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    Real time transport protocol (RTP)

     

    1. Introduction to RTP


    RTP is a real-time transmission protocol which provides end-to-end transmission service, which supports the transmission of real-time data in single target broadcast and multi-objective broadcast network service, while the real-time data transmission is monitored and controlled by RTCP protocol.


    2. RTP is defined in RFC


    The application using RTP protocol runs on RTP, while the program executing RTP runs on the upper layer of UDP, in order to use the port number and check and of UDP. RTP can be regarded as a sub layer of the transport layer. The audio and TV data blocks generated by multimedia applications are encapsulated in RTP packets, each RTP packet is encapsulated in UDP message segment, and then packaged in IP packets.


    The structure of the packet includes several domains widely used in multimedia, including audio on demand, video on demand, Internet telephone and videoconference. RTP specification does not set standards for compressed formats for sound and television, and it can be used to transmit files in normal format. For example, sound in wav or GSM (Global System for mobile communications), MPEG-1 and MPEG-2 TV can also be used to transmit sound and TV files stored in proprietary formats.


    From the perspective of application developers, RTP executors can be considered as part of the application because developers must integrate RTP into the application. At the sending end, developers must write the program executing RTP protocol into the application program that creates RTP information package, and then the application program sends RTP information package to socket interface of UDP, as shown in Figure 2; Similarly, RTP packets are input to the application through UDP socket interface at the receiver. Therefore, developers must write the program executing RTP protocol to the application that extracts media data from RTP packet.


    The paper takes RTP as an example to illustrate its working process. Suppose the sound of the sound source is a PCM encoded sound of 64 kb / s, and assume that the application takes 20 ms of encoded data as a chunk, that is, 160 bytes of sound data in a data block. The application needs to add RTP title to this sound data to generate RTP packets, which include the type, sequence number and timestamp of the sound data. RTP packets are then sent to the UDP socket interface, where they are encapsulated in the UDP packets. At the receiver, the application program receives RTP information package from the socket interface, extracts the sound data block from RTP information package, and then decodes and plays sound correctly using the information in the title field of RTP packet.


    If the application does not use proprietary solutions to provide payload type, sequence number or timestamp, but uses standard RTP protocol, the application will be easier to run with other network applications, which is what everyone hopes. For example, if two different companies are developing Internet phone software, they all incorporate RTP into their products, which is hopeful that users using different company phone software can communicate.


    It is important to emphasize that RTP does not provide any mechanism to ensure that data is delivered to the receiver in time or other service quality. It does not guarantee that the information package is not lost or the order of packets is not disturbed. Indeed, RTP encapsulation can only be seen on the system side. The router in the middle does not distinguish that IP datagram carries RTP packets.


    RTP allows each media source to be assigned a separate RTP packet stream, such as a camera or microphone. For example, a television conference with two groups involved could open four packet streams: two cameras transmitting TV streams and two microphones to transmit sound streams. However, many popular coding technologies, including MPEG-1 and MPEG-2, bind sound and TV images together to form a single data stream in the coding process, and generate a RTP packet stream in one direction.
    RTP packets are not limited to single target broadcasting, and they can also be transmitted on one to many multi-target broadcast tree or multi-to-many multi-target broadcast tree. For example, multi-target broadcast with multiple to many, in this application, all the transmitting terminals usually send their RTP packet stream to the multi-objective broadcast tree with the same multi-objective broadcast address.


    3. RTP packet header field


    RTP title consists of four packet header fields and other domains: payload type domain, sequence number domain, timestamp domain and synchronization source identifier domain.

     

    1) payload type
    The payload type field in RTP packet is 7 bits long, so RTP can support 128 different payload types. For sound flow, this field is used to indicate the type of coding used by sound, such as PCM, adaptive delta modulation, linear predictive coding, and so on. If the sender decides to change the encoding method during the session or broadcast, the sender can notify the receiver through this domain. Table 1 lists the types of sound payloads that RTP can support at present.


    For TV streams, payload types can be used to indicate the type of TV coding, such as motion JPEG, MPEG-1, MPEG-2, h.231, etc. The sender can also change the encoding method of TV at any time during the session or during the session. Table 16-02 lists some TV payloads types that RTP can support at present.


    2) serial number
    The sequence number field field is 16 bits long. Add 1 to each RTP packet sequence number. The receiver can use it to check whether the packet is missing and process the packet according to the sequence number. For example, the receiving application receives a RTP packet stream, which has an interval between sequence numbers 86 and 89, and the receiver knows that packets 87 and 88 have been lost and take measures to process the lost data.

     

    3) timestamp
    The timestamp domain is 32 bytes long. It reflects the sampling time (time) of the first byte in the RTP packet. The receiver can use this timestamp to remove the jitter of the packets caused by the network, and provide synchronization function for playback at the receiving end.


    4) synchronization source identifier
    The length of the synchronization source identifier (SSRC) domain is 32 bits. It is used to identify the origin of RTP packet flow, and each packet flow during RTP session or period has a clear SSRC. SSRC is not the IP address of the sender, but a number randomly assigned by the source at the beginning of the new packet flow.

     

     

     

     

     

     

    How far(long) the transmitter cover?

    The transmission range depends on many factors. The true distance is based on the antenna installing height , antenna gain, using environment like building and other obstructions , sensitivity of the receiver, antenna of the receiver . Installing antenna more high and using in the countryside , the distance will much more far.

    EXAMPLE 5W FM Transmitter use in the city and hometown:

    I have a USA customer use 5W fm transmitter with GP antenna in his hometown ,and he test it with a car, it cover 10km(6.21mile).

    I test the 5W fm transmitter with GP antenna in my hometown ,it cover about 2km(1.24mile).

    I test the 5W fm transmitter with GP antenna in Guangzhou city ,it cover about only 300meter(984ft).

    Below are the approximate range of different power FM Transmitters. ( The range is diameter )

    0.1W ~ 5W FM Transmitter :100M ~1KM

    5W ~15W FM Ttransmitter : 1KM ~ 3KM

    15W ~ 80W FM Transmitter : 3KM ~10KM

    80W ~500W FM Transmitter : 10KM ~30KM

    500W ~1000W FM Transmitter : 30KM ~ 50KM

    1KW ~ 2KW FM Transmitter : 50KM ~100KM

    2KW ~5KW FM Transmitter : 100KM ~150KM

    5KW ~10KW FM Transmitter : 150KM ~200KM

    How to contact us for the transmitter?

    Call me +8618078869184 OR
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    1.How far you want to cover in diameter ?
    2.How tall of you tower ?
    3.Where are you from ?
    And we will give you more professional advice.

    About Us

    FMUSER.ORG is a system integration company focusing on RF wireless transmission / studio video audio equipment / streaming and data processing .We are providing everything from advice and consultancy through rack integration to installation, commissioning and training.
     
    We offer FM Transmitter, Analog TV Transmitter, Digital TV transmitter, VHF UHF Transmitter, Antennas, Coaxial Cable Connectors, STL, On Air Processing, Broadcast Products for the Studio, RF Signal Monitoring, RDS Encoders, Audio Processors and Remote Site Control Units, IPTV Products, Video / Audio Encoder / Decoder, designed to meet the needs of both large international broadcast networks and small private stations alike.
     
    Our solution has FM Radio Station / Analog TV Station / Digital TV Station / Audio Video Studio Equipment / Studio Transmitter Link / Transmitter Telemetry System / Hotel TV System / IPTV Live Broadcasting / Streaming Live Broadcast / Video Conference / CATV Broadcasting system.
     
    We are using advanced technology products for all the systems, because we know the high reliability and high performance are so important for the system and solution . At the same time we also have to make sure our products system with a very reasonable price.
     
    We have customers of public and commercial broadcasters, telecom operators and regulation authorities , and we also offer solution and products to many hundreds of smaller, local and community broadcasters .
     
    FMUSER.ORG has been exporting more than 15 years and have clients all over the world. With 13 years experience in this field ,we have a professional team to solve customer's all kinds of problems. We dedicated in supplying the extremely reasonable pricing of professional products & services.
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    We have modernization of the factory . You are welcome to visit our factory when you come to China.

    At present , there are already 1095 customers around the world visited our Guangzhou Tianhe office . If you come to China , you are welcome to visit us .

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    This is our participation in 2012 Global Sources Hong Kong Electronics Fair . Customers from all over the world finally have a chance to get together.

    Where is Fmuser ?

    You can search this numbers " 23.127460034623816,113.33224654197693 " in google map , then you can find our fmuser office .

    FMUSER Guangzhou office is in Tianhe District which is the center of the Canton . Very near to the Canton Fair , guangzhou railway station, xiaobei road and dashatou , only need 10 minutes if take TAXI . Welcome friends around the world to visit and negotiate .

    Contact: Sky Blue
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    If the package is more than 2kg,we will ship via EMS , DHL , UPS, Fedex fast express delivery,it will take about 7~15days to your hand.

    If the package more than 100kg , we will send via DHL or air freight. It will take about 3~7days to your hand.

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    * After ship, we will send you an E-mail and give you the tracking number.

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    Contact US--->>Return the item to us--->>Receive and send another replace .

    Name: Liu xiaoxia
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    FMUSER Wirless Transmit Video And Audio More Easier !

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