<br><br><div class="gmail_quote">On Fri, Aug 19, 2011 at 6:39 AM, Jim Boykin <span dir="ltr"><<a href="mailto:boykinjim@gmail.com">boykinjim@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
convert mp3 to sln, this itself will give you quiet a big capacity boost.<br></blockquote><div><br></div><div>How does sln boost capacity?</div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div><div></div><div class="h5"><br>
On Wed, Aug 17, 2011 at 12:21 PM, Morten M. Hansen <<a href="mailto:mmh@bellcom.dk">mmh@bellcom.dk</a>> wrote:<br>
> On 2011-08-16 21:14, Warren Selby wrote:<br>
>> 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).<br>
><br>
> It's a shared stream. When testing now, new listeners doesn't spawn new<br>
> mpg123 processes.<br>
><br>
>> 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.<br>
><br>
> We are using the zaptel and ztdummy kernel module, and we haven't<br>
> noticed any problems with the audio quality yet. Should we be worried<br>
> about this when the load gets higher?<br>
><br>
>> Will you have sufficient bandwidth allocated to you for that many simultaneous calls?<br>
><br>
> Good point. We will have to do some calculation and research on what EC2<br>
> offers here.<br>
><br>
>> Is there going to be any codec transcoding going on? Can you generate your streams in the preferred codec, instead of mp3?<br>
><br>
> The source is an icecast server streaming mp3. I haven't figured out a<br>
> way to get around that. But from what I understand its just one<br>
> reencoding for all the listeners.<br>
><br>
>> 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.<br>
><br>
> I had a talk with our IAX2 trunk provider and they told me that we could<br>
> expect better performance from a SIP trunk. They also had a limit on<br>
> 2000 channels, so we may have to look for another trunk.<br>
><br>
> Are there any tools or services to simulate a lot of IAX2 or SIP users<br>
> that you can recommend? How do you test how many users an asterisk<br>
> system can handle?<br>
><br>
> Thank you for taking the time to reply.<br>
> Morten<br>
><br>
>> Thanks,<br>
>> --Warren Selby, dCAP<br>
>><br>
>> On Aug 16, 2011, at 10:16 AM, "Morten M. Hansen" <<a href="mailto:mmh@bellcom.dk">mmh@bellcom.dk</a>> wrote:<br>
>><br>
>>> Hi<br>
>>><br>
>>> I'm hoping someone could comment on how our setup will perform under<br>
>>> larger loads.<br>
>>> Its a quite simple setup, with Asterisk 1.6.2 on Debian 6 on an EC2 large<br>
>>> instance (7GB RAM, 2 virtual cores with EC2 compute units).<br>
>>> Using an IAX2 trunk we offer normal phones to dial in and listen to a mp3<br>
>>> stream using music on hold.<br>
>>><br>
>>> If we wanted to let 1000 users listen to the stream at the same time,<br>
>>> would that be possible? What limits will we hit? How about 10000 users?<br>
>>><br>
>>> Regards<br>
>>> Morten<br>
>>><br>
>>><br>
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>>><br>
><br>
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</div></div></blockquote></div><br><br clear="all"><div><br></div>-- <br><div>Regards,</div><br>Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445<br>