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Bind my number Maybe later. In typical applications, RTP is typically implemented on top of the transport protocol as part of the application. RTP by itself does not provide a reliable guarantee for sequential transmission of packets, nor does it provide traffic control and congestion control, which is done by RTCP. Typically, the RTCP uses the same distribution mechanism as RTP to periodically send control information to all members of the session, and the application receives the data, obtains information about the session participants, and the network condition, packet loss probability, etc.
The result is the ability to control the quality of the service or diagnose the condition of the network. SR: send-side report, the so-called sender is the application or terminal to emit RTP datagram, the sender can also be the receiving side. RR: The receiving side reports that the so-called receiver is the application or terminal that receives but does not send RTP datagrams. BYE: notification departure, the primary function is to indicate that one or several sources are no longer valid, that is, the other members in the notification session will exit the session themselves.
The RTCP datagram carries the necessary information of service quality monitoring, can dynamically adjust the service quality, and can control the network congestion effectively. Because the RTCP datagram is multicast, all members of the session can use the control information returned by the RTCP datagram to understand the current situation of the other participants.
In a typical application, the application that sends the media stream will periodically generate the send-side report SR, which contains synchronization information between different media streams, as well as the number of datagrams and bytes that have been sent, and the receiving end can estimate the actual data transfer rate based on this information.
On the other hand, the receiving end sends a receive-side report RR to all known senders, which contains important information such as the maximum serial number of the received datagram, the number of lost datagrams, the delay jitter, and the timestamp, which can be used by the sending application to estimate the round-trip delay.
The transmission rate can be dynamically adjusted according to the probability of datagram loss and delay jitter to improve the network congestion, or to smoothly adjust the service quality of the application according to the network condition. As an application layer protocol, RTSP provides an extensible framework, its significance is to make real-time streaming media data controlled and on-demand becomes possible.
In general, RTSP is a streaming media representation protocol that is primarily used to control data transmission with real-time characteristics, but it does not transmit data by itself, but must rely on some of the services provided by the underlying transport protocol. RTSP can provide streaming media such as play, pause, fast forward and other operations, it is responsible for defining the specific control message, operation method, status code, etc.
The set of media streams controlled by RTSP can be defined by the representation description Presentation Description , which refers to the collection of one or more media streams that the streaming media server provides to the client, while the description contains information about the various media streams in the presentation.
The RTSP protocol currently supports the following operations: Retrieving media: allows a user to submit a presentation description to the media server via HTTP or another method. If the representation is multicast, the description contains the multicast address and port number used for the media stream, and if it is unicast, the destination address should only be provided for security in the presentation description.
Add Media: notifies users of new and available media streams, which is especially useful for live lectures. If a traffic must be transmitted to multiple clients at the same time, the server must transmit a copy of the data stream to each client, TCP can dynamically adjust the transmission speed according to the network bandwidth and congestion level and resend the lost packets, so that the reliability of data transmission, but the server resources are expensive, It is difficult to guarantee the real-time data stream transmission in the case of large data stream.
The UDP protocol does not need to maintain the connection state, and does not think that each packet must reach the receiving end, so the network load is smaller than TCP, the transmission speed is faster than TCP, but the more congested the network, the more packets are lost. Shared objects are an important data type in rtmp data, and when any client changes the data, the shared object is able to update the server-side data in a timely manner, so that each client can be aware of the changes in the data.
RTMP supports more media protocols than traditional media servers. It supports the dynamic transmission of multiple lines that may contain sound, image and script data from the server to the customer and from the customer to the server. RTMP handles sound, image, and script data separately. Sound and video data are buffered separately from the server.
If the sound data reaches a certain limit in the sound buffer, all data in the buffer is discarded and the recently arrived data is allowed to start collecting in the buffer and being sent to each customer. The video data is processed in a similar manner, unlike when a new keyframe arrives and the data in the buffer is cleared.
When the old frame data is discarded, if the client's data is found to be incorrect, the old and new two different frames are fitted. Create a free Team What is Teams? Collectives on Stack Overflow. Learn more.
Ask Question. Asked 10 years, 11 months ago. Active 4 months ago. Viewed k times. Somebody correct me if I'm wrong, am I right?. Lobo Lobo 3, 7 7 gold badges 33 33 silver badges 63 63 bronze badges. Also, you may want to checkout pun unintended : Apple's open source Darwin Streaming Server to avoid reinventing the wheel - unless you have custom requirements not met by existing solutions.
JP19 Does it work on Windows? Add a comment. Active Oldest Votes. Read the documents I linked here, they are a good starting point. Krishna Oza 1, 1 1 gold badge 19 19 silver badges 46 46 bronze badges. Cipi Cipi I m trying to stream audio in J2Me apps and I m new for protocols.
You mentioned you can implement some sort of RTP-only server Does it mean that there are servers what can handle both rtp,rtcp? What are the servers rtp,rtcp we can use in our local machine for testing. Cipi , but must I use something smtg like rtp server,tools, programs,etc to make the computer stream anything? I mean if I want to stream, I get errors like this: stackoverflow. Cipi : Just curious to know whether we can actually pause a live stream and then hit play again.
Pretty much everything can be transported through TCP though. Some basics: RTSP server can be used for dead source as well as for live source. In your case if you want broadcasting streaming server then you need both RTSP for control as well as RTP broadcasting audio and video To start with you can go through sample code provided by live Community Bot 1 1 1 silver badge.
Alam Alam 1, 14 14 silver badges 25 25 bronze badges. I think thats correct. JP19 JP That is correct in most cases, there are cases where you can have something else replace Rtp in the protocol
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