Re: [asterisk-users] MeetMe Limits
On Sun, 8 Jun 2008, Matt Florell wrote: > Hello, > > We routinely run meetme with over 140 ULAW channels connected to 70 > meetme rooms with no issues on an Intel Core 2 Quad core CPU. > > The major factor in capacity would be your CPU and RAM capacity. If > you have at least a base-level P4 you don't need to worry about 12 > participants. I'd echo that too - recently been playing with Page() which creates a MeetMe conference behind the scenes - on my 1GHz boxes, 15 SIP phones in a page group produces virtually no additional load on the box. Gordon > > MATT--- > > On 6/8/08, Adrian Marsh <[EMAIL PROTECTED]> wrote: >> I've got to agree.. I've never given it much thought either... >> >> All of my calls are SIP/IAX based, coming in from the PSTN from a peer >> like that too.. >> >> I've never tracked the total number of conference users... But I'll bet >> we've hit at least 10.. And I've never seen the CPU go above 10%.. And >> that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it >> will be setup-specific.. So I would look at your CPU and memory stats, >> and run some tests and monitor that.. >> >> >> A. >> >> >> -Original Message- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of John >> covici >> Sent: 08 June 2008 16:34 >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] MeetMe Limits >> >> 12 people is nothing -- I do 20 regularly -- however you may want to >> have them come in as muted or tell them to mute themselves, because the >> latency can cause very severe echoes if they are on a speaker phone or >> cell phone. >> >> on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote > Actually I think >> they will all be calling in using regular pstn phones > and cell >> phones. >> > >> > Sam >> > >> > Al Baker wrote: >> >> The 2 big questions are: >> >> -Are all participants using QoS end to end ? >> >> >> >> -Are all of them using the SAME CODEC. As the amount of Transcoding >> goes up, > > the work on the * box goes up and can be a problem. >> >> >> >> Sam wrote: >> >>> I am thinking about using my asterisk server to host a conference >> with > >> about 12 other people from around the USA. Bandwidth issues >> aside, will > >> this work or will all the different latencies cause >> issues? Yea I know, > >> I could just "try it and find out" but it is >> going to take alot of time > >> to get everyones schedule to line up, I >> don't want to go through the > >> trouble if I will just be >> disappointed. >> >>> >> >>> Thanks, >> >>> >> >>> Sam >> >>> >> >>> ___ >> >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com >> -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or >> update options visit: >> >>>http://lists.digium.com/mailman/listinfo/asterisk-users >> >>> >> >>> >> >>> >> >> >> >> ___ >> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com >> -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update >> options visit: >> >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> > >> > ___ >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> > > asterisk-users mailing list > To UNSUBSCRIBE or update options >> visit: >> >http://lists.digium.com/mailman/listinfo/asterisk-users >> >> -- >> Your life is like a penny. You're going to lose it. The question is: >> How do >> you spend it? >> >> John Covici >> [EMAIL PROTECTED] >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
Forgot to address your second question. DAHDI, that's a good one :) The channel type doesn't seem to matter. One has all agents on Zap channels through channelbanks with all calls coming in over IAX and monitoring done through SIP. One has all SIP agents with all calls coming in over SIP trunks, and another has SIP agents with calls coming in over Zap T1 channels. MATT--- On 6/8/08, Matt Florell <[EMAIL PROTECTED]> wrote: > Hello, > > The load is usually quite high because this is VICIDIAL inbound call > center traffic with full Asterisk-based recording. On a system with > 70-80 Meetme rooms running with 2 participants each doing full > Asterisk-based recording in each Meetme room the loadavg stays between > 2.00-4.00 on a Quad-core Intel core 2 Quad processor with 4GB RAM. I > have three systems like this in place at different call centers and > the load is consistent for all three of them. Usually we put less load > on a single server, but these were inbound-only scenarios which is > less load than outbound. > > > MATT--- > > > On 6/8/08, Steve Totaro <[EMAIL PROTECTED]> wrote: > > Matt, > > > > Could you share the CPU usage, memory, and load average in the > > scenario you describe? What type of ULAW channels > > (SIP,DAHDI,IAX), or does it not matter? > > > > Thanks, > > > > Steve Totaro > > > > > > On Sun, Jun 8, 2008 at 5:29 PM, Matt Florell <[EMAIL PROTECTED]> wrote: > > > Hello, > > > > > > We routinely run meetme with over 140 ULAW channels connected to 70 > > > meetme rooms with no issues on an Intel Core 2 Quad core CPU. > > > > > > The major factor in capacity would be your CPU and RAM capacity. If > > > you have at least a base-level P4 you don't need to worry about 12 > > > participants. > > > > > > MATT--- > > > > > > On 6/8/08, Adrian Marsh <[EMAIL PROTECTED]> wrote: > > >> I've got to agree.. I've never given it much thought either... > > >> > > >> All of my calls are SIP/IAX based, coming in from the PSTN from a peer > > >> like that too.. > > >> > > >> I've never tracked the total number of conference users... But I'll > bet > > >> we've hit at least 10.. And I've never seen the CPU go above 10%.. And > > >> that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it > > >> will be setup-specific.. So I would look at your CPU and memory stats, > > >> and run some tests and monitor that.. > > >> > > >> > > >> A. > > >> > > >> > > >> -Original Message- > > >> From: [EMAIL PROTECTED] > > >> [mailto:[EMAIL PROTECTED] On Behalf Of John > > >> covici > > >> Sent: 08 June 2008 16:34 > > >> To: Asterisk Users Mailing List - Non-Commercial Discussion > > >> Subject: Re: [asterisk-users] MeetMe Limits > > >> > > >> 12 people is nothing -- I do 20 regularly -- however you may want to > > >> have them come in as muted or tell them to mute themselves, because > the > > >> latency can cause very severe echoes if they are on a speaker phone or > > >> cell phone. > > >> > > >> on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote > Actually I think > > >> they will all be calling in using regular pstn phones > and cell > > >> phones. > > >> > > > >> > Sam > > >> > > > >> > Al Baker wrote: > > >> > > The 2 big questions are: > > >> > > -Are all participants using QoS end to end ? > > >> > > > > >> > > -Are all of them using the SAME CODEC. As the amount of > Transcoding > > >> goes up, > > the work on the * box goes up and can be a problem. > > >> > > > > >> > > Sam wrote: > > >> > >> I am thinking about using my asterisk server to host a > conference > > >> with > >> about 12 other people from around the USA. Bandwidth > issues > > >> aside, will > >> this work or will all the different latencies cause > > >> issues? Yea I know, > >> I could just "try it and find out" but it > is > > >
Re: [asterisk-users] MeetMe Limits
Hello, The load is usually quite high because this is VICIDIAL inbound call center traffic with full Asterisk-based recording. On a system with 70-80 Meetme rooms running with 2 participants each doing full Asterisk-based recording in each Meetme room the loadavg stays between 2.00-4.00 on a Quad-core Intel core 2 Quad processor with 4GB RAM. I have three systems like this in place at different call centers and the load is consistent for all three of them. Usually we put less load on a single server, but these were inbound-only scenarios which is less load than outbound. MATT--- On 6/8/08, Steve Totaro <[EMAIL PROTECTED]> wrote: > Matt, > > Could you share the CPU usage, memory, and load average in the > scenario you describe? What type of ULAW channels > (SIP,DAHDI,IAX), or does it not matter? > > Thanks, > > Steve Totaro > > > On Sun, Jun 8, 2008 at 5:29 PM, Matt Florell <[EMAIL PROTECTED]> wrote: > > Hello, > > > > We routinely run meetme with over 140 ULAW channels connected to 70 > > meetme rooms with no issues on an Intel Core 2 Quad core CPU. > > > > The major factor in capacity would be your CPU and RAM capacity. If > > you have at least a base-level P4 you don't need to worry about 12 > > participants. > > > > MATT--- > > > > On 6/8/08, Adrian Marsh <[EMAIL PROTECTED]> wrote: > >> I've got to agree.. I've never given it much thought either... > >> > >> All of my calls are SIP/IAX based, coming in from the PSTN from a peer > >> like that too.. > >> > >> I've never tracked the total number of conference users... But I'll bet > >> we've hit at least 10.. And I've never seen the CPU go above 10%.. And > >> that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it > >> will be setup-specific.. So I would look at your CPU and memory stats, > >> and run some tests and monitor that.. > >> > >> > >> A. > >> > >> > >> -Original Message- > >> From: [EMAIL PROTECTED] > >> [mailto:[EMAIL PROTECTED] On Behalf Of John > >> covici > >> Sent: 08 June 2008 16:34 > >> To: Asterisk Users Mailing List - Non-Commercial Discussion > >> Subject: Re: [asterisk-users] MeetMe Limits > >> > >> 12 people is nothing -- I do 20 regularly -- however you may want to > >> have them come in as muted or tell them to mute themselves, because the > >> latency can cause very severe echoes if they are on a speaker phone or > >> cell phone. > >> > >> on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote > Actually I think > >> they will all be calling in using regular pstn phones > and cell > >> phones. > >> > > >> > Sam > >> > > >> > Al Baker wrote: > >> > > The 2 big questions are: > >> > > -Are all participants using QoS end to end ? > >> > > > >> > > -Are all of them using the SAME CODEC. As the amount of Transcoding > >> goes up, > > the work on the * box goes up and can be a problem. > >> > > > >> > > Sam wrote: > >> > >> I am thinking about using my asterisk server to host a conference > >> with > >> about 12 other people from around the USA. Bandwidth issues > >> aside, will > >> this work or will all the different latencies cause > >> issues? Yea I know, > >> I could just "try it and find out" but it is > >> going to take alot of time > >> to get everyones schedule to line up, I > >> don't want to go through the > >> trouble if I will just be > >> disappointed. > >> > >> > >> > >> Thanks, > >> > >> > >> > >> Sam > >> > >> > >> > >> ___ > >> > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com > >> -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or > >> update options visit: > >> > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > >> > >> > >> > >> > >> > > > >> > > ___ > >> > > -
Re: [asterisk-users] MeetMe Limits
Matt, Could you share the CPU usage, memory, and load average in the scenario you describe? What type of ULAW channels (SIP,DAHDI,IAX), or does it not matter? Thanks, Steve Totaro On Sun, Jun 8, 2008 at 5:29 PM, Matt Florell <[EMAIL PROTECTED]> wrote: > Hello, > > We routinely run meetme with over 140 ULAW channels connected to 70 > meetme rooms with no issues on an Intel Core 2 Quad core CPU. > > The major factor in capacity would be your CPU and RAM capacity. If > you have at least a base-level P4 you don't need to worry about 12 > participants. > > MATT--- > > On 6/8/08, Adrian Marsh <[EMAIL PROTECTED]> wrote: >> I've got to agree.. I've never given it much thought either... >> >> All of my calls are SIP/IAX based, coming in from the PSTN from a peer >> like that too.. >> >> I've never tracked the total number of conference users... But I'll bet >> we've hit at least 10.. And I've never seen the CPU go above 10%.. And >> that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it >> will be setup-specific.. So I would look at your CPU and memory stats, >> and run some tests and monitor that.. >> >> >> A. >> >> >> -Original Message- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of John >> covici >> Sent: 08 June 2008 16:34 >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] MeetMe Limits >> >> 12 people is nothing -- I do 20 regularly -- however you may want to >> have them come in as muted or tell them to mute themselves, because the >> latency can cause very severe echoes if they are on a speaker phone or >> cell phone. >> >> on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote > Actually I think >> they will all be calling in using regular pstn phones > and cell >> phones. >> > >> > Sam >> > >> > Al Baker wrote: >> > > The 2 big questions are: >> > > -Are all participants using QoS end to end ? >> > > >> > > -Are all of them using the SAME CODEC. As the amount of Transcoding >> goes up, > > the work on the * box goes up and can be a problem. >> > > >> > > Sam wrote: >> > >> I am thinking about using my asterisk server to host a conference >> with > >> about 12 other people from around the USA. Bandwidth issues >> aside, will > >> this work or will all the different latencies cause >> issues? Yea I know, > >> I could just "try it and find out" but it is >> going to take alot of time > >> to get everyones schedule to line up, I >> don't want to go through the > >> trouble if I will just be >> disappointed. >> > >> >> > >> Thanks, >> > >> >> > >> Sam >> > >> >> > >> ___ >> > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com >> -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or >> update options visit: >> > >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> >> > >> >> > >> >> > > >> > > ___ >> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com >> -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update >> options visit: >> > >http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> > >> > ___ >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> > > asterisk-users mailing list > To UNSUBSCRIBE or update options >> visit: >> >http://lists.digium.com/mailman/listinfo/asterisk-users >> >> -- >> Your life is like a penny. You're going to lose it. The question is: >> How do >> you spend it? >> >> John Covici >> [EMAIL PROTECTED] >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
Hello, We routinely run meetme with over 140 ULAW channels connected to 70 meetme rooms with no issues on an Intel Core 2 Quad core CPU. The major factor in capacity would be your CPU and RAM capacity. If you have at least a base-level P4 you don't need to worry about 12 participants. MATT--- On 6/8/08, Adrian Marsh <[EMAIL PROTECTED]> wrote: > I've got to agree.. I've never given it much thought either... > > All of my calls are SIP/IAX based, coming in from the PSTN from a peer > like that too.. > > I've never tracked the total number of conference users... But I'll bet > we've hit at least 10.. And I've never seen the CPU go above 10%.. And > that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it > will be setup-specific.. So I would look at your CPU and memory stats, > and run some tests and monitor that.. > > > A. > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of John > covici > Sent: 08 June 2008 16:34 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] MeetMe Limits > > 12 people is nothing -- I do 20 regularly -- however you may want to > have them come in as muted or tell them to mute themselves, because the > latency can cause very severe echoes if they are on a speaker phone or > cell phone. > > on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote > Actually I think > they will all be calling in using regular pstn phones > and cell > phones. > > > > Sam > > > > Al Baker wrote: > > > The 2 big questions are: > > > -Are all participants using QoS end to end ? > > > > > > -Are all of them using the SAME CODEC. As the amount of Transcoding > goes up, > > the work on the * box goes up and can be a problem. > > > > > > Sam wrote: > > >> I am thinking about using my asterisk server to host a conference > with > >> about 12 other people from around the USA. Bandwidth issues > aside, will > >> this work or will all the different latencies cause > issues? Yea I know, > >> I could just "try it and find out" but it is > going to take alot of time > >> to get everyones schedule to line up, I > don't want to go through the > >> trouble if I will just be > disappointed. > > >> > > >> Thanks, > > >> > > >> Sam > > >> > > >> ___ > > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com > -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or > update options visit: > > >>http://lists.digium.com/mailman/listinfo/asterisk-users > > >> > > >> > > >> > > > > > > ___ > > > -- Bandwidth and Colocation Provided by http://www.api-digital.com > -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update > options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > asterisk-users mailing list > To UNSUBSCRIBE or update options > visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > [EMAIL PROTECTED] > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
I've got to agree.. I've never given it much thought either... All of my calls are SIP/IAX based, coming in from the PSTN from a peer like that too.. I've never tracked the total number of conference users... But I'll bet we've hit at least 10.. And I've never seen the CPU go above 10%.. And that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it will be setup-specific.. So I would look at your CPU and memory stats, and run some tests and monitor that.. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: 08 June 2008 16:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe Limits 12 people is nothing -- I do 20 regularly -- however you may want to have them come in as muted or tell them to mute themselves, because the latency can cause very severe echoes if they are on a speaker phone or cell phone. on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote > Actually I think they will all be calling in using regular pstn phones > and cell phones. > > Sam > > Al Baker wrote: > > The 2 big questions are: > > -Are all participants using QoS end to end ? > > > > -Are all of them using the SAME CODEC. As the amount of Transcoding goes up, > > the work on the * box goes up and can be a problem. > > > > Sam wrote: > >> I am thinking about using my asterisk server to host a conference with > >> about 12 other people from around the USA. Bandwidth issues aside, will > >> this work or will all the different latencies cause issues? Yea I know, > >> I could just "try it and find out" but it is going to take alot of time > >> to get everyones schedule to line up, I don't want to go through the > >> trouble if I will just be disappointed. > >> > >> Thanks, > >> > >> Sam > >> > >> ___ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > >> > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
12 people is nothing -- I do 20 regularly -- however you may want to have them come in as muted or tell them to mute themselves, because the latency can cause very severe echoes if they are on a speaker phone or cell phone. on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote > Actually I think they will all be calling in using regular pstn phones > and cell phones. > > Sam > > Al Baker wrote: > > The 2 big questions are: > > -Are all participants using QoS end to end ? > > > > -Are all of them using the SAME CODEC. As the amount of Transcoding goes > > up, > > the work on the * box goes up and can be a problem. > > > > Sam wrote: > >> I am thinking about using my asterisk server to host a conference with > >> about 12 other people from around the USA. Bandwidth issues aside, will > >> this work or will all the different latencies cause issues? Yea I know, > >> I could just "try it and find out" but it is going to take alot of time > >> to get everyones schedule to line up, I don't want to go through the > >> trouble if I will just be disappointed. > >> > >> Thanks, > >> > >> Sam > >> > >> ___ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > >> > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
Actually I think they will all be calling in using regular pstn phones and cell phones. Sam Al Baker wrote: > The 2 big questions are: > -Are all participants using QoS end to end ? > > -Are all of them using the SAME CODEC. As the amount of Transcoding goes up, > the work on the * box goes up and can be a problem. > > Sam wrote: >> I am thinking about using my asterisk server to host a conference with >> about 12 other people from around the USA. Bandwidth issues aside, will >> this work or will all the different latencies cause issues? Yea I know, >> I could just "try it and find out" but it is going to take alot of time >> to get everyones schedule to line up, I don't want to go through the >> trouble if I will just be disappointed. >> >> Thanks, >> >> Sam >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up, the work on the * box goes up and can be a problem. Sam wrote: > I am thinking about using my asterisk server to host a conference with > about 12 other people from around the USA. Bandwidth issues aside, will > this work or will all the different latencies cause issues? Yea I know, > I could just "try it and find out" but it is going to take alot of time > to get everyones schedule to line up, I don't want to go through the > trouble if I will just be disappointed. > > Thanks, > > Sam > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe Limits
I am thinking about using my asterisk server to host a conference with about 12 other people from around the USA. Bandwidth issues aside, will this work or will all the different latencies cause issues? Yea I know, I could just "try it and find out" but it is going to take alot of time to get everyones schedule to line up, I don't want to go through the trouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
