i want to use a simple WebRTC for my clients like this one (https://appr.tc/).
The problem with the above link is that it can only take up to 2 users(A and B). Do you know if there are any others FREE programms that take clients A,B,C and D?
Thank you for your time.
Here is a simple example of multi-peer connection audio/video. With a DEMO. It uses simple-peer as wrapper of RTCPeerconnection.
https://github.com/Dirvann/webrtc-video-conference-simple-peer
The problem with this concept is that the peer connections use too much computing power.
Isn't there a possibility to get a peer connection to multiply peers.
Ps. I try to point out, there are a dict within a lot of peer connections.
Related
I have a Jitsi instance that I would like to share with 50 people for an event. I will mute everyone to make sure that only one or two people are talking at the same time but I wonder if I need to ask people to cut the video too (which would be sad because seeing other people face would be fun).
I know that Jitsi works in peer to peer when only two people are connected but what about a larger room?
Does the server centralise all the video stream?
With more than two participants Jitsi Videobridge will take over (no peer to peer nor H264) and control video streams from clients to server to clients.
As far as I understand it works very much like a turn server.
It strongly depends on your hardware setup. If Jitsi is running on a dedicated machine it should be able to handle this.
On a virtual server on the other hand...
I'm searching for 4 days, but can't get it. I built all libraries and integrated it in my custom project, but I don't know what steps should I do to make it work. The only thing that i found with code example\explanation is tech.appear.in/2015/05/25/Getting-started-with-WebRTC-on-iOS , but it is poor and unclear for me, AppRTCDemo source code too. I read about WebRTC for browsers but still can't reproduce it on iOS.
Can anybody explain or provide links to explanation on how to completely build iOS native app using WebRTC API for example p2p ios chat?
Besides the fact that I do not understand code logic provided in demo, I can't understand:
1) What is ICE servers for my iOS app? Should I take care of it? Is it something server side? Should I code and run it myself, or I can use existing Parse background?
2) What is signaling mechanism in iOS app? Is it client side only, or it must be implemented on server side too?
3) And maybe someone can explain step-by-step guide, maybe with some code, how to implement simple iOS p2p chat using WebRTC? For example:
"You have to:
Create ICE/STUN/TURN server on parse core using this =source= and this tutorial =tutorial=.
Create RTCPeerConnection using created ICEServer:
RTCPeerConnectionFactory *pcFactory = [[RTCPeerConnectionFactory alloc] init];
RTCPeerConnection *peerConnection = [pcFactory peerConnectionWithICEServers:kICEServerURL constraints:nil delegate:self];
Create DataChannel using ...
Send signal using ... explained here =link=
Set local and remote descriptions ...
Send Data ... using ...
... " or something similar.
I'm sorry for asking this, but I'm losing my mind trying to figure it out. Thank you!
I am not an expert in webrtc but i will try to explain some of your questions.
1.ICE servers-- NATs and firewalls impose significant problem in setting up IP endpoints. so IETF standards STUN, TURN and ICE were developed to address the NAT traversal problem.
STUN helps connect IP end-points:
discover whether they are behind a NAT/firewall, and if so,
to determine the public IP address and type of the firewall. STUN then uses this information to assist in establishing peer-to-peer IP connectivity.
TURN, which stands for Traversal Using Relay NAT, provides a fallback NAT traversal technique using a media relay server to facilitate media transport between end-points.
ICE is a framework that leverages both STUN and TURN to provide reliable IP set-up and media transport, through a SIP offer/answer model for end-points to exchange multiple candidate IP addresses and ports (such as private addresses and TURN server addresses).
2.Signaling is the process of coordinating communication. This signalling part needs to be implemented by you according to your needs(for ex. if you have sip structure in place then you will have to implement sip signalling). In order for a WebRTC application to set up a 'call', its clients need to exchange information:
Session control messages used to open or close communication.
Error messages.
Media metadata such as codecs and codec settings, bandwidth and media types.
Key data, used to establish secure connections.
Network data, such as a host's IP address and port as seen by the outside world.
Steps
for offerer:
first create the peer connection and pass the ice candidates into it
as parameters.
set event handlers for three events:
onicecandidate-- onicecandidate returns locally generated ICE candidates so you can pass them over other peer(s) i.e. list of ice candidates that are returned by STUN/TURN servers; these ice candidates contains your public ipv4/ipv6 addresses as well as UDP random addresses
onaddstream--onaddstream returns remote stream (microphone and camera of your friend!).
addStream` attaches your local microphone and camera for other peer.
Now create SDP offer by calling setLocalDescription function and set remote SDP by calling setRemoteDescription.
For Answerer:
setRemoteDescription
createAnswer
setLocalDescription
oniceCandidate--On getting locally generated ICE
addiceCandidate--On getting ICE sent by other peer
onaddstream--for remote stream to add
I hope this will make some of your doubts clear.
I came through the process of implementing it few month ago. What I've found was the library was not stable - sometimes it was working sometimes not.
Additionally my iPhone was always becoming hot when I was using it.
I would not suggest using this library and overall WebRTC technology for commercial projects.
This is my implementation, which was working few months ago:
https://github.com/aolszak/WebRTC-iOS
Good luck!
We are trying out an option of having multiple parallel thread meaning multiple consuming for different set of keywords by establishing single connection. Is it possible or should we establish new connection if we want to consume another set of data? Can anyone guide on this
Regards,
Balaji D
This is documented on DataSift's Multiple Streaming docs page. Some of the official API Client Libraries also support multi-streaming over WebSockets.
I am working on an MPI application, which uses threaded-MPI calls between processes. Threads are added and removed as per the load requirements. Now, I have a question, which I could not find answer in the open-mpi forum.
If a set of MPI processes ("ranks") already has a connection, ie, they are already making send-receive calls, and then a new thread comes in (either processes) which also makes the send-receive calls between the same MPI peers, would MPI open up new set of sockets?
I know that the details are implementation dependent, so there may not be a general answer. But, is there a way to find out?
There are questions on the scalability of this technique, which was chosen for other reasons. It would be great to get some stats, on the number of new sockets per connection.
Anyone knows how to do this? For instance, query which socket is a particular instance of MPI_Send writing to?
I already tried adding --mca btl self,sm,tcp --mca btl_base_verbose 30 -v -report-pid -display-map -report-bindings -leave-session-attached
Thanks a lot.
To answer my own question, here is what I learnt from brilliant folks at Open-MPI:
On Jan 24, 2012, at 5:34 PM, devendra rai wrote:
I am trying to find out how many separate connections are opened by MPI as messages are sent. Basically, I have threaded-MPI calls to a bunch of different MPI processes (who, in turn have threaded MPI calls).
The point is, with every thread added, are new ports opened (even if the sender-receiver pairs already have a connection between them)?
In Open MPI: no. The underlying connections are independent of how many threads you have.
Is there any way to find out? I went through MPI APIs, and the closest thing I found was related to cartographic information. This is not sufficient, since this only tells me the logical connections (or does it)?
MPI does not have a user-level concept of a connection. You send a message, a miracle occurs, and the message is received on the other side. MPI doesn't say anything about how it got there (e.g., it may have even been routed through some other process).
Reading Open MPI FAQ, I thought adding "--mca btl self,sm,tcp --mca btl_base_verbose 30 -display-map" to mpirun would help. But I am not getting what I need. Basically, I want to know how many ports each process is accessing (reading as well as writing).
For Open MPI's TCP implementation, it's basically one TCP socket per peer (plus a few other utility fd's). But TCP sockets are only opened lazily, meaning that we won't open the socket until you actually send to a peer.
--
Jeff Squyres
Credits to Jeff Squyres.
I need to use one logical PGM based multicast address in application while enable such application "seamlessly" running across several different geo-locations (i.e. think US/Europe/Australia).
Application is quite throughput (several million biz. messages a day) and latency demanding whith a lot of small but very frequently send messages. Classical Atom pub will not work here due some external limits of latencies.
I have come up with several options to connect those datacenters but can’t find the best one.
Options which I have considered are:
1) Forward multicast messages via VPN’s (can VPN handle such big load).
2) Translate all multicast messages to “wrapper messages” and forward them via AMQP.
3) Write specialized in-house gate which tunnels multicast messages via TCP to other two locations.
4) Any other solution
I would prefer option 1 as it does not need additional code writes from devs. but I’m afraid it will not be reliable connection.
Are there any rules to apply for such connectivity?
What the best network configuration with regard to the geographical configuration is for above constrains.
Just wanted to say hello :)
As for the topic, we have not much experience with multicasting over WAN, however, my feeling is that PGM + WAN + high volume of data would lead to retransmission storms. VPN won't make this problem disappear as all the Australian receivers would, when confronted with missing packets, send NACKS to Europe etc.
PGM specification does allow for tree structure of nodes for message delivery, so in theory you could place a single node on the receiving side that would in its turn re-multicast the data locally. However, I am not sure whether this kind of functionality is available with MS implementation of PGM. Optionally, you can place a Cisco router with PGM support on the receiving side that would handle this for you.
In any case, my preference would be to convert the data to TCP stream, pass it over the WAN and then convert it back to PGM on the other side. Some code has to be written, but no nasty surprises are to be expected.
Martin S.
at CohesiveFT we ran into a very similar problem when we designed our "VPN-Cubed" product for connecting multiple clouds up to servers behind our own firewall, in one VPN. We wanted to be able to run apps that talked to each other using multicast, but for example Amazon EC2 does not support multicast for reasons that should be fairly obvious if you consider the potential for network storms across a whole data center. We also wanted to route traffic across a wide area federation of nodes using the internet.
Without going into too much detail, the solution involved combining tunneling with standard routing protocols like BGP, and open technologies for VPNs. We used RabbitMQ AMQP to deliver messages in a pubsub style without needing physical multicast. This means you can fake multicast over wide area subnets, even across domains and firewalls, provided you are in the VPN-Cubed safe harbour. It works because it is a 'network overlay' as described in technical note here: http://blog.elasticserver.com/2008/12/vpn-cubed-technical-overview.html
I don't intend to actually offer you a specific solution, but I do hope this answer gives you confidence to try some of these approaches.
Cheers, alexis