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, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama
----- Erick Perez <eaperezh@gmail.com> wrote:> 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)?It is not possible to mix conference audio together without converting it to an uncompressed form first. app_meetme in Asterisk 1.2.x certainly does do more transcoding (both inbound and outbound) than is absolutely needed, which app_conference does not do. However, app_meetme in SVN trunk (soon to be Asterisk 1.4) tries to minimize the amount of transcoding by avoiding the decoding of incoming audio from channels that are not speaking and by re-using the transcoded output for channels that share a format (codec). This should make it perform similarly to app_conference in many situations. -- Kevin P. Fleming Senior Software Engineer Digium, Inc.
trixter aka Bret McDanel
2006-Jun-03 21:46 UTC
[Asterisk-Users] Meetme versus app_conference
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! -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060603/e61a8c3f/attachment.pgp
Kevin, if I use ulaw for my sip users and my sip providers.... will i minimize the "transcoding" hit to uncompressed mode to my server? or will the load be the same even if I use g729 everywhere? Im trying to optimize my setup as to do transcoding/uncompressing to a minimum. On 6/3/06, Kevin P. Fleming <kpfleming@digium.com> wrote:> > ----- Erick Perez <eaperezh@gmail.com> wrote: > > > 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)? > > It is not possible to mix conference audio together without converting it to an uncompressed form first. app_meetme in Asterisk 1.2.x certainly does do more transcoding (both inbound and outbound) than is absolutely needed, which app_conference does not do. However, app_meetme in SVN trunk (soon to be Asterisk 1.4) tries to minimize the amount of transcoding by avoiding the decoding of incoming audio from channels that are not speaking and by re-using the transcoded output for channels that share a format (codec). This should make it perform similarly to app_conference in many situations. > > -- > Kevin P. Fleming > Senior Software Engineer > Digium, Inc. > > _______________________________________________ > --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 >-- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama
----- Erick Perez <eaperezh@gmail.com> wrote:> Kevin, if I use ulaw for my sip users and my sip providers.... will i > minimize the "transcoding" hit to uncompressed mode to my server? > > or will the load be the same even if I use g729 everywhere? Im trying > to optimize my setup as to do transcoding/uncompressing to a minimum.The choice of codec does not effect the amount of transcoding that needs to be done, only the amount of work required to do it. If you use all G.729, and the conference normally has a small number of speakers (one or two) then the amount of transcoding required will be very small as all the non-speaking parties will receive the same output stream. app_conference takes this one step further and doesn't even decode/encode at all if there is only on speaker and everyone is using the same codec... but app_meetme cannot do that. -- Kevin P. Fleming Senior Software Engineer Digium, Inc.