Is there any way with vb.net of accomblishing MMS audio streaming?
Also, i read somewhere else within SO, that MMS streaming is no more supported by Microsoft. Is that true? Just curious..
Here's what I dug from Googling. Wikipedia:Microsoft Media Server says:
Microsoft Media Server (MMS) is the
name of Microsoft's proprietary
network streaming protocol used to
transfer unicast data in Windows Media
Services (previously called NetShow
Services). MMS can be transported via
UDP or TCP. The MMS default port is
UDP/TCP 1755.
Microsoft deprecated MMS in favor of
RTSP (TCP/UDP port 554) in 2003 with
the release of the Windows Media
Services 9 Series, but continued to
support the MMS for some time in the
interest of backwards compatibility.
Support for the protocol was finally
dropped in Windows Media Services
2008.
The spec for Microsoft Media Server (MMS) Protocol is public since 2008 by Microsoft:
The client can send MMS Protocol
request messages to the server over
the TCP connection, requesting the
server to perform actions such as
starting and stopping the flow of
multimedia data. The multimedia data
is transferred either over the same
TCP connection or as a flow of UDP
packets.
w:VLC media player apparently supports MMS streaming, and also has API binding to various languages, including C#.
w:Real Time Streaming Protocol lists some server implementations that supports Real Time Streaming Protocol.
Related
I want to develop a VoIP app with Xamarin and Xmpp server.
So far the only things that I have found is the openfire and "jitsi meet" for the server side and matrix for the client side. But the matrix has nothing to do with voice streaming and is just for text messaging and "jitsi meet" doesn’t have any sdk for .net client side.
I also have found the red5pro but this has client sdks just for native android and ios development platform and has nothing for Mono.
So what Should I look for?!
First, let's clarify some basics:
openfire is a XMPP server. Basically, this is all you need on the server side for basic VoIP support.
Alternatives include ejabberd and Prosody.
jitsi meet essentialy already is a VoIP app, so if you want to develop your own, you don't really need that.
"Jitsi Videobridge" on the other hand can be used to provide a relay server for video conferences. For your first steps with a simple VoIP app, you wont need that either, but if you want your users to be able to create video conferences with many participants, then this helps.
(Explanation: Normally, when you create a P2P-Video conference, you
have two options: First, all users send their video data to all
participants (everybody needs lots of bandwidth), or you pick one
participant ("host") that receives the video streams of every
participant end sends them to every other participant. In the second
case, a normal participant only has to upload his stream once and
download n streams, whereas the host does most of the work - so only
that one user needs high bandwidth.
Jitsi Videobridge can run on a server and act as this conference host (usually a server has a much better bandwidth than a home user), so that none of the participants has to act as a host.
In simple VoIP applications (without video), this may not be neccessary, as audio streams are usually much smaller than video streams.)
Now, as I said above, in order to write a VoIP app, you basically only need a XMPP server (openfire, prosody and ejabberd should all be sufficient for this use case), a client library that supports Jingle and client libraries for the RTP media streams (transfer and display).
Jingle is the name of a XMPP protocol extension that enables the negotiation of P2P data streams as they are needed for a VoIP call.
The relevant protocol specifications:
XEP 0166: Jingle
XEP-0167: Jingle RTP Sessions
So what you need to find is a XMPP library with support for the jingle protocol. The C# Matrix XMPP SDK (not to be confused with the "Matrix protocol", which is a different protocol and has nothing to do with XMPP except for having a common goal) is one example of such a library. According to their web site, there is support for Jingle, but I couldn't find any documentation about it.
However, as I mentioned above, Jingle is only about how to negotiate data streams, not the data streams and VoIP itself.
So what that library probably helps you with is parsing of the Jingle XMPP messages that are needed to set up a RTP data stream.
For displaying and transfering the RTP stream, however, you need additional libraries. For that, have a look at the following SO questions and answers:
Open Source .net C# library for Real Time transport Protocol
Streaming Avi files from C# using RTP
I hope I could give you some useful hints...
can anyone help me what is the technical difference between WebRTC communication and the VoIP communication?
The question doesn't exactly make sense because it makes the assumption that VoIP is a technical stack, but it's not - it's a concept. The concept of sending Voice (V) over (o) Internet Protocols (IP). This means that different technology stacks can be used for accessing/capturing the media, establishing connections, negotiating streams, and transmitting streams.
WebRTC is one such stack (set of APIs, methods, and standards) for VoIP.
VOIP - Voice over Internet Protocol was a concept which came with popularity of internet. This involved using the internet to route voice telephony data, basically using existing IP infrastructure to transport audio streams without having dedicated circuit switched telephony. Over the time popular VOIP applications like Skype, Vonage and many in enterprise telephony came in.
VOIP had two parts one signalling , basically controller part and other actual media.
Actual media usually but not necessarily followed RTP (Real Time) protocol. RTP could carry both voice and video. Problem with RTP has been that browsers don't support it natively and it is not secure. You usually needed some sort of plugin to have VOIP work inside browser.
With WebRTC now popular browsers like FF, Chrome and Opera support a variation of RTP which is secure and can be natively invoked. Using WebRTC and browser Javascript you can send Voice, Video and Screen (it's video only) data to any other browser, which is cool.
VOIP : Voice over internet protocol uses DSL/Cable Modem voice over Wi-Fi/3G (VoWiFi/3G), voice over LTE (VoLTE), and Rich Communication Suite (RCS). VoIP is cloud-based, calls are sent as digital data and no cables are needed to send the call so any kind of Internet connection can be used to make calls and from a plethora of devices.
Web RTC: Web Real time communication use only OS browsers to communicate.
WebRTC requires the use of two main component JavaScript APIs.
WebRTC is an extension of VoIP to the browser world. It can reuse the existing VoIP infrastructure with incremental upgrades. This is good news for VoIP, as adoption of WebRTC only serves to increase overall VoIP proliferation.
Also, WebRTC is ideal for low-cost browser-based contact center applications. VoIP can serve embedded operator-driven VoLTE applications. Consequently, between WebRTC and VoIP, they can support wide range of consumer and enterprise applications.
I'm testing Audio/Video P2P connection between mobile devices.
Studying WebRTC, I've noticed NAT traversal(uses STUN server) and UDP-hole-punching is the key to make P2P possible.
On the other hand, I've noticed HLS(HttpLiveStreaming) on iOS devices is very optimized for A/V live streaming, and widely available even with Android4.x(3.x unstable)
So, here is my question if I use HLS for mobile P2P:
a) HLS is a protocol on TCP(HTTP) not UDP, so isn't there a performance drawback?
See: TCP vs UDP on video stream
b) How about NAT traversal? Will it be easier since HLS is HTTP(port:80)?
I have read wikipedia http://en.wikipedia.org/wiki/HTTP_Live_Streaming
Since its requests use only standard HTTP transactions, HTTP Live
Streaming is capable of traversing any firewall or proxy server that
lets through standard HTTP traffic, unlike UDP-based protocols such as
RTP. This also allows content to be delivered over widely available
CDNs.
c) How about android device compatibility? Is there a lot of problems to invoke StreamingLive distribution?
Thanks.
The reason why firewalls are not an issue for HLS is that it's a client-server protocol where all requests are done via HTTP on port 80. If you are implementing a P2P application, you won't be able to attach it to a port below 1024 unless you have root privileges.
This means that exchanging data via HLS (port 80) won't work for P2P. Unless you have a translation server in the middle, which defeats the purpose of P2P.
Comparing HTTP Live Streaming to P2P video streaming over UDP/RTP is almost like comparing apples and oranges. More like oranges and tangerines... read on.
HTTP Live Streaming was designed as client-server protocol without P2P or NAT traversal consideration. The idea being that the streaming server is already over HTTP/TCP and accessible from the public internet as if it was just an ordinary web server. The key features of HLS is its ability to dynamically switch the bitrate based on how well the client receives the stream. If the client connection to the server hiccups trying to stream down a 1080p video, it can transparently switch to sending a lower bitrate video (and likely switch back to streaming at higher bitrate if network conditions improve). Good example: Netflix.
WebRTC and ICE were designed to stream real time video bidirectionally between devices that might both behind NATs. As such, traversing a NAT through UDP is much easier than TCP. UDP lends itself to real-time (less latency) than TCP. Most video-chat clients (ala Skype) have dynamic bandwidth adjustments built in to their codecs and protocols to achieve something similar to what HLS does.
I suppose you could combine TCP NAT traveral and HLS together. Doing HLS over UDP implies that you build a TCP like reliability layer on top of your UDP stream.
Hope this helps
http://www.garymcgath.com/streamingprotocols.html
HTTP Live Streaming
The new trend in streaming is the use of HTTP with protocols that
support adaptive bitrates. This is theoretically a bad fit, as HTTP
with TCP/IP is designed for reliable delivery rather than keeping up a
steady flow, but with the prevalence of high-speed connections these
days it doesn't matter so much. Apple's entry is HTTP Live Streaming,
aka HLS or Cupertino streaming. It was developed by Apple for iOS and
isn't widely supported outside of Apple's products. Long Tail Video
provides a testing page to determine whether a browser supports HLS.
Its specification is available as an Internet Draft. The draft
contains proprietary material, and publishing derivative works is
prohibited.
The only playlist format allowed is M3U Extended (.m3u or .m3u8), but the format of the streams is restricted only by the implementation.
I could achieve P2P on top of HLS using WebRTC on a Android with a Mozilla Firefox browser and two others desktop browsers (Chrome and Firefox) on the same swarm.
Here's a screenshot of a presentation that I've made on the University: https://www.dropbox.com/s/zyfgs4o8al9ovd0/Screenshot%202014-07-17%2019.58.15.png
This screenshot was made by acessing http://bem.tv/demo.html.
If you want to know more about, this is my masters project and I'm publishing my advances on http://bem.tv and http://github.com/bemtv.
I am trying to develop a live video chat app using flash and rtmfp protocol but I have doubts as to
how rtmfp guarantees to connect peers especially when the peers are located in different networks.
RTMFP relies on a central rendezvous server to "introduce" 2 clients that are in different networks and potentially behind firewalls. To navigate through firewalls it uses NAT traversal techniques which essentially amounts to:
2 clients (Joe, and Mary) connect to central rendezvous server
Adobe's public RTMFP (non-commercial) rendezvous server rtmfp://p2p.rtmfp.net/
Or you can host your own using the GPL Open Sourced Cumulus or ArcusNode
Joe shares his peer ID with Mary or they can use a shared NetGroup ID
Mary communicates with the central rendezvous server which then uses a variety of NAT and Firewall traversal techniques to establish a peer 2 peer UDP connection
There is no guarantee that any 2 clients networks/firewalls are compatible for RTMFP P2P connections (you can diagnose with the RTMFP connection tester), that's why Adobe provides fallback solutions through their LiveCycle Collaboration Service or Adobe Flash Media Server to a central relay service (basically all network traffic would pass through a Server that both clients can publicly access) if a direct P2P connection can't be established.
Adobe's RTMFP is their proprietary bundling of Peer 2 Peer network rendevous service, as well as providing some other higher level P2P network mesh features.
If you're interested in a more open standard P2P protocol you should look into WebRTC. Essentially WebRTC is the same concept of allowing clients to connect P2P over UDP but geared towards being adopted as a web browser standard, and can also be implemented on a variety of native devices (e.g. Android, iPhone etc) under the hood it uses standard NAT and firewall traversal technology using STUN, ICE, TURN, RTP-over-TCP and support for proxies. I believe WebRTC is a standardization of some of the work done in the libjingle P2P Google Talk library.
I am planning a software application where the user will be able to select a given media channel from a list of RTMP streams available on one or more media servers on the internet. The list should ideally be dynamically created through some kind of service that knows about the available and active channels.
My question is: Would this be possible through some kind of protocol between the service and the media server. I understand that RTMP by itself doesn't allow this. A therefore assume that some outbound mechanism will be required.
No...
...there is no native application discovery in RTMP. If you'd like this kind of functionality you'll need to program some sort of discovery service for which ever streaming server you are running.