SIP in higher education

[John Todd <>]

I have had tests with 23 users on PRI coming into an Asterisk MeetMe 
conference with good success.  I think that the viability of such 
experiments is very site-dependent and telco-dependent.

Note that there are quite a few more echo cancellation tricks that 
have been incorporated into Asterisk in recent months for PRI, so 
there are hopefully even better methods for coping with PSTN echo 
than before.

I have had 40+ members in SIP/IAX combined conferences, but many of 
those were muted because I added them to the conference on a 
different access number which was one-way.  I've found that above 10 
"normal" people (not telephony admins) things fall apart quickly 
because someone has an umuted phone, or music on hold, or on a cell 
phone, or whatever - the noise floor rises to an unacceptable level 
without selective muting.

JT


> Aaron:
>
> The MeetMe service on * is intended for 6-way, but with good engineering
> I suppose you could exceed that.
>
> The best place for info is at:
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MeetMe
>
> One SA said they get 'tunnel' at 8-9 users, and echo at 10+ using
> ISDN/PRI. If it is a total SIP connection, I'm not sure we haven't
> experimented with that just yet.
>
> The links at the bottom of the VOIP-INFO page will give you better info
> on dimensioning * as well as the MeetMe application including how to
> limit users.
>
> The beauty of * is that since its Open Source you can re-engineer it to
> your needs. And if you figure out how to improve on the MM or MM2 apps
> I'm sure others would be ecstatic to want a copy.
>
> Derek Abrams
> Oregon State University
>
>
> -----Original Message-----
> From: Aaron Solomon using FreeBSD 6.0 
> [mailto:]
> Sent: Thursday, February 09, 2006 6:41 PM
> To: 
>
> Cc: 
>
> Subject: Re: [sip.edu] SIP.edu Call Notes - 2/2
>
> On Thu, Feb 09, 2006 at 11:38:20AM -0500, Dennis Baron wrote:
> >  IP protocols. It was first used for voicemail
> >  service, and now also provides voicemail-to-email service and
> >  conference bridge service.
>
> Hi Mark,
>
> I have a question about the capacity of the conference bridge service of
> Asterisk.  Did you have any idea how many people can join a session
> concurrently?
>
> In Taiwan, we are still surveying the "best" conference bridge we need,
> but many users told us they couldn't wait, and Skype is an available
> solution which supports conference calls for them, so they just gave up
> SIP and adopted Skype as their VoIP solution. 
> This really makes me sad.  :-( 
>
> However, as I know, Skype only supports up to 5 people in a conference
> call.  According to your experience, if you have any suggestion about
> SIP solution for conference bridge, I shall appreciate very much.
> Certainly, a less expensive solution is preferred, because it won't be
> appropriate for us to spend too much money on this single device when
> the utilization rate is not high.
>
> Sincerely,
> Quincy
> Feb. 10



--
John Todd
Networking, Tello Corp.

  +1-650-581-2405