Improving P2P Video Chat in Bandwidth limited Situations

I did a little work this past week to improve our performance, and automatically scale back our bandwidth usage when we were in low bandwidth situations. Primarily we were trying to combat choppy audio, video delay, and low video framerate. It turned into more work than I expected.
Anyway I thought I'd share our approach in hopes that other P2P video chat people here will do the same and we can all learn how to make a LCCS P2P video chat the best it can be.
Let me know what you think, if I suck, etc etc.
Here's a link to a pdf about it: http://www.familyhealthnetwork.com/Improving_LCCS_P2P_Video_Chat_In_Bandwidth_Limited_Situ ations.pdf
-Eric

Hi Eric,
Sorry for not responding earlier (we've been running around w/ hair on
fire... You'll all see why shortly...), but this is really amazing! Thanks a
ton for putting this together and sharing it with us.
Any technical reason to use XMPP as the message relay for the STFU? Seems
like a simple transient message via LCCS could work here.
I'd love to discuss this a little more deeply once the dust settles on our
side.
nigel

Similar Messages

Making P2P Video Chat with FMS

Hi, I'm trying to make a P2P video chat application with Flash Builder 4.5 and Flash Media Server 4.5.
First, I've tried a simple sample application using Cirrus(Stratus). It worked very well!
NetConnection -> NetGroup -> 2 NetStreams -> netstream play and publish -> last, netgroup post
The sample was all OK.
But now, I've changed the server url to my flash media development server.
rtmfp://p2p.rtmfp.net/XXXXXXXXXXXXXX -> rtmfp://my-fms/application
Here is my application trace log about NetStatusEvent:
[object NetConnection] NetConnection.Connect.Success
[object NetConnection] NetGroup.Connect.Success
[object NetStream] NetStream.Play.Reset
[object NetStream] NetStream.Play.Start
[object NetStream] NetStream.Publish.Start
[object NetConnection] NetStream.Connect.Success
[object NetConnection] NetStream.Connect.Success
I couldn't get events about NetGroup and NetStream any more!!!!
When using Stratus, It worked well...
I am not familiar with FMS.. did I skip something in FMS setting?

Nikhil Pavan Kalyan wrote:
Hi,
Thanks for trying RTMFP.
When you made your own application on the development server, did you make any server side scripting on this application or is this a blank application ? The point here is that, for P2P, the Cirrus server has a server script that takes care of bootstrapping one client to the other, which in your case (after moving to your server) is not happening. You need to probably add the scripting for bootstrapping one client to the other.
The following resources can help :
http://help.adobe.com/en_US/flashmediaserver/devguide/WSa4cb07693d1238 84520b86f312a354ba36d-7ffe.html#WSa4cb07693d123884520b86f312a354ba36d- 7ff9
I think I have a similar problem. I am using FMS 4.0 and I can't receive any NetGroup event except NetGroup.Connect.Success. There is no events as NetGroup.Neighbor.Connect or NetGroup.Posting.Notify. I made a discussion (http://forums.adobe.com/message/4429330#4429330) and one of the Adobe employees wrote that I don't need to write any bootstrapping (any server side code):
Chandan_Abhishek worte:
You don't have to implement any server side code. Just keep a blank app on the server and make two client side apps.
Can please someone explain, do I have to write any server side code or not? Do I have to bootstrap clients or not?

Best practice for p2p video chatting

I use netgroup to implement a video chatting program. The # of memebers in this group is small (2 or 3). I find the audio is delayed (sometimes by 4s) and even cut (e.g. I count number from 1 to 10 but the other memeber only heard 1,2,3 then jump to 7). This happens both when the two members are in totally different locations and when they are in the same location using different computers.
I wonder if there is any article about improving the performance and reduce the delay.
Thanks.
ilsh

NetStream.audioReliable is only for 1:1 (DIRECT_CONNECTIONS) or client-server streaming. all data in a P2P multicast stream is partially reliable (where the reliability is controlled by the NetStream.multicastWindowDuration).
a P2P multicast may have a jitter of up to the multicast window duration. shorter values for the window can decrease latency and jitter but may also decrease the reliability of the stream (leading to more glitches and artifacts).
for interactive chat between a small number of participants, you should use the DIRECT_CONNECTIONS form.

P2P video chat.

hi,
I am new to the form and I would like somone to point me in the right direction.
I would like to develop a video/audio chat that also two or more users to video/audio chat.I will also want admin options.
I need to learn this fast, I have programming experience in php and c++ so I am not a total noob.
I guess it should be done using Real Time Media Flow Protocol but I don't know much about it.
Please tell me everything I need to learn in order make this work:books, tutorials, existing scripts, ecc.
thanks.

Hi, I'm new, too. Nice to meet you. I had a little bit of trouble signing in and was reading everything I could to fix the problem, so maybe that's why I noticed this. When I click the "Overview" tab at the top of this forum, the forum page reloads with three things. There's a couple of links to the FAQs and things to read before you post. Above those, there's a caution sign and the text:
"Please note that this forum category is for help with the Flash Player and NOT for users seeking assistance with programming Flash and/or Flex. If you are seeking assistance with programming, click here and choose the proper category to post in."
If you click there, you're directed to a list of forums. I gather there's likely one that better suits your topic.

Reliable p2p video-chat

Hi!
Saying p2p I mean direct (without some kind of account) chat between two workstations each with static IP and opened all needed ports. I have tried plenty pieces of software (linphone, qutecom, jitsi, ekiga, empathy, ...), but have not found acceptable one (direct sip call - like sip:[email protected] - doesn't work at all, or segfaults, or memory leak, and so on).
Is anybody more lucky rather I am?
P.S. I don't mean it must be sip client - probably another p2p ways exists.
Last edited by student975 (2011-08-24 21:51:05)

student975 wrote:
tuxg wrote:It has to be user friendly ?
Probably script-based way is also acceptable.
tuxg wrote:... ssh, ...
OTOH, I'm not sure opening an ssh access is my main dream
I have found this interesting package (it exists in AUR)
http://www.creytiv.com/baresip.html
but have not found any documentation for it yet (searching results in openwrt integration).
I can't help you with SIP things, 'cause they never worked for me, sorry.
I used my proposition (ssh .....) for audio chatting once (and yes there were lags).
Last edited by tuxg (2011-08-25 19:45:06)

Bandwidth for group video chats

Hi everyone
A few things I would like clarified about group video chats before I set one up.
1) Will everyone who joins the chat be able to see each other's videos? Or does only the host have access to see all the videos?
2) How much bandwidth is taken up by a group video chat, and does this affect the host only or all users?
3) What is the maximum number of users allowed on group video chats, including the host? I've seen various figures from 5 to 25.
Thanks, your insight on this is much appreciated!

Hello there,
== https://support.skype.com/en/faq/FA10801/making-a-group-video-call-mac ==
== https://support.skype.com/en/faq/FA1417/how-much-bandwidth-does-skype-need ==
Hth
On ne regarde pas l'avenir dans un rétroviseur !
IMac Intel Core i3 3.2 GHz - RAM 12 GB - OS 10.10.3
Skype 7.8.391
Logitech usb headset or Jabra 250 bt

Excessive bandwidth error when trying to video chat with buddy

I have been unable to video chat with people on my buddylist...when I ask them to join or they ask me to join...my image comes up but soon returns an error box stating "unable to connect due to excessive bandwidth."
When I use menu item "Connection Doctor" it returns ERROR -22
Any ideas ?
Does this have anything to do with OSX10.4.10 upgrade or Safari beta 3.0.2 (522.12) update?

Hi
have you set the Quicktime streaming setting, goto sys prefs/quicktime/streaming/streaming speed, set to 1.5mbps(dont use automatic)
In ichats prefs click on video and change bandwidth limit to NONE.
Restart iChat.
Tell us your down and upload speeds http://www.speedtest.net/
Tony

Video chat p2p

Hi,
I want to build a chat applciation that use webcam (just 1 to 1 webcam ).
So if i use p2p is it free to use the LiveCycle Collaboration Service or i have to buy quota fo usage ??
And for this my appliction waht is better Cirrus or LiveCycle Collaboration Service ??

Hi,
You dont have to buy quota for usage. There is always free limit to get you started. (15 USD). More info can be found at - http://forums.adobe.com/message/2292267
If you use P2P you save on bandwidth usage needed for webcam chat, but might get charged on user minutes.
Thanks
arun

IChat and "Insufficient bandwidth" for video chats

I have spent hours and hours trying to solve this problem, after having spent a lot of money for an iSight specifically for use with iChat, which I now find don't work together. I just made the switch from PC to MAC and have been nothing but amazed until this... Here I have to give a huge "thumbs down", though. Apple has really screwed this one up.
The error when trying to connect video is "Insufficient bandwidth to maintain..." and sometimes "A communications error has occurred".
If anybody out there, especially APPLE can help me fix this, I'd be very appreciative. Note that I had GREAT success with aMSN, Yahoo Messenger, and Skype (for the mic).
Based on lots of reading, here are things I've done without success. Note that I have a very good ADSL-2 20MB connection! The machine is a new Mac mini Intel core duo.
1. Set the QuickTime streaming bandwidth to below what I know the ADSL line can do.
2. Set port forwarding for all the ports I found in all the FAQs, Apple technical documents, etc., etc.
3. Tried turning the UPNP of the ADSL modem both ON and OFF.
4. Made sure there is no firewall.
5. Tried every setting for the video bandwidth setting in iChat.
6. Tried both .mac and AIM accounts. (You never know...)
7. Made sure I have none of the known conflicting software indicated by Apple.
8. Made sure that Apple Remote connections are disabled.
9. Upgraded to 10.4.7.
I have actually read that some people got around this problem by connecting their MAC directly to their ADSL modem and (presumably) turning off its routing functions. But what am supposed to do with my 4 other computers, then?
Apple, please FIX THIS! There are abosolutely tons of people with the same problem and no solution.
Mac mini intel duo Mac OS X (10.4.6)

Ok Ralph... Thanks again.
When you said "two lots", I thought you may have meant that the DHCP server might be giving out IP addresses on two sub-nets (192.168.0.xxx, 192.168.1.xxx, etc.).
Since all of my machines have a static IP (except my business laptop, which I have to drag around to customer sites), I actually tried turning off the DHCP server to see if it helped. It didn't.
Also, I don't quite follow what you're saying about the NIC actually having two addresses (masquerading)... one from the modem and one from the DHCP. This should not be true, as it is the specific purpose of NAT to avoid this. Only the modem/router should really know "who is who" and relay packets appropriately.
The only pseudo-exception I can see for this is if the iChat software on the client and/or the AIM server is passing through the external/internal IP addresses as part of its data packets... ie., not in the IP headers to be handled normally by the NAT layer. If so, then this is just quite simply a design flaw in iChat... and a really stupid one.
One other point... Should I understand from your reply that if I set the modem up for "single user" mode (ie., turn off NAT and DHCP), then I would need to manually forward the right ports?
Mac mini intel duo Mac OS X (10.4.7)
Mac mini intel duo Mac OS X (10.4.6)

P2P multicast and upstream bandwidth

Hi,
I'm in the process of developing a realtime video chat application where multiple users can send video streams simultanuously. The number of users receiving the streams can be very big, e.g. 10 broadcasters and 500 receivers, each receiver should get all streams.
I use RTMFP connections to an FMS and streams are published in P2P multicast groups by passing the groupspec to the NetStream constructor. Currently I'm having problems with audio/video synchronization and video stream 'jumps' (not continuous). From what I read on other threads, this is related to the fact that there is not enough upstream bandwidth for sending the streams. So my questions are:
How to calculate the required upstream bandwidth on every peer for the given example of 10 broadcasters and 500 receivers (is it 10*bandwidth of one stream)?
What settings (on NetStream, Camera, Microphone etc.) should be used for best results and how to adapt them based on the number of broadcasters?
I hope my questions make sense!
Thanks,
Haykel

I have done some tests to find out how much upstream bandwidth is used for different situations. The data is taken from the 'multicastInfo' property of all involved 'NetStream' objects as follows:
Multicast Data: average of 'multicastInfo.sendDataBytesPerSecond' of all streams.
Multicast Control : average of 'multicastInfo.sendControlBytesPerSecond' of all streams.
Different measurements have been done for different situations:
The user is broadcasting and not broadcasting
Varying number of connected users
Varying number of users broadcasting
I have limited the maximum video bandwidth to 16 KBytes with 'Camera.setQuality(1024 * 16, 0)'.
The application uses only application level multicast (passing a groupspec to the NetStream constructor). In this test all peers are on the same LAN and the FMS server is on a remote machine.
The results in Bytes/s (click to enlarge):
What I have noticed is that the outgoing multicast data volume grows by ~15 KByte with every new connected user (receiver) and goes down when the number of broadcasters grows. Is that normal? Does it mean that only the broadcasters are sharing the streams with the other peers? I thought that every peer would share the data with a number of neighbours (i.e. 3) which would share with their neighbours and so on.
During the tests I have also checked for a/v delays and noticed the following:
For every new stream, at the beginning audio and video are in sync but have a delay of ~3 seconds
After ~20 seconds the video delay becomes marginal but the audio remains delayed and so goes out of sync with video
After ~30 seconds the audio delay becomes marginal and goes in sync with video (a/v are now stable)
The delays of 20 and 30 seconds grow up to more than a minute when the number of broadcasters grows
My questions:
Are my measurements correct?
Why is the outgoing multicast data volume growing with every new receiver? Is the publishing stream sending the data to all peers?
Should I expect the bandwidth to grow indefinitly with the number of receivers or will it stabilize at some value?
How to decrease the time required for a/v to become stable (in-sync with a small delay)?
Is P2P multicast a good choice for this kind of applications (up to 10 broadcasters and a very big number of receivers)?
Any advices???
Thanks.

Detecting the optimal upload bandwidth of cam video stream (Dynamic Bandwidth Detection Approach)

Hello folks,
i am discovering the wide world of adobe technologies and i am impressed how seamless all is working.
Anyway i have a tricky problem, at least it seems tricky to me. Here we go: i build up a client (camera+audio conferencing) streaming its camera right to my server (red5). Every client can see the camera streams of each connected particpant. So far so good all is working as expected, we can call this a conferencing solution. Now i would like to improve that by implementing some kind of automatic bandwidth detection in order to adjust the video resolution depending on the current bandwidth "situation" of every participant. Some might have stronger upload bandwidth, some might have a bad or even too low upload bandwidth to even up-stream their video. My client should be able to handle them individually. Those clients who have a good upload bandwidth should be able to up-stream their video in the most high resolution as possible regarding their bandwidth. For those clients having a bad upload connection i assume to reduce their video resolution automatically. The aim behind this automatic bandwidth detection should be to give the most priority to the audio streaming by taking care to not overhelming the bandwidth with heavy camera usage.
I know there is upload speed tests (onBWCheck, etc), but i assume this test is annoying for the participants as well as they are not reflecting dynamic changes.
So lets say i start with a default camera resolution of 640 x 480 px for every client. What i need to know is in general two informtion in order to judge about the current bandwidth is enough for the currently up-streamed camera video:
1) the actual bandwidth used for the up-streaming camera video (based on specific resolution, quality and fps) which is of course dynamicly changing
2) the max. needed bandwidth which is necessary to perfectly up-stream the camera video regarding the specific resolution, quality and fps of the camera settings
So here is my current solutions:
1) the actual bandwidth: every second the currentBytesPerSecond() method of the NetStreamInfo object is of my uploading stream is called to get the currently used bandwidth speed (http://help.adobe.com/en_US/FlashPlatform/reference/actionscript/3/flash/net/NetStreamInfo .html#currentBytesPerSecond). The documentation says about the currentBytesPerSecond() method: Specifies the rate at which the NetStream buffer is filled in bytes per second. I understood from that phrase that literally this is the amount of bytes filled into the network depending on the current network situation. I tested this by using bandwidth limitation and in fact the currentBytesPerSecond() returns the dynamicly changed amount of bytes sent through the network. So thats pretty OK.
2) the max. needed bandwidth: here i struggle. I have three principal ideas to know how much bytes/second are max. needed to up-stream the camera video to the server in a proper way:
* either by experience, which means i manually make several tests to know the max. bytes/second needed for every possible specific resolution, quality, fps (i discovered that different cameras can even produce different results with the same camera settings!)
* calculate the needed bytes/second (actually i use the H246 codec)
* or finally the most crazy but maybe most proper solution: beside the up-stream to the server the client streams additionally to the same client application over the localhost network. I find this last attempt the most interesting because it gives very practicaly accurate result how much bytes must be send because almost the bandwidth of the network through the internet isnt used either. The client sends a stream to himself over the internal network IP (localhost, 127.0.0.1). To achieve this i tried to let the NetConnection being connected to localhost and null and publish the video. But the stream didnt worked. Here is my code:
          private function init():void {
                nc = new NetConnection();
                nc.addEventListener(NetStatusEvent.NET_STATUS, netStatusHandler);
                nc.connect(null);
                ns = new NetStream(nc);
                ns.addEventListener(NetStatusEvent.NET_STATUS, netStatusHandler);
                ns.client = {};
                var camera:Camera = Camera.getCamera();
                if (camera) {
                    videoDisplay.attachCamera(camera);
                } else {
                    Alert.show("You don't seem to have a camera.");
                ns.attachCamera(camera);
                ns.publish("streamname");
                mainTimer.addEventListener (TimerEvent.TIMER, onMainTimerTickHandler);
                mainTimer.start();
            private function onMainTimerTickHandler (evt:TimerEvent): void
                info.text += Number(ns.info.currentBytesPerSecond / 1000).toFixed(2) + " kBytes (OUT)\n";
            private function netStatusHandler(event:NetStatusEvent):void {
                info.text += event.info.code+"\n";
The result is a connection success, but the stream didnt send any data due to the fact that ns.info.currentBytesPerSecond returns 0. Also strange is that the NetSteam or NetConnection didn't return any error.
So folks here i struggle and i ask does anyone out there have any hints or ideas how to solve this tricky one?
Thanks in advance,
Markus

Thanks,I have read that article. Based on that article the NetStreamInfo.maxBytesPerSecond is not an accurate measurement to base dynamic switching on. This seems to be the basis of the bitrate switching in both the longtail player, and the adobe examples that I have tried.   That article suggests using the dropped frames property, in conjunciton with bufferlength to determine if switching is neccessary. Unfortunately I can't seem to find a player online which handles this successfully. That being said, I can't believe I'm the only person trying to implement dynamic bitrate switching for live streams so surely there are some players out there which can do this successfully? If anyone knows of any code available which does this successfully I would appreciate knowing where! The examples provided by Adobe https://www.adobe.com/cfusion/entitlement/index.cfm?e=fms35 unfortunately don't work either.

Once and for all: MSN VIDEO chat on the mac... can it work?

the situation: my friend in France uses an msn account on a PC with the latest version of windows live or some such. I'm in the states on my blackbook. I set up a shiny new msn account, hoping to connect up to him. I've tried adium, msn mac, mercury, and ichat, and nothing will let me connect to his cam, or him to mine.
so here's the golden question: can a mac user, as of right now, video chat with a PC user through the MSN network? if so, and if this is a waste of a new post, please direct me to the instructions. please do not tell us to sign up for AIM accounts, like I saw in another post.
if it's not possible, please explain why not. is it microsoft's msn client for the mac that's about 3 versions behind the pc one? is it a protocol issue? why is this so impossible? is it a social thing, i.e. europeans use msn and PCs and americans use AIM/yahoo and macs? I'm baffled.

It IS possible for a Mac user to do video with people using MSN/Windows Live messenger on Windows, but not possible to do audio.
To do video you will need either Mercury or aMSN. I haven't used either of these, and wouldn't touch the MSN service with a barge-pole, so can't give instructions on how to set it up, but both of them do have video capabilities. As i said though, neither do audio, so unless both participants know sign-language it may be of limited use.
The reason why it is not possible is both a protocol issue and a Microsoft one.
MSN uses secret, proprietary protocols for their service which means that anyone wanting to interoperate with the service has to reverse engineer the protocols and then implement them (and Microsoft can of course move the goal posts at any time by changing the protocols in a new version).
It's a Microsoft problem because they simply aren't interested in developing no cost applications for the Mac, as evidenced by the dropping of both IE and Windows Media Player for OS X (it wouldn't surprise me if Messenger gets dropped by the end of the year), so MSN Messenger lags well behind the Windows version and lacks many of its features.
The situation with MSN/Windows Live Messenger is unlikely to change until Microsoft switches to the IETF standard for IM, XMPP. It may be years before Microsoft switches but it will have to eventually as XMPP is the daddy in enterprise IM and becoming increasingly popular in the consumer space with services such as Google Talk, Gizmo (IM portion) and Live Journal (among thousands of others) utilising it.
Fortunately iChat already supports XMPP (Jabber) for IM, although doesn't yet use the XMPP standards for video or audio (though i'm still hopeful this will change with Leopard), so once MS switches you should be able to set it up in iChat. But like i said, it may take years before all the major IM companies converge on XMPP as the protocol of choice.
Anyway, if you're just interested in video AND audio chats between Windows and Mac OS X you don't have a lot of choices. There's the AIM/Trillian Pro <-> iChat route that you don't want to take, there's the Skype route (much as i loath Skype) which seems to be the preferred option among other forum posters here, and there are a few other alternatives too which i haven't paid much attention too, but there is currently NO way to do both video AND audio between Mac OS X and Windows using MSN/Windows Live.
Or i suppose you could wait for Leopard and hope that Apple provides MSN functionality but i suspect you'd be disappointed.
Not the answer you were looking for i'm afraid but i hope my post helps you find an alternative solution. Don't forget to send feedback to Microsoft and demand they switch to XMPP in MSN/Windows Live Messenger. The more requests the better. ^_^

HELP !! Trying to video chat with a friend I haven't seen in 23 years.

I'm getting the "insufficient bandwidth" error when trying video or audio chat. PowerMac on my end, new iMac on the other end. My friend has successfully done video chat with a friend overseas so it seems that her end is fine. I do have a Linksys WRT54GS (ver 2.0) router and have read a lot about the issues there.
Here's my situation. First of all, when I video conf from my PowerMac to my iMac over our home network (Linksys router), no problems at all. Video and audio are great. I know it's different when going outside my home, but at least that is working. What I'm trying first is just disconnecting the router altogether and running my cable modem direct to the PowerMac. I have not had the chance to test this with a "live" chat partner. I've used "[email protected]" and the video works but no audio (does this mean anything?). In fact, I get the same result whether I go through the Linksys router or direct to the cable modem using this test.
I've done the following so far:
1. Bypassed the Linksys and going direct to the cable modem (I'm going to keep it this way until I get it to work, then I'll mess around with the router).
2. Set streaming preference for Quicktime to 1.5Mbps.
3. Tested my bandwidth at Speakeasy (6.8Mpbs down; 480kbps up)
4. Verified under iChat Preferences that bandwidth limit is set to NONE, microphone is set to iSight, and sound output is set to Line Out (since I have external speakers.)
5. All software is updated to the latest versions.
6. Verified that the mac's firewall is off.
Granted, as I said, I haven't been able to do a test with a live chat partner yet, but as I described above, the test with appleu3test01 acts exactly the same with or without the router so I'm skeptical that my live chat is going to work even with the router gone.
Can anyone help? For now, I just want to get this working without the router.
Thanks,
Ric

In case it's helpful to anyone, my iChatAV video chat appears to work just fine going through my Linksys WRT54GS v2.0 (with latest firmware upgrade) with no special settings for port openings or the like. In addition, the Linksys firewall is enabled (but not the Mac's firewall). The only change I made was to set Quicktime streaming at 1.5Mbps versus Automatic. However, I was not able to do an adequte test before I made this change, only after, so I don't know if this setting was necessary.

Screen Sharing and video chats don't work

I've been searching this forum for awhile and I haven't found a concrete answer to my problem, so if this question has been answered please excuse me. Ok, So here's my situation am running iChat on Mac OS 10.5.1. Whenever I am using my laptop at my school on the network their, I am able to use both screen sharing and video chat. But when I am at home, I am unable to use either one. It begins to try to start the session, but then i receive a message that tells me that a connection could not be established. I know that I have internet access, but I don't know what could be wrong with it.

After investing far to long in looking at this exact issue on my end, reading the seemingly endless threads on the exact same error, trying all the various tweeks and QT settings, bandwidth limit settings, etc etc. proposed here, I have now solved this problem, and the none of the above were part of the solution.
I have perhaps one of the the most common routers on the market, Linksys WRT54GL.
In a totally unrelated network problem (a printing issue), Linksys tech support advised me to reset my router. There is a small button on the bottom. I depressed it for 30 seconds. Rebooted everything and my printer miraculously worked. Yea, I had to reenter my WAP passkey, router admin password etc., but that took 2 secs. So, what does this have to do with iChat?.... The very next time I tried iChat video and turned on and enabled my camera, the recipient immedialty saw me. No connection error.
Problem solved, for me art least. Not sure if anyone has tried this, but makes sooooo much more sense than the other unApple like suggestions I previous tried. Hope this helps.

How many video chats can you use with this server?

I am looking to develop a multi-video chat app that's easy to
build. Can I use Stratus with this? Or do I need to use FMS.

none of the video streams will pass through Stratus. Stratus
only enables end-to-end streams. You are limited only by the amount
of bandwidth the client ends have as far as how many different
streams can be flowing.
if you need to relay streams through a server (the to-many
case, or the end-to-end is blocked by firewalls or certain NAT
configurations cases), you'll need to use FMS.
-mike

Improving P2P Video Chat in Bandwidth limited Situations

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