On Fri, May 27, 2011 at 05:30:02PM +0000, daniel@danielknoll.de wrote:
>
> What is better more cores (eg. 2x quadcore) or more CPU speed for a server
> that handle a lot of of Meetme Concerences with hundreds of concurrent G711
> alaw Channels (no transcoding) ?
>
> in my opinion, more cores are better, because Asterisk ist multithreded and
> each channel has a good chance to distribute to the cores.
>
> is that right or what do you think?

I don’t have an answer for you (too many unknown variables to say one way or
another) but I do know that MeetMe currently is serialized onto a single CPU.

For example, all the conference participants threads queue up their audio data
on their pseudo channels which can happen in parallel on SMP systems. Then
they wait for the “masterspan” to mix the audio for all conferences currently
active. This happens on a single CPU to simplifly keeping channels
synchronized to whatever timing source is your master. When this was
originally written there were not too many SMP systems in use. Then the mixed
audio is queued back on the channels. Finally, all the user space channel
threads can read out the mixed audio which can happen in parallel on SMP.

If you have too many conferences, one CPU may not be able to mix all the audio
and you will have audio problems even if there are 7+ other CPUs that are
essentially idle while waiting for one CPU to mix everything. You should be
able to handle 512 conference participants on a modern server system without
problem. The current trunk of DAHDI linux limits the number of open pseudo
channels to 512 for this reason. [1]

[1] http://svn.asterisk.org/view/dahdi?view=revision&revision=9610

The new ConfBridge module in the upcoming 1.10 release may not have this
limitation.

On Tue, Mar 22, 2011 at 12:53 PM, satish patel wrote:
> Hey Guys!
>
> We have two Asterisk with A102D Sangoma cards now i want to connect them
> back-to-back over PRI line via Cross-cable so what would be the
> configuration specially timing source and all? anybody did it before like
> this ?
>
> I want to make sure everything before putting in production.. (saving my
> downtime)
>
> -S
>

If is no different then setting up the card to connect with a telco.
One Asterisk box will be the net and the other is cpe. You can use
whatever protocol national, 5ess, etc you like. Any reason not to join
the boxes via SIP?

Ryan

> I’d suggest try recording in ulaw first and then convert all to wav after.

> It may have something to do with timing since you’re using iax, what are you using as a timing source? Hardware or software?
>
Sent from my iPhone

Hi All

Just finished setting up a vm with centos 5.5 and asterisk 1.8.3

Using timerfd as a timing source.

Has anyone got a similar setup in production ?

How’s performance?

Thanks,

Neeraj 

Hi All,

for a vicidial server which uses only voip,
which is the minimum telephony card which would provide the required clock timing source for conferences to work properly ?

Maybe the Digium TDM410PLF card
without any daughter card
would do the job ?

Thank you very much for supporting.

Have a nice week-end,
Mike