I have implemented basic WEBRTC peer connection between two clients and uses RTCPeerConnection and getUserMedia APIs for it. I am fetching Audio tracks only from the stream.
Even though my signalling works, ice agents are shared, streams are shared as well.
The audio comes out distorted completely.
However, if I utilize a VPN on one device the other device can hear the audio coming from the device with VPN very clearly with no hiccups.
NOTE:
I tried this with only stun and with a combination of stun and turn servers(viagenie).
Can someone tell me how do i debug the issue, what to look for?
Is my ISP involved in messing around with the media stream? (I live in the UAE.)
Thanks in advance!
i think the problem is UAE ISP block the STUN/TURN PORT (3478,5766 ...)
you can try to use 443/80 port?
From the open source code of AbstractKaaClient.java below, i'm sure kaaproject supports TCP protocol between endpoint and kaa server.
KaaDataChannel operationsChannel = new DefaultOperationTcpChannel(
kaaClientState, failoverManager, failureListener);
But there are some cases, especially in IoT domain, UDP is more popular than TCP, for its bigger transport overheads. Does kaaproject.org have any plan to support udp transport ? If it has already supported, could you please give me a solution?
Thanks!
It is not supported in 0.x versions of KAA platform. But in 1.0 version we planned to add support for CoAP that built over UDP.
I'm trying to stream a video via rtp with vlc, the default protocol is UDP, but its performance is very low (bitrate, resolution, smooth,..)
So, I want to use vlc with rtp to stream video over tcp protocol to gain higher performance.
Are there anyone can help me to force vlc streaming rtp over tcp protocol.
Please help me if you have any idea.
Thank you so much!
If you're using the command-line client then add the --rtsp-tcp options:
vlc --rtsp-tcp rtsp://path/to/stream
This article may be useful:
https://web.archive.org/web/20151117041313/http://www.wowza.com/forums/content.php?64-How-to-configure-VLC-media-player-for-RTSP-RTP-playback-(RTSP-RTP-interleaved-and-tuning)
This article explains how to configure the VideoLAN VLC media player to stream live or on demand streams through RTSP/RTP over TCP (RTSP/RTP interleaved). While HTTP streaming is a newer standard offering more features, RTSP/RTP streaming may work better when streaming through a firewall/router, which doesn't have UDP streaming open.
Could i inject packets to Linux TCP stack without modifying the ethernet driver? Could i do this with using a library or sth ?
Thank you,
If by 'inject packets to Linux TCP stack' you mean send some data that the Linux kernel will treat as a frame coming from an Ethernet interface then you can use a 'tap' device. If an IP packet (layer 3) is good enough, then use a 'tun' device.
http://en.wikipedia.org/wiki/TUN/TAP
http://www.kernel.org/pub/linux/kernel/people/marcelo/linux-2.4/Documentation/networking/tuntap.txt
Libnet
Libnet is a generic networking API that provides access to several protocols. It is not designed as a 'all in one' solution to networking. Currently many features that are common in some network protocols are not available with Libnet, such as streaming via TCP/IP. We feel that Libnet should not provide specific features that are possible in other protocols. If we restrict Libnet to the minimal needed to communicate (datagram/packets) then this allows it to support more interfaces.
Otherwise, if you're just wondering about injecting hand-crafted packets into the network, read the man pages and look for online help with raw sockets. Some good places to start are man 7 raw, man packet, and there are some ok tutorials at security-freak.net, though the code there is not written particularly well for my tastes.
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.