Conference Rooms
I have setup a conference room and it works like a treat, I have had 10 people connected with no problems, all users where connecting via a SIP trunk to the conference room box
I have all user connecting using g729 codec (8k)
Questions I have are
1. How many people can you have in a conference room ? and what are the limitations eg bandwith,processor power etc
2. When there is a conference and one person is talking only and the rest listening and I am using g729 is there only 8k of bandwith required or is there still 8k used by every user connected ?
eg 10 user 80k or 10 user 8k ?
I want to be able to setup a conference call with 200 users and need to understand how it all works ?
Thanks in advance
Piri
What type of connection do you have? It's not just the bandwidth that can cause problems. A consumer DSL or cable modem can't handle all the sessions required for more than 5-6 calls.
The actual figure is closer to 14k per session. Depending on what your MTU is set to and other traffic issues it could run as high as 20k. Transport adds even more overhead (DOCSIS, ATM, HDLC, PPP etc).
g.729 is not a variable rate CODEC, Annex D and E support adaptive compression however Asterisk does not support this feature and the Annex has yet to be ratified.
Bottom line is each channel occupies the full bandwidth up and down.
Can you give me an example for what band width would be required for
10 people in a conference call
100 people in a conference call
Also is there additional CPU Load ?
My understanding is that if there is only one person talking as one then no much bandwidth would be required all, am I wrong ?
Thanks
Piri
Did you read my reply?
You did not answer my questions.
No matter how many ways you answer the question the answer will still be the same. G.729a is not an adaptive CODEC and every channel occupies the bandwidth. The bridge has to repeat the audio to each leg of the conference (every participant).
For 200 users with G.729a you would be stressing every limit of Asterisk. At least 8 CPU cores would be required.
Additionally you would have to load share the incoming calls across several providers.
Lastly you would need at least 4Mbps of symetric bandwidth. You may get away with two bonded (not load shared) MLPPP T1's. Verizon and AT&T have bonded offerings available.
15M Down / 2M Up VDSL
Not a chance, even if you could sustain the 2M upload speed (it's probably closer to 1.5M) the amount of TCP sessions would crash the DSL modem.
You might get the bandwidth if your server was in a data center.
How much do you reckon I would need ?
As I indicated 4M would do the trick.
If you are interested we have tier 1 colo space available in Cleveland OH.
We can install and maintain the conference server for you.
Please send me a PM if you would like more information.
for a conference server you should look at FreeSwitch it is much better at it than asterisk
Here is Voip-Info's section on Bandwidth:
http://www.voip-info.org/wiki-Bandwidth+consumption
But using LOTS of G.729 is only heavily CPU intensive if Asterisk is TRANSCODING to or from it - here is the page that I think you are asking about - Dimensioning an Asterisk System:
http://www.voip-info.org/wiki/view/Asterisk+dimensioning
If you had 200 people in a conference bridge and all of them were using G.729, the load should only be slightly more than if they were ALL using G.711 - however, if there was a mix, then transcoding would come into play, and CPU utilization would go through the roof.
Finally, the most people I have ever been able to put into a conference simultaneously was 27 - this is on a Dual-Processor Dual-Core Xeon @ 3.2 GHz with 4 Gig's of RAM and I ran out of phones and lines not CPU - with every phone we have, plus 6 cel phones from offsite, we didn't even break 3% CPU utilization (looking at top) when the conference is active - this was a G.711 conference, so no transcoding.
Greg
Basically the cost for you to get 200 lines would be very high. With a typical Cbeyond t1 line SIP account you get 6 lines you can add more. With a PRI you get 24 lines. So talk to a telco and see what the cost of 200 lines would be then you would need 2 bonded t1 lines. Your looking at 800$ a month for the t1 lines and more for the extra lines it would take to get your system to have 200 simultaneous calls. If you do this please let us know how it works out. Most people say 1 trixbox is only good for 60-70 simultaneous calls so you might need 2 or 3 servers. I don't know because I have not had more than 50 phones on an account. My company will not accept jobs larger than that.
M2M - Ths is more of a Asterisk issue and not trixbox. Asterisk could be tuned for 200 conferences. FreeSwitch works even better and it might be just the excuse I am looking forward for a FreeSwitch deployment.
Some of the folks my company works with already have a large FreeSwitch bridge up so we have the resources to do this on either technology.
You may have noted I offered him to collocate his server in our data center. It would be far less than the numbers you quotes. In fact the it would be per minute only plus a small fee for the server space and power.
I am very interested in at least talking to the OP. This is an exciting opportunity for OST and properly done can save huge OpEx dollars over a traditional conference provider.
What do you need to know ? PM ?
What do you need to know ? PM ?
If you are interested in discussing possible solutions please send me a PM.
I won't send unsolicited PM's, there is of course no obligation.
I am very interested in the application and may be able to assist.
Thanks...Scott
I was trying to work out what PM stands for but now I know, What is your email address ?
PM = Private Message
Click on my Screen Name in the box next to the message, I can then send you my email without having to post it in a public forum.


Member Since:
2007-02-08