[asterisk-users] Asterisk scaling

[Abdul Basit]

On Fri, Aug 19, 2011 at 6:39 AM, Jim Boykin <boykinjim at gmail.com> wrote:

> convert mp3 to sln, this itself will give you quiet a big capacity boost.
>

How does sln boost capacity?


>
> On Wed, Aug 17, 2011 at 12:21 PM, Morten M. Hansen <mmh at bellcom.dk> wrote:
> > On 2011-08-16 21:14, Warren Selby wrote:
> >> 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).
> >
> > It's a shared stream. When testing now, new listeners doesn't spawn new
> > mpg123 processes.
> >
> >> 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.
> >
> > We are using the zaptel and ztdummy kernel module, and we haven't
> > noticed any problems with the audio quality yet. Should we be worried
> > about this when the load gets higher?
> >
> >> Will you have sufficient bandwidth allocated to you for that many
> simultaneous calls?
> >
> > Good point. We will have to do some calculation and research on what EC2
> > offers here.
> >
> >> Is there going to be any codec transcoding going on?  Can you generate
> your streams in the preferred codec, instead of mp3?
> >
> > The source is an icecast server streaming mp3. I haven't figured out a
> > way to get around that. But from what I understand its just one
> > reencoding for all the listeners.
> >
> >> 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.
> >
> > I had a talk with our IAX2 trunk provider and they told me that we could
> > expect better performance from a SIP trunk. They also had a limit on
> > 2000 channels, so we may have to look for another trunk.
> >
> > Are there any tools or services to simulate a lot of IAX2 or SIP users
> > that you can recommend? How do you test how many users an asterisk
> > system can handle?
> >
> > Thank you for taking the time to reply.
> > Morten
> >
> >> 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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-- 
Regards,

Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445
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