Participants limits of webrtc app in chrome

[Scott Haynes]

I've got a webrtc app that is working and we've been able to get about 8 people into one session. I'm having some audio and video issues( video is choppy and/or garbabled ) after we've been in the call for a while, or if we quit and try to rejoin.

I've based my work largely on the webrtc app listed on webrtc.org site, and we're using the release candidate of chrome. V 26.1410.64 or whatever. I'm using the stun server, but not the turn server when setting up the connections.

Questions:
What's the best version of chrome/chromium to use for webrtc?
Are there current limits to the number of people in a call?
Would adding a turn server or removing the stun performance?

Thanks Scott

[Vikas]

Hi,

Sorry, but not aware of any video choppiness issues for M26. You
should check if you are getting packet loss. There had been some
reports related to missing audio which you can check in webrtc
tracker. Regarding the limit, currently only 10 peer connections can
be created, see issue 1343.

AFAIK, TURN server can help improve connectivity between 2 clients as
compared to STUN, but yes their is added latency for the calls that go
through TURN server.

/Vikas

[Scott Haynes]

How do I check the tracker?

Scott

[Vikas Marwaha]

https://code.google.com/p/webrtc/issues/list

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[Scott Haynes]

If I have a 4 way call, how many peerConnections should each client have?  3?  

Scott

[Justin Uberti]

You are creating a full mesh conference. This type of conference will start to consume a lot of CPU and bandwidth as the number of users grows - can you tell if you are hitting CPU or bandwidth limits?

For mesh video conferences, I would suggest having 5 or fewer users. (Audio conferences can be significantly larger.)

[Silvia Pfeiffer]

I've successfully used SimpleWebRTC and signalmaster on GitHub to do 4-way video conferencing:
https://github.com/andyet/signalmaster
https://github.com/HenrikJoreteg/SimpleWebRTC

Silvia.

[Scott Haynes]

I'll double check the CPU on Monday, and I just saw the peerConnection getStats function.  I'm going to implement something to grab the stat info this weekend, and I'll have more details on monday afternoon.

mesh video conferences?  

We're on a private network, and the participants are located from Massachusetts to Florida on the east coast.  

Scott 

[Vikas]

Hi,

Yes you can retrieve that information through getstats api. You can
refer to the constraints-and-stats demo here:-
http://webrtc.googlecode.com/svn/trunk/samples/js/demos/html/constraints-and-stats.html.
Also you can monitor some stats through chrome://webrtc-internals on a
separate chrome tab.

/Vikas

On May 13, 6:52 am, Piotr Małecki <piotr...@gmail.com> wrote:
> Hi,
>
> I'm new in webrtc topic, but could someone tell me, if there is any way to
> check what latency and packets loss is during audio/ video conferencing.
>
> Piotr
>
>
>
>
>
>
>
>

[Hrishikesh Kulkarni]

Mesh based conferencing has limitations in terms of bandwidth and CPU. Check if you observe any packet loss failures. 

We have a hub-spoke model server at OneKlikStreet. We have limited per conference to 4way, but we can do VGA at 300kbps.

regards,

Rishi

[Silvia Pfeiffer]

Hi Sumanta,

webrtc2sip is a gateway to go from WebRTC calls to SIP calls, such as
making a phone call in your browser to connect to a normal phone.

A websocket server is just a Web server that supports a connection
from a Web browser through web sockets (see
http://en.wikipedia.org/wiki/WebSocket). I'm using a websocket server
to do the signalling between WebRTC calls (without going out to the
phone network).

I've used node.js as my web server, first with
https://github.com/Worlize/WebSocket-Node for websockets and later
http://socket.io/, the latter of which is more flexible.

HTH.

Regards,
Silvia.

On Thu, May 16, 2013 at 10:31 PM, Sumanta Sen
<sen.suma...@gmail.com> wrote:
> Hi Silivia,
>
> Have you used webrtc2sip as the websocket server.
> Or any other server?
>
>
> Thanks,
> Sumanta