[asterisk-users] Asterisk scaling

[Warren Selby]

Sorry for the top post, this is from my phone. 

What you need to look at are the following:

Is it going to be just one mp3 stream that is shared across all users (I.e everyone hears the same thing at the same time), or is it 1000 separate mp3 streams (everyone always starts at the beginning of whatever they are hearing). 

Are you going to have reliable timing generation on an EC2 instance, since IAX streams and music on hold playback will sound bad if the timing isn't good. 

Will you have sufficient bandwidth allocated to you for that many simultaneous calls?

Is there going to be any codec transcoding going on?  Can you generate your streams in the preferred codec, instead of mp3?

I think if you're just using one stream spread across all the callers, you'll have much better performance from the system as a whole. You may want to look at the quality differences between a SIP trunk and an IAX trunk as well. 

Thanks,
--Warren Selby, dCAP

On Aug 16, 2011, at 10:16 AM, "Morten M. Hansen" <mmh at bellcom.dk> wrote:

> Hi
> 
> I'm hoping someone could comment on how our setup will perform under
> larger loads.
> Its a quite simple setup, with Asterisk 1.6.2 on Debian 6 on an EC2 large
> instance (7GB RAM, 2 virtual cores with EC2 compute units).
> Using an IAX2 trunk we offer normal phones to dial in and listen to a mp3
> stream using music on hold.
> 
> If we wanted to let 1000 users listen to the stream at the same time,
> would that be possible? What limits will we hit? How about 10000 users?
> 
> Regards
> Morten
> 
> 
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