I'm trying to stream a live video on a website and I already have a udp mpeg-ts. I cannot show this stream on html so I wanna convert this stream to http on server then send it to clients. how can I do that using ffmpeg?
any other solution accepted too.
thank you
The key question is - do you need a low latency live streaming or 15-30 seconds latency is OK for you. If you don't need low latency, use ffmpeg to ingest your udp mpeg-ts and output HLS.
For low latency live streaming to web browsers, you will have to install a media server software, such as Wowza / Unreal Media Server / Red5 or similar.
The media server will ingest your udp mpeg-ts and will convert it to WebRTC streams playable by web browsers.
I'm investigating RTMP, because I'm going to implement the option to broadcast a live stream from an Android device to an RTMP server. I found out that RTMP uses TCP by default and TCP guarantees delivery of packets, by retrying if they fail. Doesn't this make RTMP very unsuitable for broadcasting live streams? If the broadcaster's connection temporarily weakens, it will lead to packets that are not able to be sent in time. The stream will then fall further and further behind because of this, instead of just skipping the missed fragments.
Is this conclusion correct, or am I missing something here? I am aware btw of RMFP, which uses UDP instead of TCP. Is this what broadcasters use for live broadcasting of RTMP?
The client that is pushing the video has the option to drop a video/audio frame if it begin to fall behind.
I am trying to play the RTP playload in flex but no success. Can some enlighten me how to achieve this without using RTMP Server as middle ware.
You can't do that without using an RTMP server. The NetConnection class you find in Flex can send and receive RTMP streams, and those streams can have the same payload you find in RTP packets. Although, to unpack RTP packets and create RTMP packets you need an RTMP server like Wowza Media Server, or something alike.
There are several open source media servers you can use:
Red5
Wowza
RTMPD
Any of these would suit your purpose. Flex makes the client side pretty trivial too.
I need to send video stream between 2 mobile devices. Bada has no support for rtsp so i found its possible to tunnel it through http. can someone help me out with a sample on how to start of. Im new to bada application development.
bada has no RTSP per se, but it has sockets - Osp::Net::Sockets::Socket. I think your best bet is studying the description of RTSP and reimplementing it by hand over sockets. Since the bada sockets are not the same as POSIX sockets in terms of API (although I'm pretty sure bada sockets are a thin layer on top of POSIX ones), it's unlikely you'll find a ready made open source RTSP library.
This might be a silly question:
Does HTTP ever use the User Datagram Protocol?
For example:
If one is streaming MP3 or video using HTTP, does it internally use UDP for transport?
From RFC 2616:
HTTP communication usually takes place
over TCP/IP connections. The
default port is TCP 80, but other
ports can be used. This does not
preclude HTTP from being implemented
on top of any other protocol on the
Internet, or on other networks. HTTP
only presumes a reliable transport;
any protocol that provides such
guarantees can be used; the mapping
of the HTTP/1.1 request and response
structures onto the transport data
units of the protocol in question is
outside the scope of this
specification.
So although it doesn't explicitly say so, UDP is not used because it is not a "reliable transport".
EDIT - more recently, the QUIC protocol (which is more strictly a pseudo-transport or a session layer protocol) does use UDP for carrying HTTP/2.0 traffic and much of Google's traffic already uses this protocol. It's currently progressing towards standardisation as HTTP/3.
Typically, no.
Streaming is seldom used over HTTP itself, and HTTP is seldom run over UDP. See, however, RTP.
For something as your example (in the comment), you're not showing a protocol for the resource. If that protocol were to be HTTP, then I wouldn't call the access "streaming"; even if it in some sense of the word is since it's sending a (possibly large) resource serially over a network. Typically, the resource will be saved to local disk before being played back, so the network transfer is not what's usually meant by "streaming".
As commenters have pointed out, though, it's certainly possible to really stream over HTTP, and that's done by some.
Maybe just a bit of trivia, but UPnP will use HTTP formatted messages over UDP for device discovery.
Yes, HTTP, as an application protocol, can be transferred over UDP transport protocol.
Here are some of the services that use UDP and an underlying protocol for transferring HTTP data and streaming it to the end-user:
XMPP's Jingle Raw UDP Transport Method
A number for services that use UDT --- UDP-based Data Transfer Protocol, which is the a superset of UDP protocol.
The Transport Layer Security (TLS) protocol encapsulating HTTP as well as the above mentioned XMPP and other application protocols does have an implementation that uses UDP in its transport layer; this implementation is called Datagram Transport Layer Security (DTLS).
Push notifications in GNUTella are HTTP requests sent over UDP transport.
This article contains further details on streaming over UDP and its reliable superset, the RUDP: Reliable UDP (RUDP): The Next Big Streaming Protocol?
Of course, it doesn't necessarily have to be transmitted over TCP. I implemented HTTP on top of UDP, for use in the Satellite TV Broadcasting industry.
If you are streaming an mp3 or video that may not necessarily be over HTTP, in fact I'd be suprised if it was. It would probably be another protocol over TCP but I see no reason why you cannot stream over UDP.
If you do you have to take into account that there is no certainty that your data will arrive at the other end, but I can take it that you know about UDP.
To answer you question, No, HTTP does NOT use UDP.
For what you talk about though, mp3/video streaming COULD happen over UDP and in my opinion should never happen over HTTP.
Maybe some change on this topic with QUIC
QUIC (Quick UDP Internet Connections, pronounced quick) is an experimental transport layer network protocol developed by Google and implemented in 2013. QUIC supports a set of multiplexed connections between two endpoints over User Datagram Protocol (UDP), and was designed to provide security protection equivalent to TLS/SSL, along with reduced connection and transport latency, and bandwidth estimation in each direction to avoid congestion. QUIC's main goal is to optimize connection-oriented web applications currently using TCP.
I think some of the answers are missing an important point. The choice between UDP and TCP should not be based on the type of data (e.g., audio or video) or whether the application starts to play it before the transfer is completed ("streaming"), but whether it is real time. Real time data is (by definition) delay-sensitive, so it is often best sent over RTP/UDP (Real Time Protocol over UDP).
Delay is not an issue with stored data from a file, even if it's audio and/or video, so it is probably best sent over TCP so any packet losses can be corrected. The sender can read ahead and keep the network pipe full and the receiver can also use lots of playout buffering so it won't be interrupted by the occasional TCP retransmission or momentary network slowdown. The limiting case is where the entire recording is transferred before playback begins. This eliminates any risk of a playback stall, but is often impractical.
The problem with TCP for real-time data isn't retransmissions so much as excessive buffering as TCP tries to use the pipe as efficiently as possible without regard to latency. UDP preserves application packet boundaries and has no internal storage, so it does not introduce any latency.
(This is an old question, but it deserves an updated answer.)
In all likelihood, HTTP/3 will be using the QUIC protocol, which is described as
multiplexed transport over UDP
So, from a certain point of view, you could say that HTTP/3 will be using UDP.
The answer: Yes
Reason: See the OSI model.
Explaination:
HTTP is an application layer protocol, which could be encapsulated with a protocol that uses UDP, providing arguably faster reliable communication than TCP. The server daemon and client would obviously need to support this new protocol. Quake 2 protocol proves that UDP can be used over TCP to provide a basis for a structured communication system insuring flow control (e.g. chunk ids).
http over udp is used by some torrent tracker implementations (and supporteb by all main clients)
In theory yes it is possible to use UDP for http but that might be problematic. Say for instance in your example a mp3 or a video is being streamed there will be problem of ordering and some bits might go missing as UDP is not connection oriented there is no retransmit mechanism.
HTTP/3 (aka QUIC) uses UDP instead of TCP.
https://http3-explained.haxx.se/en/the-protocol/feature-udp
UDP is the best protocol for streaming, because it doesn't make demands for missing packages like TCP. And if it doesn't make demands, the flow is far more faster and without any buffering.
Even the stream delay is lesser than TCP. That is because TCP (as a far more secure protocol) makes demands for missing packages, overwriting the existing ones.
So TCP is a protocol too advanced to be used for streaming.