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Real Time Messaging Protocol (RTMP) was initially a proprietary protocol developed by Macromedia for streaming audio, video and data over the Internet, between a Flash player and a server. Macromedia is now owned by Adobe, which has released an incomplete version of the specification of the protocol for public use. The RTMP protocol has multiple variations: # The "plain" protocol which works on top of and uses TCP port number 1935 by default. # RTMPS, which is RTMP over a TLS/SSL connection. # RTMPE, which is RTMP encrypted using Adobe's own security mechanism. While the details of the implementation are proprietary, the mechanism uses industry standard cryptography primitives. # RTMPT, which is encapsulated within HTTP requests to traverse firewalls. RTMPT is frequently found utilizing cleartext requests on TCP ports 80 and 443 to bypass most corporate traffic filtering. The encapsulated session may carry plain RTMP, RTMPS, or RTMPE packets within. While the primary motivation for RTMP was to be a protocol for playing Flash video, it is also used in some other applications, such as the Adobe LiveCycle Data Services ES. == Basic operation == RTMP is a TCP-based protocol which maintains persistent connections and allows low-latency communication. To deliver streams smoothly and transmit as much information as possible, it splits streams into fragments, and their size is negotiated dynamically between the client and server. Sometimes, it is kept unchanged; the default fragment sizes are 64 bytes for audio data, and 128 bytes for video data and most other data types. Fragments from different streams may then be interleaved, and multiplexed over a single connection. With longer data chunks, the protocol thus carries only a one-byte header per fragment, so incurring very little overhead. However, in practice, individual fragments are not typically interleaved. Instead, the interleaving and multiplexing is done at the packet level, with RTMP packets across several different active channels being interleaved in such a way as to ensure that each channel meets its bandwidth, latency, and other quality-of-service requirements. Packets interleaved in this fashion are treated as indivisible, and are not interleaved on the fragment level. The RTMP defines several virtual channels on which packets may be sent and received, and which operate independently of each other. For example, there is a channel for handling RPC requests and responses, a channel for video stream data, a channel for audio stream data, a channel for out-of-band control messages (fragment size negotiation, etc.), and so on. During a typical RTMP session, several channels may be active simultaneously at any given time. When RTMP data is encoded, a packet header is generated. The packet header specifies, amongst other matters, the ID of the channel on which it is to be sent, a timestamp of when it was generated (if necessary), and the size of the packet's payload. This header is then followed by the actual payload content of the packet, which is fragmented according to the currently agreed-upon fragment size before it is sent over the connection. The packet header itself is never fragmented, and its size does not count towards the data in the packet's first fragment. In other words, only the actual packet payload (the media data) is subject to fragmentation. At a higher level, the RTMP encapsulates MP3 or AAC audio and FLV1 video multimedia streams, and can make remote procedure calls (RPCs) using the Action Message Format. Any RPC services required are made asynchronously, using a single client/server request/response model, such that real-time communication is not required. 抄文引用元・出典: フリー百科事典『 ウィキペディア（Wikipedia）』 ■ウィキペディアで「Real Time Messaging Protocol」の詳細全文を読む スポンサード リンク
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