Real-Time Transport Protocol (RTP) is an Internet Protocol standard that specifies the way programs manage the real-time transmission of multimedia data over unicast or multicast network services.
In comparison to TCP (Transmission Control Protocol) which favors data integrity rather than delivery speed, RTP favors rapid delivery and has mechanisms to compensate for any minor loss of data integrity.
RTP defines the standardized packet format for delivering audio and video over IP networks and used in conjunction with Real-Time Transport Control Protocol (RTCP) to ensure that multiple streams of media can be synchronized and Quality of Service (QoS) can be maintained.
In distributed computing, losing information can be catastrophic but in media streaming, packet losses can be catered for through clever algorithms that patch undelivered/late packet data in real time. Up to a point - there are obviously Quality of Service limits that are deemed acceptable/unacceptable. RTP allows the opportunity to apply frame padding to prevent viusal stuttering and to apply algorithms that patch audio dropouts and prevent clicks or extraneous digital noise.
The most important recent application of RTP is the introduction of VoIP (Voice over Internet Protocol) systems which are becoming very popular as alternatives to regular telephony circuits.
Real-Time Transport Protocol is used extensively in entertainment and communication systems that involve streaming video such as video teleconference applications and Voice over Internet Protocol.
RTP is used in conjunction with Real-Time Transport Control Protocol (RTCP), which allows monitoring of transmission statistics and Quality of Service (QoS) assessment.
When both protocols are engaged, even-numbered ports are assigned to RTP while alternately, odd numbered ports are assigned to RTCP. This gives them discreet communications ports through which their data can be exchanged, so neither is dependent upon the delivery timing of the packet streams of the other but are delivered in strict alternating sequence so that their timing is very close.
The layering of ports in an alternating send/receive stack strives to ensure timing stability between the 2 streams. The RTP traffic is typically 95% of the total with RTCP only 5% or less of total traffic; this ratio ensures that the synchronization signal gets priority due to its small size and goes some way in keeping audio and video streams in time with each other.
RTP compensates for jitter and detects of out-of-sequence data arrival, both of which are common during IP network transmission.
In comparison to TCP (Transmission Control Protocol) which favors data integrity rather than delivery speed, RTP favors rapid delivery and has mechanisms to compensate for any minor loss of data integrity.
RTP defines the standardized packet format for delivering audio and video over IP networks and used in conjunction with Real-Time Transport Control Protocol (RTCP) to ensure that multiple streams of media can be synchronized and Quality of Service (QoS) can be maintained.
In distributed computing, losing information can be catastrophic but in media streaming, packet losses can be catered for through clever algorithms that patch undelivered/late packet data in real time. Up to a point - there are obviously Quality of Service limits that are deemed acceptable/unacceptable. RTP allows the opportunity to apply frame padding to prevent viusal stuttering and to apply algorithms that patch audio dropouts and prevent clicks or extraneous digital noise.
The most important recent application of RTP is the introduction of VoIP (Voice over Internet Protocol) systems which are becoming very popular as alternatives to regular telephony circuits.
Real-Time Transport Protocol is used extensively in entertainment and communication systems that involve streaming video such as video teleconference applications and Voice over Internet Protocol.
RTP is used in conjunction with Real-Time Transport Control Protocol (RTCP), which allows monitoring of transmission statistics and Quality of Service (QoS) assessment.
When both protocols are engaged, even-numbered ports are assigned to RTP while alternately, odd numbered ports are assigned to RTCP. This gives them discreet communications ports through which their data can be exchanged, so neither is dependent upon the delivery timing of the packet streams of the other but are delivered in strict alternating sequence so that their timing is very close.
The layering of ports in an alternating send/receive stack strives to ensure timing stability between the 2 streams. The RTP traffic is typically 95% of the total with RTCP only 5% or less of total traffic; this ratio ensures that the synchronization signal gets priority due to its small size and goes some way in keeping audio and video streams in time with each other.
RTP compensates for jitter and detects of out-of-sequence data arrival, both of which are common during IP network transmission.
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