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Fmuser Sites
- es.fmuser.net
- it.fmuser.net
- fr.fmuser.net
- de.fmuser.net
- af.fmuser.net ->Afrikaans
- sq.fmuser.net ->Albanian
- ar.fmuser.net ->Arabic
- hy.fmuser.net ->Armenian
- az.fmuser.net ->Azerbaijani
- eu.fmuser.net ->Basque
- be.fmuser.net ->Belarusian
- bg.fmuser.net ->Bulgarian
- ca.fmuser.net ->Catalan
- zh-CN.fmuser.net ->Chinese (Simplified)
- zh-TW.fmuser.net ->Chinese (Traditional)
- hr.fmuser.net ->Croatian
- cs.fmuser.net ->Czech
- da.fmuser.net ->Danish
- nl.fmuser.net ->Dutch
- et.fmuser.net ->Estonian
- tl.fmuser.net ->Filipino
- fi.fmuser.net ->Finnish
- fr.fmuser.net ->French
- gl.fmuser.net ->Galician
- ka.fmuser.net ->Georgian
- de.fmuser.net ->German
- el.fmuser.net ->Greek
- ht.fmuser.net ->Haitian Creole
- iw.fmuser.net ->Hebrew
- hi.fmuser.net ->Hindi
- hu.fmuser.net ->Hungarian
- is.fmuser.net ->Icelandic
- id.fmuser.net ->Indonesian
- ga.fmuser.net ->Irish
- it.fmuser.net ->Italian
- ja.fmuser.net ->Japanese
- ko.fmuser.net ->Korean
- lv.fmuser.net ->Latvian
- lt.fmuser.net ->Lithuanian
- mk.fmuser.net ->Macedonian
- ms.fmuser.net ->Malay
- mt.fmuser.net ->Maltese
- no.fmuser.net ->Norwegian
- fa.fmuser.net ->Persian
- pl.fmuser.net ->Polish
- pt.fmuser.net ->Portuguese
- ro.fmuser.net ->Romanian
- ru.fmuser.net ->Russian
- sr.fmuser.net ->Serbian
- sk.fmuser.net ->Slovak
- sl.fmuser.net ->Slovenian
- es.fmuser.net ->Spanish
- sw.fmuser.net ->Swahili
- sv.fmuser.net ->Swedish
- th.fmuser.net ->Thai
- tr.fmuser.net ->Turkish
- uk.fmuser.net ->Ukrainian
- ur.fmuser.net ->Urdu
- vi.fmuser.net ->Vietnamese
- cy.fmuser.net ->Welsh
- yi.fmuser.net ->Yiddish
Why we choose SRT instead of RTMP?
Why choose SRT?
There is no doubt that RTMP is the largest live broadcast protocol in stock today, but with the continuous development of new technologies and the continuous expansion of usage scenarios, continuing to use RTMP will make people feel a little powerless. The defects of RTMP protocol mainly include the following four aspects:
● First of all, the RTMP protocol is too old, and the last update was in 2012; at the same time, there are no official definitions for video formats such as HEVC/H.265/AV1, so that domestic CDN manufacturers need to define them.
● The RTMP connection process is longer. Because RTMP is based on TCP (TCP has a three-way handshake), in addition, it also has a three-way handshake from c0/s0 to c2/s2, plus connection, createstream, play/publish, and so on. For the RTMP to complete a connection establishment, 9 sessions are required, which is barely acceptable for the PC side, and the requirements for the network quality of the mobile terminal are very high.
● The congestion control of RTMP completely relies on the transport layer, that is, it completely relies on the congestion control algorithm of the TCP transport layer for congestion management, with almost no optimization; RTMP itself is based on TCP transmission and cannot provide bandwidth adaptive algorithms.
● In this context, many manufacturers have begun to provide some new live broadcast protocols for industry reference. Such as QUIC, SRT, etc. This time we will focus on the characteristics and applications of SRT.
Haivision and Wowza jointly proposed the SRT protocol for real-time audio and video based on UDT. SRT is a UDT-based protocol (the UDT protocol is a UDP-based transmission protocol, and 4 versions have been submitted in the IETF). It has a very good packet loss retransmission mechanism. The control messages for packet loss retransmission are very rich. It also supports ACK, ACKACK, NACK.
We all know that audio and video are very concerned about time, and SRT's time-based message transmission enables it to have a good ability to prevent traffic bursts. SRT provides rich congestion control statistics to the upper layer, including RTT, packet loss rate, inflight, send/receive bitrate, etc. Using this rich information, we can realize bandwidth prediction, and do adaptive dynamic coding and congestion control at the coding layer according to the change of bandwidth.
