[Asterisk-Users] Meetme versus app_conference

[trixter aka Bret McDanel]

On Sat, 2006-06-03 at 23:20 -0500, Erick Perez wrote:
> As stated here:
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe
> 
> A Meetme room uses Ulaw as the audio codec, so if the other channels
> use different codecs, then * will transcode.
> 
> Does the app_conference application works the same way?
> Or if i have SIP/g729 users and i create a conference with other users
> also at g729 asterisk will not transcode (when using app_conference)?
> 
> Thanks,
> 

app_conference doesnt require a timer unlike meetme

app_conference claimed (I dont know if meetme has upgraded) that it only
transcodes once per codec in question for everyone where meetme would
transcode for each person.  IE you have 3 callers, 1 on GSM 2 on speex.
Any frames from the GSM caller get transcoded twice, one for each
participant using speex.  With app_conference it will transcode once and
send the same frame to both callers - so its slightly more efficient in
that aspect.

meetme I believe has some additional functionality, such as the menu
system.  I dont know if app_conference has added in the DTMF detection
stuff to add menus or not.

I believe that there is a mysql/postgress addon to app_conference that
sticks the info about the current users in a database in realtime that
way you can see who is on, even comes with a web based example php
program to pull this info and display it to callers.  I dont know where
this modification is offhand.

For any given one situation one is probably better than the other,
however becuase they work slightly differently you may have to use one
over the other since they dont afaik support identical features.

I have heard rumors but no facts that app_conference generally can
support a higher caller load too.

> 
-- 
Trixter http://www.0xdecafbad.com     Bret McDanel
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