RTMP being an insecure unencrypted protocol, suffers from attacks like MITM and network sniffing. Naturally so, FB uses RTMPS: https://ppc.land/streaming-facebook-moves-live-videos-completely-to-rtmps/
Why is Google lackluster about adopting this secure approach? Is it due to less compatibility for the RTMPS protocol with streaming software and providers?
Azure does not provide RTMPS, at least their official developer doc does not mention that.
RTMPS was never an official standard. Adobe has a custom encryption called RTMPE, but that was never really supported outside Adobe software.
Because RTMPS is not official most high end video equipment and a lot of server software does not support it.
Related
I am wondering what the benefit of a WebRTC based P2P system would be?
Is the performance improved?
Is the security stronger?
... as opposed to a traditional vanilla P2P solution in, fx, Java.
The benefit of WebRTC/RTCWeb (first is the implementation and w3c API, second is the ietf standard) is in its name: Web. It does not have anything special, it's just an standard inspired by SIP whose implementation is shared by all the main browsers. The benefit is that you don't have to code a client for it because it's already embedded in the browser next to GetUserMedia which enables you to easily capture video and audio streams from the devices.
Performance is determined mostly by the codecs and they use open source codecs you could use in your own implementation. It is also secure, but you can make any p2p connection secure through encryption.
It is just an standard meant to make web developers' lifes easier when dealing with p2p real time voice, video and data streaming.
I noticed that JavaCard 3.0 may have the ability to use HTTPS from the Oracle website (oracle.com/technetwork/articles/javase/javacard3-142122.html).
Are there any ways to create HTTPS connections to a normal Internet website ?
Basically with Java Card Classic you are limited to the APDU interface. This interface has been specified in the Java Card API and the ISO/IEC 7816-4 standard.
It is of course possible to channel any kind of protocol through an APDU interface, but you would have to program it yourself. Furthermore, you would have to do so on the terminal side as well, because Java doesn't know anything about TCP/IP, name resolution etc. As Java Card environments are very limited, it would be tricky to create something that resembles an HTTP client.
There have been demonstrations that implemented a tiny web server on a Java Card. Those obviously also require some kind of proxy on the terminal side.
The Connected Edition - if you can find it anywhere - uses the same idea; it implements a web-server for e.g. authentication. It doesn't provide a client to my knowledge.
A1: There are no JavaCard Connected (which describes such option) devices publicly available.
A2: Classic JavaCard does not specify/allow any kind of connections.
I am working on a webRTC application and would like to be able to support multiple calls and be able to call from the browser to legacy VoIP or Videoconferencing systems as well as browser to browser.
now that Asterisk has added websocket in their latest builds would you need SIP and a SIP proxy in order to communicate with VoIP systems or will Asterisk allow this?
now that H.264 has been open sourced by Cisco would you still need a transcoder in order to call a legacy VTC system?
Is Node.js the preferred technology for implementing webrtc client/server deployments? I've looked into Mobicents SIP Servlets a bit but that seems to be the only alternative technology available beside a node.js solution.
If needed I am planning on creating a SIP trunk between an Asterisk server and our Polycom VBP so the webrtc clients should be able to get presence information through that connection so if no media transcoding is required with the recent changes then media should be able to pass directly from polycom endpoint to browser with the asterisk handling the signalling.
Thank you anyone who is able to answer any of these questions, it is still early in the r&d portion of this project for me and i'd like to get as much information as possible.
also: i did see SIP over websockets to true SIP. I understand that "something" needs to stand in between the webRTC client and the VoIP phone or Legacy SIP endpoint. what I would like to know is if that can be just asterisk with the recent update. if asterisk is all that is required, is there a way to include a media transcoder like red5? I haven't seen anything in the webrtc API that would allow you to include a transcoder, asterisk has transcoding mods but none that will do vp8 to h.26x or Opus to anything as far as i know.
Answer on that question higly depend of destination "legacy" system. Cisco "legacy" systems use h323 and sip, which is not compatible with webRTC.
Sure there are alot of ways to setup asterisk, red5, opensips or other as translation level.
Webrtc goal is call from browser. It never supposed have any API for transcode. That have be done by server part(which require special knowledge and experience to be propertly setup)
There are alot of availible documentation in internet, no any way put answer in less then 30 pages of text.
I study the construction of mobile networks and began to study MVAS. But could not find a specific iinformation what protocols are used in the VAS or MVAS.
I understood that main protocol using SMS - it SMPP.
It would be great if someone made a list of the protocols used, or links where I could read more information about the protocols used.
There is such a list; it is published by 3GPP in specification TS 23.039.
3GPP (earlier ETSI) specified the GSM, UMTS and LTE systems, with standard protocols for most of the interfaces. They did not specify any standard protocol between Short Message Service Centres and external messaging servers though.
Instead, this was left open, and each SMSC developer specified their own protocol. An early and successful SMSC developer was an Irish company called Aldiscon, which was later taken over by Logica. They developed the Short Message Peer-to-Peer protocol (SMPP), and published it as an open standard, which is the reason why it's so widely used today.
I need to implement a one-to-one videoconferencing solution server-based, runnable by browser, free (or not expensive), ssl support and good quality video and audio. What would you advise me?
See WebRTC. Chrome and Firefox will be supporting it (early support is in Chrome now, soon will be in Firefox). It appears Microsoft will be supporting it too - they're hiring engineers specifically for WebRTC work.