I have done a lot of testing and modifications to the available app_conference code in the last few weeks and can confirm that it is much more efficient than using meetme in the 1.2 Asterisk tree. I have altered app_conference to do some other things that meetme does like entry/exit sounds and some things that meetme doen't allow you to do like optional DTMF inband and/or RFC broadcasting to conference participants.
Overall I have found the app_conference code easier to work with and modify than meetme, and it seems to be a much more efficient and streamlined conferencing platform. The icing on the cake is no pseudo zap channels or required zaptel timer. You can also drop an AGI script into the conference by an Originate and use it locally unlike with meetme. It has not been tested as much as the meetme code of course, but it can function rather well as a meetme replacement in most cases. In our specific experience(for VICIDIAL) we saw the overall load on one of our servers cut in half after switching from meetme to app_conference. I am doubtful as to whether it will ever be included as part of Asterisk due to the reluctance of it's original author to sign the code over to Digium. If you are interested here is our altered app_conference code, tested to work on Asterisk 1.2.8: http://sourceforge.net/project/shownotes.php?release_id=421962&group_id=95133 MATT--- On 6/4/06, trixter aka Bret McDanel <[EMAIL PROTECTED]> wrote:
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 Belfast IE +44 28 9099 6461 DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.3 (GNU/Linux) iD8DBQBEgmWc+1olxlzQw5cRAswDAJ0ZKWgv+xKaikBoKunMnAwi9JoL2ACeNuJS exctSR9dUT96NdiSs5itx78= =3YCf -----END PGP SIGNATURE----- _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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