About-network-video:
video-solution:
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1 Introduction
With computer technology, network technology, video
compression coding technology such as the rapid development of key technologies,
network with its real-time video transmission technology and intuitive to wait
until more and more attention. However, constantly changing network conditions,
real-time transmission on the network audio / video streaming, must take
measures to prevent congestion, the quality of its services in order to
guarantee QoS.
Real-time Transport Protocol used to adjust the code RTP flow
to match the changes in network bandwidth, the use of feedback RTP protocol to
estimate the network status, and then based on the evaluation of the network
state and adjust the transmission rate of the sending end, in order to support
real-time transmission network services, provide real-time multimedia data
transmission standards, the achievement of self-adaptive network transmission.
2 Real-time Transport Protocol RTP
In fact by the RTP protocol
Real-time Transport Protocol RTP (Real-time Transport Protocol) and real-time
transmission control protocol RTCP (Real-time Transport Control Protocol) is
composed of two parts. RTP protocol is based on multicast or unicast network to
provide users with real-time continuous media data transmission services; RTCP
protocol is part of RTP control protocol for real-time monitoring of the quality
of data transmission, in order to provide congestion control and flow control.
RTP can carry on as a result of any agreement, so the application of RTP
protocol can not only confined to the field of multimedia transmission, but also
can be applied to other areas. Although the application of specific areas in the
RTP 'agreement only, based on the original field of application-specific further
limited, but so far, apart from the outside a number of research projects, RTP
protocol is mainly used in audio / video transmission.
3, video data
transmission
3.1 sub-media data strategy
Video data during network
transmission, we should first partition the size of data packets. This is
because the RTP protocol is usually based on UDP protocol, while the UDP
protocol on the size of the packet is provided for more than some may be lost,
for video data, there were some very key-frame compression, it is likely to
exceed the UDP protocol provides the maximum; In addition, the media keyframe
data and non-key frame is a big gap between the amount of data, and the router
is generally used to support the principle of priority packet may cause the
amount of data smaller than its non-critical frame in front of large amount of
data key-frame to reach the receiver earlier, which led to out-of-order. Media
data during transmission, therefore prior to the first sub.
For the
sub-media data is generally based on two conditions: (1) to determine the size
of data packets. Primarily on the basis of the provision of network bandwidth,
multi-media types and the transport protocol used to determine; (2) minimize the
data block that contains the complete information. Mainly on the encoding and
compression.
3.2 the process of sending packets
Send media data
compression process is to generate data streams (compressed encoder have a fixed
frequency in accordance with the data packets) sent to the media buffer, after
fault-tolerant after the media layer to the RTP data packets to re-group,
together with the RTP header, and then transmitted to the lower (as shown in
Figure 1).
Figure 1 the process of sending the media stream
Send the
process the media data the following steps:
(1) to determine whether a user
needs;
(2) if so, will be produced by compression Add network transmission
packet buffer, for fault-tolerant processing, the RTP layer;
(3) RTP media
layer data re-organization, together with the RTP header, RTP packets generated
to the lower deck;
(4) in accordance with the system according to the IP
address will be sent to the appropriate users.
3.3 to receive the course
packet
Since the system uses the UDP transport protocol is the protocol,
data packets during transmission, the possible out-of-order, it must be the
recipient of the measures adopted must receive the data. One possible approach
is to build a side in the receiving ring buffer queue for storing the data
received. The course of their work as shown in Figure 2, (a) for the initial
state, (b) to receive the first packet, (c) to receive two data packets, (d) To
take a reading packet.
Figure 2 buffer process
Two pointer queue buffer
pEmpty and pData, pEmpty instruction queue buffer ring section of the location
of an empty buffer, pData instruction queue buffer ring of the location of the
first data. In the communications before the queue in a sign of relief area for
"empty", pEmpty point to a buffer ring queue the first team, pData is empty.
After the communication, the system will receive packets pEmpty pointer into the
instruction buffer, the buffer is set to sign "non-empty," At the same time,
move the pointer to the next pEmpty an empty buffer location, media player
Remove a module for each packet, they instructed the pData buffer pointer to
re-sign for "empty", and under the pData pointer to the location of a data
buffer.
(1) receiver algorithm
Set buffer priority queue Q store has
been received but not yet dealt with the frame to give priority to bring the
number of frames of the serial number of the elements of Q-based collection for
the R, | R | = nQ, Q tolerance for the NQ (nQ <= NQ), located just removed
the serial number of the frame for the T0, when the serial number for TP the
arrival of the new P-frame will be as shown in Figure 3 algorithm to deal with:
Figure 3 process the buffer zone
Of which: (1) In order to ensure
real-time; (2) In order to ensure the transmission of non-repetitive; (3) select
the smallest number of discarded frames priority to make each element of the
queue to deal with ahead of time, This is to ensure that real-time.
(2)
algorithm for video data access
Data from the buffer zone to take steps to
check:
-identify the smallest number of buffer priority frames;
-will
receive the case of RTP packet monitor this QoS;
-In the RTP layer after
unpack, remove the RTP packet header;
-to the top player;
-Q in the
buffer zone to remove the frame.
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