Payload Type===		===Payload Type===

Brain: /* Payload Type */

2009-03-27T11:13:09Z

Payload Type

← Older revision		Revision as of 11:13, 27 March 2009
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	===Payload Type===		===Payload Type===
	~~describes~~ the ~~transport~~ of ~~a particular~~ media ~~format~~. ~~Some values of the~~ audio payload ~~formats~~ are standard ~~others need~~ to be ~~described in a service description protocol, SDP~~ (Session ~~DescriptionProtocol~~);
			Each RTP packet carries 7-bit information describing the type of the media data - so called payload type. For most common audio and video formats standard payload types are defined. The non-standard types have to be negotiated by the communicating parties by other means/protocols (e.g. using the Session Description Protocol SDP). For that reason a range of <i>dynamic payload types</i> is reserved.

	The following table shows the defined RTP payload types used in Barix firmware.		The following table shows the defined RTP payload types used in Barix firmware.
	Note that Payload types 0, 8, 10, 11 and 14 are defined by the RTP standard while types 96 to 112 (dynamic payload types) are Barix specific.		Note that Payload types 0, 8, 10, 11 and 14 are defined by the RTP standard while types 96 to 112 (dynamic payload types) are Barix specific.
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	\|}		\|}


	===Sequence Number===		===Sequence Number===

Brain: /* RTP Protocol */

2009-03-27T11:02:45Z

RTP Protocol

← Older revision		Revision as of 11:02, 27 March 2009
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	Implementation is widely based on the [[wikipedia:User_Datagram_Protocol\|User Datagram Protocol (UDP)]]. The [[wikipedia:Transmission Control Protocol\|Transmission Control Protocol (TCP)]] is not often used by RTP for its high latency caused by connection establishment and error correction. The Barix implementation employs UDP as a transmission protocol.		Implementation is widely based on the [[wikipedia:User_Datagram_Protocol\|User Datagram Protocol (UDP)]]. The [[wikipedia:Transmission Control Protocol\|Transmission Control Protocol (TCP)]] is not often used by RTP for its high latency caused by connection establishment and error correction. The Barix implementation employs UDP as a transmission protocol.

	In multimedia applications low latency is essential, whereas occasional packet loss can be tolerated. A packet typically carries a fraction of a second worth of audio, which makes the lost packet almost inaudible. RTP adds a sequence number and a time stamp to each packet to allow detection and recovery from network errors.		In multimedia applications low latency is essential, whereas an occasional packet loss can be tolerated. A packet typically carries a fraction of a second worth of audio, which makes the lost packet almost inaudible. RTP adds a sequence number and a time stamp to each packet to allow detection and recovery from network errors.

	===Payload Type===		===Payload Type===

Brain: /* RTP Protocol */

2009-03-27T11:02:19Z

RTP Protocol

← Older revision		Revision as of 11:02, 27 March 2009
Line 5:		Line 5:
	Implementation is widely based on the [[wikipedia:User_Datagram_Protocol\|User Datagram Protocol (UDP)]]. The [[wikipedia:Transmission Control Protocol\|Transmission Control Protocol (TCP)]] is not often used by RTP for its high latency caused by connection establishment and error correction. The Barix implementation employs UDP as a transmission protocol.		Implementation is widely based on the [[wikipedia:User_Datagram_Protocol\|User Datagram Protocol (UDP)]]. The [[wikipedia:Transmission Control Protocol\|Transmission Control Protocol (TCP)]] is not often used by RTP for its high latency caused by connection establishment and error correction. The Barix implementation employs UDP as a transmission protocol.

	In multimedia applications low latency is essential, whereas ~~certain~~ packet loss can be tolerated.		In multimedia applications low latency is essential, whereas occasional packet loss can be tolerated. A packet typically carries a fraction of a second worth of audio, which makes the lost packet almost inaudible. RTP adds a sequence number and a time stamp to each packet to allow detection and recovery from network errors.

	===Payload Type===		===Payload Type===

Brain: /* RTP Protocol */

2009-03-27T10:25:07Z

RTP Protocol

← Older revision		Revision as of 10:25, 27 March 2009
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	=== RTP Protocol===		=== RTP Protocol===

	RTP (Real-time Transport Protocol) has been designed for ~~the~~ transport of multimedia streams ~~over~~ IP networks. Implementation is based on the User Datagram Protocol (UDP). ~~As IP networks are asynchronous~~, ~~packets may~~ be ~~lost or delivered in different order. RTP provides no guarantee of the delivery, but sequencing of the data makes it possible to detect missing packets~~.		[[wikipedia:Real-time Transport Protocol\|RTP (Real-time Transport Protocol)]], as a standard application protocol of the [[wikipedia:Internet Protocol Suite\|Internet Protocol Suite]], has been designed for transport of multimedia streams in IP networks.

			Implementation is widely based on the [[wikipedia:User_Datagram_Protocol\|User Datagram Protocol (UDP)]]. The [[wikipedia:Transmission Control Protocol\|Transmission Control Protocol (TCP)]] is not often used by RTP for its high latency caused by connection establishment and error correction. The Barix implementation employs UDP as a transmission protocol.

			In multimedia applications low latency is essential, whereas certain packet loss can be tolerated.

	===Payload Type===		===Payload Type===

Domenico: /* RTP */

2009-03-26T18:14:49Z

RTP

← Older revision		Revision as of 18:14, 26 March 2009
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	=== RTP ===		=== RTP Protocol===

			RTP (Real-time Transport Protocol) has been designed for the transport of multimedia streams over IP networks. Implementation is based on the User Datagram Protocol (UDP). As IP networks are asynchronous, packets may be lost or delivered in different order. RTP provides no guarantee of the delivery, but sequencing of the data makes it possible to detect missing packets.

	~~RTP (Real-time Transport Protocol) has been designed for~~ the transport of ~~multimedia streams over IP networks~~. ~~It works on top of UDP.~~		===Payload Type===
	~~It was developed by the Audio-Video Transport Working Group~~ of the ~~IETF [http://en.wikipedia.org/wiki/IETF] and first published in 1996 as RFC 1889, and superseded by RFC 3550 in 2003.~~		describes the transport of a particular media format. Some values of the audio payload formats are standard others need to be described in a service description protocol, SDP (Session DescriptionProtocol);

	~~IP networks are asynchronous and may lose or reorder packets.~~

	~~Some fields in RTP~~ are ~~defined~~ to ~~address these problems:~~

	~~'''Payload Type''' tags the content of the packet. Some standardized values exist, but for other codecs, the type must~~ be described in a service description protocol, SDP (Session DescriptionProtocol);

	The following table shows the defined RTP payload types used in Barix firmware.		The following table shows the defined RTP payload types used in Barix firmware.
	Note that Payload types 0, 8, 10, 11 and 14 are defined by the RTP standard while types 96 to 112 (dynamic payload types) are Barix specific.		Note that Payload types 0, 8, 10, 11 and 14 are defined by the RTP standard while types 96 to 112 (dynamic payload types) are Barix specific.
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	~~'''~~Sequence Number~~'''~~ is primarily used to identify and detect lost packets and secondly to reconstruct the order in which packets where sent, which may make loss detection easier;		===Sequence Number===
			is primarily used to identify and detect lost packets and secondly to reconstruct the order in which packets where sent, which may make loss detection easier;
	~~'''Time Stamp''' is the sampling instant for the first octet of media data in a packet. It can be used to help recover the clock frequency at the receiving side, if it is not given by other means.~~

	~~RTP does not include~~ a ~~loss recovery mechanism in case of~~ packet ~~losses~~.		===Time Stamp===
			is the sampling instant for the first octet of media data in a packet. It can be used to help recover the clock frequency at the receiving side, if it is not given by other means.

	Check the Wikipedia for an introduction to the ~~topic~~ [http://en.wikipedia.org/wiki/Real-time_Transport_Protocol].		===References===
			Check the Wikipedia for an introduction to the RTP protocol [http://en.wikipedia.org/wiki/Real-time_Transport_Protocol].

Domenico: /* RTP */

2009-03-26T17:55:05Z

RTP

← Older revision		Revision as of 17:55, 26 March 2009
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	'''~~sequence~~ Number''' is primarily used to identify and detect lost packets and secondly to reconstruct the order in which packets where sent, which may make loss detection easier;		'''Sequence Number''' is primarily used to identify and detect lost packets and secondly to reconstruct the order in which packets where sent, which may make loss detection easier;

	'''Time Stamp''' is the sampling instant for the first octet of media data in a packet. It can be used to help recover the clock frequency at the receiving side, if it is not given by other means.		'''Time Stamp''' is the sampling instant for the first octet of media data in a packet. It can be used to help recover the clock frequency at the receiving side, if it is not given by other means.

Domenico: /* RTP */

2009-03-26T17:54:33Z

RTP

← Older revision		Revision as of 17:54, 26 March 2009
Line 9:		Line 9:
	Some fields in RTP are defined to address these problems:		Some fields in RTP are defined to address these problems:

	~~- PT (~~Payload Type) tags the content of the packet. Some standardized values exist, but for other codecs, the type must be described in a service description protocol, SDP (Session DescriptionProtocol);		'''Payload Type''' tags the content of the packet. Some standardized values exist, but for other codecs, the type must be described in a service description protocol, SDP (Session DescriptionProtocol);

	The following table shows the defined RTP payload types used in Barix firmware.		The following table shows the defined RTP payload types used in Barix firmware.
Line 114:		Line 114:


	- sequence ~~number~~ is primarily used to identify and detect lost packets and secondly to reconstruct the order in which packets where sent, which may make loss detection easier;		'''sequence Number''' is primarily used to identify and detect lost packets and secondly to reconstruct the order in which packets where sent, which may make loss detection easier;

	~~- the timestamp~~ is the sampling instant for the first octet of media data in a packet. It can be used to help recover the clock frequency at the receiving side, if it is not given by other means.		'''Time Stamp''' is the sampling instant for the first octet of media data in a packet. It can be used to help recover the clock frequency at the receiving side, if it is not given by other means.

	RTP does not include a loss recovery mechanism in case of packet losses.		RTP does not include a loss recovery mechanism in case of packet losses.

	Check the Wikipedia for an introduction to the topic [http://en.wikipedia.org/wiki/Real-time_Transport_Protocol].		Check the Wikipedia for an introduction to the topic [http://en.wikipedia.org/wiki/Real-time_Transport_Protocol].