So no one has any ideas about how to get MeetMe to work with a codec other than ulaw? Is anyone successfully doing it? ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dan Morin Sent: Tuesday, May 03, 2005 10:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] MOH Core uses ulaw... I'm trying to get Asterisk setup as a conference bridge. When I originally tried MeetMe, I was using GSM and as the conference got longer, the delay got worse and worse. From my research, I assumed that it was because MeetMe uses ulaw at its core, so everything is getting transcoded twice and each instant adds more and more delay to the cycle. To test this, I changed all of my connections to ulaw and now I get very minimal delay. However, this is not acceptable for me. I'm anticipating most of my meeting attendees to come in over my VoIP connection and if this voip line is using ulaw, it will significantly reduce the number of simultaneous users that my internet connection can handle. So, it seems to me that I need to change the core codec of MeetMe to something like GSM so that I can get OK call quality, while getting the most out of my Internet connection. Does anyone know how to do this? Am I on the right track or way off with this one? Is anyone using MeetMe with GSM or any other non ulaw codec and not having a problem? Also (sorry so many questions) I'm not thrilled with GSM or iLBC. I know there are a lot of people who like G.729...what are the costs involved with using this one? Thanks in advance. Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050504/a9bd1aed/attachment.htm
Look at the app_conference description on the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20app_conference I believe it does what you want to do, but I really don't know if it works with CVS_HEAD or stable releases. I'd be curious to hear how it affects performance as well. MATT--- -----Original Message----- From: Dan Morin [mailto:DMorin@ABBCOInc.com] Sent: Wednesday, May 04, 2005 1:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] MEETME core uses ulaw? So no one has any ideas about how to get MeetMe to work with a codec other than ulaw? Is anyone successfully doing it? From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dan Morin Sent: Tuesday, May 03, 2005 10:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] MOH Core uses ulaw... I'm trying to get Asterisk setup as a conference bridge. When I originally tried MeetMe, I was using GSM and as the conference got longer, the delay got worse and worse. From my research, I assumed that it was because MeetMe uses ulaw at its core, so everything is getting transcoded twice and each instant adds more and more delay to the cycle. To test this, I changed all of my connections to ulaw and now I get very minimal delay. However, this is not acceptable for me. I'm anticipating most of my meeting attendees to come in over my VoIP connection and if this voip line is using ulaw, it will significantly reduce the number of simultaneous users that my internet connection can handle. So, it seems to me that I need to change the core codec of MeetMe to something like GSM so that I can get OK call quality, while getting the most out of my Internet connection. Does anyone know how to do this? Am I on the right track or way off with this one? Is anyone using MeetMe with GSM or any other non ulaw codec and not having a problem? Also (sorry so many questions) I'm not thrilled with GSM or iLBC. I know there are a lot of people who like G.729...what are the costs involved with using this one? Thanks in advance. Dan
Thank you both for your responses. From looking at the app_conference page in the wiki, it seems as though it is best used for one or two speakers and everyone else just listening. Unfortunately, my company likes to have conferences with 15 or so people and everyone can talk. Am I just reading into the example too much? Will it allow many speakers? Does anyone have experience with app_conference that would be able to comment on its effectiveness? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of BJ Weschke Sent: Wednesday, May 04, 2005 4:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MEETME core uses ulaw? I was going to recommend the same to him last night, but then I started digging into the code there and realized they were transcoding back to LINEAR at their core as well. Now they're not passing that back through a ZapTel psuedo channel like app_meetme does, but I'd be interested to see if that fixes the delay issue discussed. On 5/4/05, mattf <mattf@vicimarketing.com> wrote:> Look at the app_conference description on the wiki: > http://www.voip-info.org/tiki-index.php?page=Asterisk%20app_conference > > I believe it does what you want to do, but I really don't know if itworks> with CVS_HEAD or stable releases. I'd be curious to hear how itaffects> performance as well. > > MATT--- > > -----Original Message----- > From: Dan Morin [mailto:DMorin@ABBCOInc.com] > Sent: Wednesday, May 04, 2005 1:03 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] MEETME core uses ulaw? > > So no one has any ideas about how to get MeetMe to work with a codecother> than ulaw? > > Is anyone successfully doing it? > > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of DanMorin> Sent: Tuesday, May 03, 2005 10:26 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] MOH Core uses ulaw... > > I'm trying to get Asterisk setup as a conference bridge. When Ioriginally> tried MeetMe, I was using GSM and as the conference got longer, thedelay> got worse and worse. From my research, I assumed that it was becauseMeetMe> uses ulaw at its core, so everything is getting transcoded twice andeach> instant adds more and more delay to the cycle. To test this, Ichanged all> of my connections to ulaw and now I get very minimal delay. > > However, this is not acceptable for me. I'm anticipating most of mymeeting> attendees to come in over my VoIP connection and if this voip line isusing> ulaw, it will significantly reduce the number of simultaneous usersthat my> internet connection can handle. > > So, it seems to me that I need to change the core codec of MeetMe to > something like GSM so that I can get OK call quality, while gettingthe most> out of my Internet connection. Does anyone know how to do this? Am Ion> the right track or way off with this one? > > Is anyone using MeetMe with GSM or any other non ulaw codec and nothaving a> problem? > > Also (sorry so many questions) I'm not thrilled with GSM or iLBC. Iknow> there are a lot of people who like G.729...what are the costs involvedwith> using this one? > > Thanks in advance. > Dan > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Dan Morin wrote:> Thank you both for your responses. From looking at the app_conference > page in the wiki, it seems as though it is best used for one or two > speakers and everyone else just listening. Unfortunately, my company > likes to have conferences with 15 or so people and everyone can talk.There is no way to bridge (mix) compressed audio streams together. They must be decompressed into some simple audio format, mixed, and then recompressed. app_conference can do a little better when all the callers are using the same codec and only one is speaking, because then it does not need to mix the audio together. Anytime mixing is required, though, the audio is going to be transcoded.