Re: [on-asterisk] Conference bridge
Gentlemen, let me run this one by you
A dtmf digit is used in a push to talk mode could be a novel solution. If you
want to join the I wish to
talk next queue press # and as soon as the last person has finished talking
and toggled the # digit
the next person has the floor. This solution would only require the setting
up of a channel queue on a
first in - first out basis. Obviously some bugs would need to be ironed out but
it does follow the KISS
principle.
Uncle Henry
=
Henry L.Coleman [www.VoIP-PBX.ca]
Tel: 647-723-5160 Ext.203
=
{ D. Hugh Redelmeier}
I should say that I'm approaching this from first principles and not
from any practical knowledge. So this is probably not of interest to
Rachel. But I do find the problem interesting.
| From: Jim Van Meggelen j...@vanmeggelen.ca
| If memory serves correctly, the conference mixer doesn't have to mix all
| incoming audio, but rather only has to mix relevant audio (i.e. figure out
| who's talking, and take that single audio stream and send it out to all the
| participating channels). One challenge I would expect would be figuring out
| the noise threshold (i.e. what is talking and what is just background
noise),
| and knowing to quickly enable a channel when somebody is speaking. A good
| mixer should be able to handle more than one person speaking, but since for
| the most part people can only handle one person talking at a time, if the
| mixer is good, it doesn't have to work so hard at that.
You also asked whether the problem was to handle M conferences of M
people (where perhaps M * N = 1000) or one conference of N people
(where N = 1000).
A very good question. In a face to face conference, people behave
differently as the number of participants increases. In particular,
speaker selection gets to be more and more formal because the problem
gets harder to solve.
Things don't get easier with telephone conferencing:
- some out of band signals are lost
- eye contact, gaze
- standing, sticking hand in the air
- designation by chairperson
- leaning over and whispering to a neighbour
- some signals are degraded
- only some frequencies are carried and the accuracy is reduced
- dynamic range is reduced (speaking up works in real conferences
but not nearly as well over a phone)
- even modest time delays confuse informal conversational protocols
- (with current systems) localization clues/cues are lost. The human ear
can tell (with some ambiguity) where a sound comes from. This turns
out to help quite a bit in understanding what is going on with
several auditory things going on at once.
I don't immediately see how a largish conference can be run as
anything other than broadcasting by a single speaker or a small number
of speakers designated manually.
As a thought experiment, consider how one can hear a speaker in a lecture
even over coughing. I don't see that working in a telephone
conference with all mikes open.
| I suspect the math involved is pretty complex, though.
Math I can handle (perhaps). What I don't know are the practical
considerations. The psycho-acoustics are not obvious.
| This also gets me wondering if multiple, discreet conferences eat up more
| horsepower than a single conference would, even with a large number of
| participants.
I imagine that small conferences would be more amenable to automatic
solutions and hence could take more processing (per participant) than
large conferences when simple designation must be used.
I have no idea what the thresholds would be. I don't even know how
many different strategies there would be (i.e. how many thresholds).
| I suspect there's a lot more to it than that, though.
Agreed.
-
To unsubscribe, e-mail: asterisk-unsubscr...@uc.org
For additional commands, e-mail: asterisk-h...@uc.org
-
To unsubscribe, e-mail: asterisk-unsubscr...@uc.org
For additional commands, e-mail: asterisk-h...@uc.org
Re: [on-asterisk] Conference bridge
I should say that I'm approaching this from first principles and not from any practical knowledge. So this is probably not of interest to Rachel. But I do find the problem interesting. | From: Jim Van Meggelen j...@vanmeggelen.ca | If memory serves correctly, the conference mixer doesn't have to mix all | incoming audio, but rather only has to mix relevant audio (i.e. figure out | who's talking, and take that single audio stream and send it out to all the | participating channels). One challenge I would expect would be figuring out | the noise threshold (i.e. what is talking and what is just background noise), | and knowing to quickly enable a channel when somebody is speaking. A good | mixer should be able to handle more than one person speaking, but since for | the most part people can only handle one person talking at a time, if the | mixer is good, it doesn't have to work so hard at that. You also asked whether the problem was to handle M conferences of M people (where perhaps M * N = 1000) or one conference of N people (where N = 1000). A very good question. In a face to face conference, people behave differently as the number of participants increases. In particular, speaker selection gets to be more and more formal because the problem gets harder to solve. Things don't get easier with telephone conferencing: - some out of band signals are lost - eye contact, gaze - standing, sticking hand in the air - designation by chairperson - leaning over and whispering to a neighbour - some signals are degraded - only some frequencies are carried and the accuracy is reduced - dynamic range is reduced (speaking up works in real conferences but not nearly as well over a phone) - even modest time delays confuse informal conversational protocols - (with current systems) localization clues/cues are lost. The human ear can tell (with some ambiguity) where a sound comes from. This turns out to help quite a bit in understanding what is going on with several auditory things going on at once. I don't immediately see how a largish conference can be run as anything other than broadcasting by a single speaker or a small number of speakers designated manually. As a thought experiment, consider how one can hear a speaker in a lecture even over coughing. I don't see that working in a telephone conference with all mikes open. | I suspect the math involved is pretty complex, though. Math I can handle (perhaps). What I don't know are the practical considerations. The psycho-acoustics are not obvious. | This also gets me wondering if multiple, discreet conferences eat up more | horsepower than a single conference would, even with a large number of | participants. I imagine that small conferences would be more amenable to automatic solutions and hence could take more processing (per participant) than large conferences when simple designation must be used. I have no idea what the thresholds would be. I don't even know how many different strategies there would be (i.e. how many thresholds). | I suspect there's a lot more to it than that, though. Agreed. - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
Re: [on-asterisk] Conference bridge
On Sat, Feb 28, 2009 at 2:27 AM, Jim Van Meggelen j...@vanmeggelen.ca wrote: Dave Donovan wrote: Personally, I'd be less interested in which processors do what than seeing how things scale with processor power. You mean how things behave at the limit? No, I'm thinking about if 2GB ram and 4 Ghz gives me 150 conference channels, does 4GB and 8Ghz give me 300 channels or more, or less? The advantage of using a VM host is that you could adjust processor and memory with a few key clicks and re-run your automated test. Are there still significant timing issues with Vmware? If that can be overcome there are some serious advantages to using it as a test platform. You can dial the CPU and memory up and down on demand. Would that be beneficial or would it just muddy the waters? It would both muddy the waters and be beneficial :-p It would certainly be interesting, but the more I think of it, the more I think that we're as likely to to be limited by some quirk of the virtualization platform than of the software we're actually testing. It looks like Bill has a good platform that would get to the heart of the original question which I understand as Is it currently possible to use a general purpose processor in a large scale conferencing bridge approaching 1000 channels, or is hardware DSP required? On that note, there should be some agreement on exactly what premise or configurations are to be tested/proven. Dave - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Conference bridge
| From: Rachel Quin rac...@beanfield.com | | I think I'm not making myself clear, sorry. Our t3's and Megalink circuit | from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A | conferencing server would only handle RTP streams, mixing channels for many | large-ish volume conferences. The box I'm talking about would have 2 10gig | nics, one or two DSP cards, and whatever software is needed to handle | managing conferencing and directing RTP/G.711 content channels to and from | the DSP card(s). I am not looking to build a stand alone phone system. Naively, I would think mixing RTP streams of G.711 should not be too hard for a regular CPU. G.711 is PCM so decoding and encoding is a snap. Mixing is just a kind of averaging, I imagine. But: I did say naively. I've never done any of this. I don't know whether automatic gain control can be done simply and cheaply. I don't know how you can sum a hundred channels and not get overloaded with noise. I'll waive my hands and say that different channels don't need to be transformed to use the same timebase, but maybe I'm wrong. I know nothing about echo-cancellation issues. So, naively, the tasks of the processor would be: - take samples from N RTP streams - average them - send the result out on N RTP streams. The actual amount of computation, for the naive process, ought to be within the realm of any modern processor for values of N up to perhaps 1000. 8K samples / channel / second == 8KB bandwidth / sec modern processors can do (guess) 40MW main memory accesses / second (the bottleneck, I think) Which of the things that I've skipped are necessary and expensive? - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
Re: [on-asterisk] Conference bridge
Dave Donovan wrote: [snip] It looks like Bill has a good platform that would get to the heart of the original question which I understand as Is it currently possible to use a general purpose processor in a large scale conferencing bridge approaching 1000 channels, or is hardware DSP required? Which raised another interesting question: is there a difference between 10 conferences with 10 particpants each, vs 1 conference of 100, or 4 of 25, or whatever? Point being: does it matter if we can support 1000 participants, if nobody will ever need a conference that large? On that note, there should be some agreement on exactly what premise or configurations are to be tested/proven. That's for sure what we'd want to do when we define the scope of the test. In a way it doesn't matter as long as we're consistent. Some of this might boil down to what can be done in a reasonable length of time. -- Jim Van Meggelen j...@vanmeggelen.ca http://www.oreillynet.com/pub/au/2177 A child is the ultimate startup, and I have three. This makes me rich. Guy Kawasaki -- - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
Re: [on-asterisk] Conference bridge
If memory serves correctly, the conference mixer doesn't have to mix all incoming audio, but rather only has to mix relevant audio (i.e. figure out who's talking, and take that single audio stream and send it out to all the participating channels). One challenge I would expect would be figuring out the noise threshold (i.e. what is talking and what is just background noise), and knowing to quickly enable a channel when somebody is speaking. A good mixer should be able to handle more than one person speaking, but since for the most part people can only handle one person talking at a time, if the mixer is good, it doesn't have to work so hard at that. I suspect the math involved is pretty complex, though. This also gets me wondering if multiple, discreet conferences eat up more horsepower than a single conference would, even with a large number of participants. I suspect there's a lot more to it than that, though. Jim D. Hugh Redelmeier wrote: | From: Rachel Quin rac...@beanfield.com | | I think I'm not making myself clear, sorry. Our t3's and Megalink circuit | from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A | conferencing server would only handle RTP streams, mixing channels for many | large-ish volume conferences. The box I'm talking about would have 2 10gig | nics, one or two DSP cards, and whatever software is needed to handle | managing conferencing and directing RTP/G.711 content channels to and from | the DSP card(s). I am not looking to build a stand alone phone system. Naively, I would think mixing RTP streams of G.711 should not be too hard for a regular CPU. G.711 is PCM so decoding and encoding is a snap. Mixing is just a kind of averaging, I imagine. But: I did say naively. I've never done any of this. I don't know whether automatic gain control can be done simply and cheaply. I don't know how you can sum a hundred channels and not get overloaded with noise. I'll waive my hands and say that different channels don't need to be transformed to use the same timebase, but maybe I'm wrong. I know nothing about echo-cancellation issues. So, naively, the tasks of the processor would be: - take samples from N RTP streams - average them - send the result out on N RTP streams. The actual amount of computation, for the naive process, ought to be within the realm of any modern processor for values of N up to perhaps 1000. 8K samples / channel / second == 8KB bandwidth / sec modern processors can do (guess) 40MW main memory accesses / second (the bottleneck, I think) Which of the things that I've skipped are necessary and expensive? - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org -- -- Jim Van Meggelen j...@vanmeggelen.ca http://www.oreillynet.com/pub/au/2177 A child is the ultimate startup, and I have three. This makes me rich. Guy Kawasaki -- - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
Re: [on-asterisk] Conference bridge
Rachel, In my opinion freeswitch has the best base conference bride features, no dependency on hardware or the ztdummy timer and loads more features. For a comparison of the FS Asterisk features here is link to a comparison http://www.freeswitch.org/node/100 Also here is a small article ( http://www.junctionnetworks.com/blog/charlotte/2008/05/21/freeswitch-asterisk-replacement ) and their rational of picking FS over Asterisk for their conference bridge product. Hope this helps, Mike Rachel Quin wrote: I want to build a conference bridge using dedicated DSP hardware, running on FreeBSD. Does anyone have recomendations on HW/SW? Rachel Quin Beanfield Metroconnect audace fortuna iuvat - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org -- Mike Ashton Quality Track Intl CTO Ph: 647-724-3500 x 301 Cell: 416-527-4995 Fax:416-352-6043 QTI CONFIDENTIAL AND PROPRIETARY INFORMATION The contents of this material are confidential and proprietary to Quality Track International, Inc. and may not be reproduced, disclosed, distributed or used without the express permission of an authorized representative of QTI. Use for any purpose or in any manner other than that expressly authorized is prohibited. If you have received this communication in error, please immediately delete it and all copies, and promptly notify the sender. - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
Re: [on-asterisk] Conference bridge
Digium cards work well Rachel Quin wrote: I want to build a conference bridge using dedicated DSP hardware, running on FreeBSD. Does anyone have recomendations on HW/SW? Rachel Quin Beanfield Metroconnect audace fortuna iuvat - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Conference bridge
I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield Metroconnect audace fortuna iuvat _ From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 9:23 AM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, In my opinion freeswitch has the best base conference bride features, no dependency on hardware or the ztdummy timer and loads more features. For a comparison of the FS Asterisk features here is link to a comparison http://www.freeswitch.org/node/100 Also here is a small article ( http://www.junctionnetworks.com/blog/charlotte/2008/05/21/freeswitch-asteris k-replacement ) and their rational of picking FS over Asterisk for their conference bridge product. Hope this helps, Mike Rachel Quin wrote: I want to build a conference bridge using dedicated DSP hardware, running on FreeBSD. Does anyone have recomendations on HW/SW? Rachel Quin Beanfield Metroconnect audace fortuna iuvat - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org -- Mike Ashton Quality Track Intl CTO Ph: 647-724-3500 x 301 Cell: 416-527-4995 Fax:416-352-6043 QTI CONFIDENTIAL AND PROPRIETARY INFORMATION The contents of this material are confidential and proprietary to Quality Track International, Inc. and may not be reproduced, disclosed, distributed or used without the express permission of an authorized representative of QTI. Use for any purpose or in any manner other than that expressly authorized is prohibited. If you have received this communication in error, please immediately delete it and all copies, and promptly notify the sender. No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.3/1971 - Release Date: 02/27/09 07:05:00
Re: [on-asterisk] Conference bridge
Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield Metroconnect audace fortuna iuvat _ From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 9:23 AM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, In my opinion freeswitch has the best base conference bride features, no dependency on hardware or the ztdummy timer and loads more features. For a comparison of the FS Asterisk features here is link to a comparison http://www.freeswitch.org/node/100 Also here is a small article ( http://www.junctionnetworks.com/blog/charlotte/2008/05/21/freeswitch-asteris k-replacement ) and their rational of picking FS over Asterisk for their conference bridge product. Hope this helps, Mike Rachel Quin wrote: I want to build a conference bridge using dedicated DSP hardware, running on FreeBSD. Does anyone have recomendations on HW/SW? Rachel Quin Beanfield Metroconnect audace fortuna iuvat - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Conference bridge
Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield Metroconnect audace fortuna iuvat _ From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 9:23 AM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, In my opinion freeswitch has the best base conference bride features, no dependency on hardware or the ztdummy timer and loads more features. For a comparison of the FS Asterisk features here is link to a comparison http://www.freeswitch.org/node/100 Also here is a small article ( http://www.junctionnetworks.com/blog/charlotte/2008/05/21/freeswitch-asteris k-replacement ) and their rational of picking FS over Asterisk for their conference bridge product. Hope this helps, Mike Rachel Quin wrote: I want to build a conference bridge using dedicated DSP hardware, running on FreeBSD. Does anyone have recomendations on HW/SW? Rachel Quin Beanfield Metroconnect audace fortuna iuvat - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.3/1971 - Release Date: 02/27/09 07:05:00 - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
Re: [on-asterisk] Conference bridge
asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield Metroconnect audace fortuna iuvat _ From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 9:23 AM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, In my opinion freeswitch has the best base conference bride features, no dependency on hardware or the ztdummy timer and loads more features. For a comparison of the FS Asterisk features here is link to a comparison http://www.freeswitch.org/node/100 Also here is a small article ( http://www.junctionnetworks.com/blog/charlotte/2008/05/21/freeswitch-asteris k-replacement ) and their rational of picking FS over Asterisk for their conference bridge product. Hope this helps, Mike Rachel Quin wrote: I want to build a conference bridge using dedicated DSP hardware, running on FreeBSD. Does anyone have recomendations on HW/SW? Rachel Quin Beanfield Metroconnect audace fortuna iuvat - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.3/1971 - Release Date: 02/27/09 07:05:00 - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
Re: [on-asterisk] Conference bridge
Rachel, You could go down the road of using DSP cards but I'm not sure you really need the complexity. What sort of capacity/features are you looking for? When you say media mixing, do you mean things like: -injecting background music, or a prerecorded audio stream into the conference? If so FS can do it. - adjust individual channel or the conference volume? yup - background noise reduction? yup I've heard of freeswitch handling 500 channels on a standard server box. It has a full feature list. Even sipX is merging freeswitch in, starting with conferencing. http://sipx-wiki.calivia.com/index.php/Conferenceing_Service_for_sipXecs SipX also has a gui front end to manage the freeswitch conference bridge, so might be a easy way to test it out and it also offers an easy installer which can also configure a high availability system. http://sipx-wiki.calivia.com/index.php/CD_Installation_of_sipXecs Mike Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield Metroconnect audace fortuna iuvat _ From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 9:23 AM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, In my opinion freeswitch has the best base conference bride features, no dependency on hardware or the ztdummy timer and loads more features. For a comparison of the FS Asterisk features here is link to a comparison http://www.freeswitch.org/node/100 Also here is a small article ( http://www.junctionnetworks.com/blog/charlotte/2008/05/21/freeswitch-asteris k-replacement ) and their rational of picking FS over Asterisk for their conference bridge product. Hope this helps, Mike Rachel Quin wrote: I want to build a conference bridge using dedicated DSP hardware, running on FreeBSD. Does anyone have recomendations on HW/SW? Rachel Quin Beanfield Metroconnect audace fortuna iuvat - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.3/1971 - Release Date: 02/27/09 07:05:00 - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org -- Mike Ashton Quality Track Intl CTO Ph: 647-724-3500 x 301 Cell: 416-527-4995 Fax:416-352-6043 QTI CONFIDENTIAL AND PROPRIETARY INFORMATION The contents of this material are confidential and proprietary to Quality Track International, Inc. and may not be reproduced, disclosed, distributed or used without the express permission of an authorized representative of QTI. Use for any purpose or in any manner other than that expressly authorized is prohibited. If you have received this communication in error, please immediately delete it and all copies
RE: [on-asterisk] Conference bridge
I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield Metroconnect audace fortuna iuvat _ From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 9:23 AM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, In my opinion freeswitch has the best base conference bride features, no dependency on hardware or the ztdummy timer and loads more features. For a comparison of the FS Asterisk features here is link to a comparison http://www.freeswitch.org/node/100 Also here is a small article ( http://www.junctionnetworks.com/blog/charlotte/2008/05/21/freeswitch-asteris k-replacement ) and their rational of picking FS over Asterisk for their conference bridge product. Hope this helps, Mike Rachel Quin wrote: I want to build a conference bridge using dedicated DSP hardware, running on FreeBSD. Does anyone have recomendations on HW/SW? Rachel Quin Beanfield Metroconnect audace fortuna iuvat - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.3/1971 - Release Date: 02/27/09 07:05:00 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.3/1971 - Release Date: 02/27/09 07:05:00 - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
RE: [on-asterisk] Conference bridge
What you're saying is that if I have 20 active conferences, of 25 G.711 channels each, a generic two dual-core Xeon server can do all the mixing? Four Xeon cores can mix that load, and do everything else? Wow, has anyone got anything like that working? I would love to see it. I mean, our AS5400 fully loaded, with all the DSP card slots occupied, can only handle assembling 675 DS0's. Even our Convedia media server is stuffed full of DSPs. Rachel From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 11:32 AM To: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, You could go down the road of using DSP cards but I'm not sure you really need the complexity. What sort of capacity/features are you looking for? When you say media mixing, do you mean things like: -injecting background music, or a prerecorded audio stream into the conference? If so FS can do it. - adjust individual channel or the conference volume? yup - background noise reduction? yup I've heard of freeswitch handling 500 channels on a standard server box. It has a full feature list. Even sipX is merging freeswitch in, starting with conferencing. http://sipx-wiki.calivia.com/index.php/Conferenceing_Service_for_sipXecs SipX also has a gui front end to manage the freeswitch conference bridge, so might be a easy way to test it out and it also offers an easy installer which can also configure a high availability system. http://sipx-wiki.calivia.com/index.php/CD_Installation_of_sipXecs Mike Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield Metroconnect audace fortuna iuvat _ From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 9:23 AM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, In my opinion freeswitch has the best base conference bride features, no dependency on hardware or the ztdummy timer and loads more features. For a comparison of the FS Asterisk features here is link to a comparison http://www.freeswitch.org/node/100 Also here is a small article ( http://www.junctionnetworks.com/blog/charlotte/2008/05/21/freeswitch-asteris k-replacement ) and their rational of picking FS over Asterisk for their conference bridge product. Hope this helps, Mike Rachel Quin wrote: I want to build a conference bridge using dedicated DSP hardware, running on FreeBSD. Does anyone have recomendations on HW/SW? Rachel Quin Beanfield Metroconnect audace fortuna iuvat - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.3/1971 - Release Date: 02/27/09 07:05:00 - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands
Re: [on-asterisk] Conference bridge
gotcha, in that case take a gander at the digium tc400b card, that handles dsp offloading for asterisk and also acts as a transcoding accelerator. Rachel Quin wrote: I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield Metroconnect audace fortuna iuvat _ From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 9:23 AM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, In my opinion freeswitch has the best base conference bride features, no dependency on hardware or the ztdummy timer and loads more features. For a comparison of the FS Asterisk features here is link to a comparison http://www.freeswitch.org/node/100 Also here is a small article ( http://www.junctionnetworks.com/blog/charlotte/2008/05/21/freeswitch-asteris k-replacement ) and their rational of picking FS over Asterisk for their conference bridge product. Hope this helps, Mike Rachel Quin wrote: I want to build a conference bridge using dedicated DSP hardware, running on FreeBSD. Does anyone have recomendations on HW/SW? Rachel Quin Beanfield Metroconnect audace fortuna iuvat - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.3/1971 - Release Date: 02/27/09 07:05:00 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.3/1971 - Release Date: 02/27/09 07:05:00 - To unsubscribe, e-mail: asterisk-unsubscr...@uc.org For additional commands, e-mail: asterisk-h...@uc.org
Re: [on-asterisk] Conference bridge
yes, but you need to buy the digium dsp card (from what ive been told) Rachel Quin wrote: No, that flexibility is exactly what I'm looking for, but you simply can't mix that many G.711 channels in Xeon cores. My question is, does anyone know of any open source software that will utilize DSP cards for the actual voice stream crunching of G.711 channels? All of the signalling and management function would be in the software running on the host hardware. Every Telco grade media mixer does this, every edge T3 or OC gateway. Can FreeSwitch or Asterisk do all the conferencing work, using DSP hardware offloading? Rachel From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 12:03 PM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, I think you may have a misconception of what Asterisk ad/or FreeSwitch are. They are really telephony/media software platforms that can be configured to do many things. The most frequent uses are as full blown PBX phone systems, but they can be used strictly as, a VM platform, an IVR application server, media gateways, etc. Mike Rachel Quin wrote: I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield Metroconnect audace fortuna iuvat _ From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 9:23 AM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, In my opinion freeswitch has the best base conference bride features, no dependency on hardware or the ztdummy timer and loads more features. For a comparison of the FS Asterisk features here is link to a comparison http://www.freeswitch.org/node/100 Also here is a small article ( http://www.junctionnetworks.com/blog/charlotte/2008/05/21/freeswitch-asteris k-replacement ) and their rational of picking FS over Asterisk for their conference bridge product. Hope this helps, Mike Rachel Quin wrote: I want to build a conference bridge using dedicated DSP hardware, running on FreeBSD. Does anyone have recomendations on HW/SW? Rachel Quin Beanfield Metroconnect audace fortuna iuvat
Re: [on-asterisk] Conference bridge
The TC400B seems indeed a good fit: The TC400B decompresses G.729a (8.0kbit) or G.723.1 (5.3kbit) into u-law or a-law; or, compresses u-law or a-law into G.729a (8.0kbit) or G.723.1 (5.3kbit). The TC400B is rated to handle up to 120 bi-directional G.729a transformations or 92 bi-directional G.723.1 transformations. The TC400B does not require additional licensing fees for the use of these codecs nor does it require the registration process associated with Digium's software-based G.729a codec licensing. Quoting Philip Mullis philip.mul...@syx.ca: yes, but you need to buy the digium dsp card (from what ive been told) Rachel Quin wrote: No, that flexibility is exactly what I'm looking for, but you simply can't mix that many G.711 channels in Xeon cores. My question is, does anyone know of any open source software that will utilize DSP cards for the actual voice stream crunching of G.711 channels? All of the signalling and management function would be in the software running on the host hardware. Every Telco grade media mixer does this, every edge T3 or OC gateway. Can FreeSwitch or Asterisk do all the conferencing work, using DSP hardware offloading? Rachel From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 12:03 PM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, I think you may have a misconception of what Asterisk ad/or FreeSwitch are. They are really telephony/media software platforms that can be configured to do many things. The most frequent uses are as full blown PBX phone systems, but they can be used strictly as, a VM platform, an IVR application server, media gateways, etc. Mike Rachel Quin wrote: I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield Metroconnect audace fortuna iuvat _ From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 9:23 AM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, In my opinion freeswitch has the best base conference bride features, no dependency on hardware or the ztdummy timer and loads more features. For a comparison of the FS Asterisk features here is link to a comparison http://www.freeswitch.org/node/100 Also here
RE: [on-asterisk] Conference bridge
Ok, almost there, but not quite. I don't want to do dedicated codec conversion. In fact, we will never use anything other than G.711. I'm looking for a more generic DSP card that can be used to do all the channel mixing, offloading the work from the server's processors. And!! The software that can utilize it. I can find the cards easy enough, there are quite a few to choose from, but of the OSS, I know not. To quote from the Digium page: These transformations in software are very expensive, in terms of MIPS, and require a substantial amount of CPU time to accomplish. Channel mixing also quite expensive, when you're talking about the high hundreds of channels. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 12:04 PM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge gotcha, in that case take a gander at the digium tc400b card, that handles dsp offloading for asterisk and also acts as a transcoding accelerator. Rachel Quin wrote: I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield Metroconnect audace fortuna iuvat _ From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 9:23 AM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, In my opinion freeswitch has the best base conference bride features, no dependency on hardware or the ztdummy timer and loads more features. For a comparison of the FS Asterisk features here is link to a comparison http://www.freeswitch.org/node/100 Also here is a small article ( http://www.junctionnetworks.com/blog/charlotte/2008/05/21/freeswitch-asteris k-replacement ) and their rational of picking FS over Asterisk for their conference bridge product. Hope this helps, Mike Rachel Quin wrote: I want to build a conference bridge using dedicated DSP hardware, running on FreeBSD. Does anyone have recomendations on HW/SW? Rachel Quin Beanfield Metroconnect audace fortuna iuvat
Re: [on-asterisk] Conference bridge
Rachel, I don't have any experience utilizing dsp cards for off loading, and do not personally have loads like this, our server usually has about 20 3 channel conferences running. From the freeswitch developers : The conference is more resource intensive than normal bridging but the general rule for media channels is about 190 channels (95 bridges) per 1 gigahertz of CPU on a 64 bit platform. If you don't need to run media into FS you can bet on a lot more. If you look at this thread you'll get some good information on it's capabilities. http://lists.freeswitch.org/pipermail/freeswitch-users/2008-May/003397.html Mike Rachel Quin wrote: What you're saying is that if I have 20 active conferences, of 25 G.711 channels each, a generic two dual-core Xeon server can do all the mixing? Four Xeon cores can mix that load, and do everything else? Wow, has anyone got anything like that working? I would love to see it. I mean, our AS5400 fully loaded, with all the DSP card slots occupied, can only handle assembling 675 DS0's. Even our Convedia media server is stuffed full of DSPs. Rachel From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 11:32 AM To: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, You could go down the road of using DSP cards but I'm not sure you really need the complexity. What sort of capacity/features are you looking for? When you say media mixing, do you mean things like: -injecting background music, or a prerecorded audio stream into the conference? If so FS can do it. - adjust individual channel or the conference volume? yup - background noise reduction? yup I've heard of freeswitch handling 500 channels on a standard server box. It has a full feature list. Even sipX is merging freeswitch in, starting with conferencing. http://sipx-wiki.calivia.com/index.php/Conferenceing_Service_for_sipXecs SipX also has a gui front end to manage the freeswitch conference bridge, so might be a easy way to test it out and it also offers an easy installer which can also configure a high availability system. http://sipx-wiki.calivia.com/index.php/CD_Installation_of_sipXecs Mike Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield Metroconnect audace fortuna iuvat _ From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 9:23 AM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, In my opinion freeswitch has the best base conference bride features, no dependency on hardware or the ztdummy timer and loads more features. For a comparison of the FS Asterisk features here is link to a comparison http://www.freeswitch.org/node/100 Also here is a small article ( http://www.junctionnetworks.com/blog/charlotte/2008/05/21/freeswitch-asteris k-replacement ) and their rational of picking FS over Asterisk for their conference bridge product. Hope this helps, Mike Rachel Quin wrote: I want to build a conference bridge using dedicated DSP hardware, running on FreeBSD. Does anyone have recomendations on HW/SW? Rachel Quin Beanfield Metroconnect audace fortuna iuvat
RE: [on-asterisk] Conference bridge
I'll dig through the dox to see if it can do things other than codec conversion. The funny thing is that I use a UA DSP card to digital audio mixing and effects for music, and there is a tonne of software to support it. We've spent over half a million dollars so far on our voice infrastructure, and it is big iron, but beginner's big iron. I'd like to build more agile system, so we can put hardware closer to the customers. I've already started designing integrating vmail, email, fax, IM, into a multimode communications storage front end (we are so not a M$ shop here ;^). And this will be a fun one, but I'd really like to do conferencing as well. -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 12:24 PM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge since this is a zaptel card, it will do channel bridging and offload the dsp operations to the card from what ive been told. your probably best to pick one up for testing in house at this point :) Rachel Quin wrote: Ok, almost there, but not quite. I don't want to do dedicated codec conversion. In fact, we will never use anything other than G.711. I'm looking for a more generic DSP card that can be used to do all the channel mixing, offloading the work from the server's processors. And!! The software that can utilize it. I can find the cards easy enough, there are quite a few to choose from, but of the OSS, I know not. To quote from the Digium page: These transformations in software are very expensive, in terms of MIPS, and require a substantial amount of CPU time to accomplish. Channel mixing also quite expensive, when you're talking about the high hundreds of channels. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 12:04 PM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge gotcha, in that case take a gander at the digium tc400b card, that handles dsp offloading for asterisk and also acts as a transcoding accelerator. Rachel Quin wrote: I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more
RE: [on-asterisk] Conference bridge
Would anyone like to get scientific about it? I'm now really curious to know how many G.711 channels a quad core Xeon could mix using Asterisk, FreeSWITCH, and YATE respectively. Does anyone have a spare available that we might run some automated testing against? And if we're feeling really sick, we could also do some MOS tests (Mean Opinion Score); the test is outlined in ITU P.800: http://www.itu.int/rec/T-REC-P.800-199608-I/en Cheers, spd -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | Network: http://www.linkedin.com/in/spditner | http://facebook.com/people/Simon-P-Ditner/776370031 | http://twitter.com/spditner On Fri, 27 Feb 2009, Rachel Quin wrote: No, that flexibility is exactly what I'm looking for, but you simply can't mix that many G.711 channels in Xeon cores. My question is, does anyone know of any open source software that will utilize DSP cards for the actual voice stream crunching of G.711 channels? All of the signalling and management function would be in the software running on the host hardware. Every Telco grade media mixer does this, every edge T3 or OC gateway. Can FreeSwitch or Asterisk do all the conferencing work, using DSP hardware offloading? Rachel From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 12:03 PM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, I think you may have a misconception of what Asterisk ad/or FreeSwitch are. They are really telephony/media software platforms that can be configured to do many things. The most frequent uses are as full blown PBX phone systems, but they can be used strictly as, a VM platform, an IVR application server, media gateways, etc. Mike Rachel Quin wrote: I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield Metroconnect audace fortuna iuvat _ From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 9:23 AM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, In my opinion freeswitch has the best base conference bride features, no dependency on hardware or the ztdummy timer and loads more features. For a comparison of the FS Asterisk features here is link to a comparison
RE: [on-asterisk] Conference bridge
That, I would like to see! Our existing system supports 3 party conferencing just fine, but when I start looking at 20-30 party conferences, with 20-30 concurrent conferences, I really, really have a hard time believing that even two quad-core 64bit Xeons are going to handle the load of real-time mixing of 64kbit streams, doing the AGC, cleanly, without jitter, without artefacts, telco quality sound, and still handle all the rest of the systems load. -Original Message- From: Simon P. Ditner [mailto:si...@uc.org] Sent: February 27, 2009 12:17 PM To: asterisk@uc.org Subject: RE: [on-asterisk] Conference bridge Would anyone like to get scientific about it? I'm now really curious to know how many G.711 channels a quad core Xeon could mix using Asterisk, FreeSWITCH, and YATE respectively. Does anyone have a spare available that we might run some automated testing against? And if we're feeling really sick, we could also do some MOS tests (Mean Opinion Score); the test is outlined in ITU P.800: http://www.itu.int/rec/T-REC-P.800-199608-I/en Cheers, spd -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | Network: http://www.linkedin.com/in/spditner | http://facebook.com/people/Simon-P-Ditner/776370031 | http://twitter.com/spditner On Fri, 27 Feb 2009, Rachel Quin wrote: No, that flexibility is exactly what I'm looking for, but you simply can't mix that many G.711 channels in Xeon cores. My question is, does anyone know of any open source software that will utilize DSP cards for the actual voice stream crunching of G.711 channels? All of the signalling and management function would be in the software running on the host hardware. Every Telco grade media mixer does this, every edge T3 or OC gateway. Can FreeSwitch or Asterisk do all the conferencing work, using DSP hardware offloading? Rachel From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 12:03 PM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, I think you may have a misconception of what Asterisk ad/or FreeSwitch are. They are really telephony/media software platforms that can be configured to do many things. The most frequent uses are as full blown PBX phone systems, but they can be used strictly as, a VM platform, an IVR application server, media gateways, etc. Mike Rachel Quin wrote: I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open
RE: [on-asterisk] Conference bridge
I have a spare dual-quad core blade that we aren't using (yet) in our datacenter. I could make it available for this test...but I will need it back in 3-4 weeks -Original Message- From: Simon P. Ditner [mailto:si...@uc.org] Sent: Friday, February 27, 2009 12:17 PM To: asterisk@uc.org Subject: RE: [on-asterisk] Conference bridge Would anyone like to get scientific about it? I'm now really curious to know how many G.711 channels a quad core Xeon could mix using Asterisk, FreeSWITCH, and YATE respectively. Does anyone have a spare available that we might run some automated testing against? And if we're feeling really sick, we could also do some MOS tests (Mean Opinion Score); the test is outlined in ITU P.800: http://www.itu.int/rec/T-REC-P.800-199608-I/en Cheers, spd -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | Network: http://www.linkedin.com/in/spditner | http://facebook.com/people/Simon-P-Ditner/776370031 | http://twitter.com/spditner On Fri, 27 Feb 2009, Rachel Quin wrote: No, that flexibility is exactly what I'm looking for, but you simply can't mix that many G.711 channels in Xeon cores. My question is, does anyone know of any open source software that will utilize DSP cards for the actual voice stream crunching of G.711 channels? All of the signalling and management function would be in the software running on the host hardware. Every Telco grade media mixer does this, every edge T3 or OC gateway. Can FreeSwitch or Asterisk do all the conferencing work, using DSP hardware offloading? Rachel From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 12:03 PM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, I think you may have a misconception of what Asterisk ad/or FreeSwitch are. They are really telephony/media software platforms that can be configured to do many things. The most frequent uses are as full blown PBX phone systems, but they can be used strictly as, a VM platform, an IVR application server, media gateways, etc. Mike Rachel Quin wrote: I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like to offer conferencing services, but we can't do anything completely amateur hour. I've heard of someone using four dual core Xeon to process 180 channels, and I had a nice little chuckle ;^) In thinking back over the problem, I guess I have to look at the actual DSP cards, Sharks, TI's, and see what I like, but does anyone have any experience with any open source software using DSP offload cards? At this juncture I'm more worried about H/W support than features. I'll probably be looking at a 16 or 32 core DSP card, but as I said, I've got to do some shopping. Any thoughts, suggestions? Rachel Quin Beanfield
Re: [on-asterisk] Conference bridge
You know what? That's a fantastic idea! We should set up a basic idea of what we want to test (keep it simple), and the get together and have a 'mix-off' between YATE, FS and Asterisk and see which one can handle the most channels before it starts having issues. We probably wouldn't be able to be too formal in our methodology, so some purists might scoff, but if we did the same sort of test on the same sort of system, it'd at the least give some sort of benchmark as to where the performance differences lie. I would love to test this on the following: - Atom (I have a lab box for that) - Core 2 Duo - Core 2 Quad - Xeon - Dual Xeon Sorting out performance differences between the various projects would be interesting enough, but what would also be neat to discover is whether, say a core2 quad is that much better than a core2 duo, and whether a xeon is far better, or only a little better. I have an intel Atom box that I can bring to the party. Can't currently help with the others, though. Jim Simon P. Ditner wrote: Would anyone like to get scientific about it? I'm now really curious to know how many G.711 channels a quad core Xeon could mix using Asterisk, FreeSWITCH, and YATE respectively. Does anyone have a spare available that we might run some automated testing against? And if we're feeling really sick, we could also do some MOS tests (Mean Opinion Score); the test is outlined in ITU P.800: http://www.itu.int/rec/T-REC-P.800-199608-I/en Cheers, spd -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | Network: http://www.linkedin.com/in/spditner | http://facebook.com/people/Simon-P-Ditner/776370031 | http://twitter.com/spditner On Fri, 27 Feb 2009, Rachel Quin wrote: No, that flexibility is exactly what I'm looking for, but you simply can't mix that many G.711 channels in Xeon cores. My question is, does anyone know of any open source software that will utilize DSP cards for the actual voice stream crunching of G.711 channels? All of the signalling and management function would be in the software running on the host hardware. Every Telco grade media mixer does this, every edge T3 or OC gateway. Can FreeSwitch or Asterisk do all the conferencing work, using DSP hardware offloading? Rachel From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 12:03 PM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, I think you may have a misconception of what Asterisk ad/or FreeSwitch are. They are really telephony/media software platforms that can be configured to do many things. The most frequent uses are as full blown PBX phone systems, but they can be used strictly as, a VM platform, an IVR application server, media gateways, etc. Mike Rachel Quin wrote: I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end BSCs, and currently Iperia for vmail (though I'd like to build my own solution for that). I'd like
Re: [on-asterisk] Conference bridge
What no amds in there? We should also toss an intel i7 in there for testing. Jim Van Meggelen wrote: You know what? That's a fantastic idea! We should set up a basic idea of what we want to test (keep it simple), and the get together and have a 'mix-off' between YATE, FS and Asterisk and see which one can handle the most channels before it starts having issues. We probably wouldn't be able to be too formal in our methodology, so some purists might scoff, but if we did the same sort of test on the same sort of system, it'd at the least give some sort of benchmark as to where the performance differences lie. I would love to test this on the following: - Atom (I have a lab box for that) - Core 2 Duo - Core 2 Quad - Xeon - Dual Xeon Sorting out performance differences between the various projects would be interesting enough, but what would also be neat to discover is whether, say a core2 quad is that much better than a core2 duo, and whether a xeon is far better, or only a little better. I have an intel Atom box that I can bring to the party. Can't currently help with the others, though. Jim Simon P. Ditner wrote: Would anyone like to get scientific about it? I'm now really curious to know how many G.711 channels a quad core Xeon could mix using Asterisk, FreeSWITCH, and YATE respectively. Does anyone have a spare available that we might run some automated testing against? And if we're feeling really sick, we could also do some MOS tests (Mean Opinion Score); the test is outlined in ITU P.800: http://www.itu.int/rec/T-REC-P.800-199608-I/en Cheers, spd -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | Network: http://www.linkedin.com/in/spditner | http://facebook.com/people/Simon-P-Ditner/776370031 | http://twitter.com/spditner On Fri, 27 Feb 2009, Rachel Quin wrote: No, that flexibility is exactly what I'm looking for, but you simply can't mix that many G.711 channels in Xeon cores. My question is, does anyone know of any open source software that will utilize DSP cards for the actual voice stream crunching of G.711 channels? All of the signalling and management function would be in the software running on the host hardware. Every Telco grade media mixer does this, every edge T3 or OC gateway. Can FreeSwitch or Asterisk do all the conferencing work, using DSP hardware offloading? Rachel From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 12:03 PM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, I think you may have a misconception of what Asterisk ad/or FreeSwitch are. They are really telephony/media software platforms that can be configured to do many things. The most frequent uses are as full blown PBX phone systems, but they can be used strictly as, a VM platform, an IVR application server, media gateways, etc. Mike Rachel Quin wrote: I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start with some background: Beanfield Metroconnect is a dark-fibre, lan-ex, and Internet provider for office buildings in the downtown core. We have an extensive 10gig backbone, two large pops and datacenters in Toronto, and one in NY NY. We own the fibre end to end in our core, and we offer business services exclusively. We are just branching into voice services, and our initial setup is the following: we're fully redundant with each site having Sylantro for switching, Convedia media mixers, AS5400-t3 links to Bell, Bell Megalink circuits for wholesale long distance, top end
Re: [on-asterisk] Conference bridge
Personally, I'd be less interested in which processors do what than seeing how things scale with processor power. Are there still significant timing issues with Vmware? If that can be overcome there are some serious advantages to using it as a test platform. You can dial the CPU and memory up and down on demand. Would that be beneficial or would it just muddy the waters? We have an ESX box here with 2 x 3.2ghz dual cores in it. Like Bill, mine will have to go into production some time in the next month or so but let me know if it would help. Dave On Fri, Feb 27, 2009 at 4:16 PM, Jim Van Meggelen j...@vanmeggelen.ca wrote: You know what? That's a fantastic idea! We should set up a basic idea of what we want to test (keep it simple), and the get together and have a 'mix-off' between YATE, FS and Asterisk and see which one can handle the most channels before it starts having issues. We probably wouldn't be able to be too formal in our methodology, so some purists might scoff, but if we did the same sort of test on the same sort of system, it'd at the least give some sort of benchmark as to where the performance differences lie. I would love to test this on the following: - Atom (I have a lab box for that) - Core 2 Duo - Core 2 Quad - Xeon - Dual Xeon Sorting out performance differences between the various projects would be interesting enough, but what would also be neat to discover is whether, say a core2 quad is that much better than a core2 duo, and whether a xeon is far better, or only a little better. I have an intel Atom box that I can bring to the party. Can't currently help with the others, though. Jim Simon P. Ditner wrote: Would anyone like to get scientific about it? I'm now really curious to know how many G.711 channels a quad core Xeon could mix using Asterisk, FreeSWITCH, and YATE respectively. Does anyone have a spare available that we might run some automated testing against? And if we're feeling really sick, we could also do some MOS tests (Mean Opinion Score); the test is outlined in ITU P.800: http://www.itu.int/rec/T-REC-P.800-199608-I/en Cheers, spd -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | Network: http://www.linkedin.com/in/spditner | http://facebook.com/people/Simon-P-Ditner/776370031 | http://twitter.com/spditner On Fri, 27 Feb 2009, Rachel Quin wrote: No, that flexibility is exactly what I'm looking for, but you simply can't mix that many G.711 channels in Xeon cores. My question is, does anyone know of any open source software that will utilize DSP cards for the actual voice stream crunching of G.711 channels? All of the signalling and management function would be in the software running on the host hardware. Every Telco grade media mixer does this, every edge T3 or OC gateway. Can FreeSwitch or Asterisk do all the conferencing work, using DSP hardware offloading? Rachel From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 12:03 PM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, I think you may have a misconception of what Asterisk ad/or FreeSwitch are. They are really telephony/media software platforms that can be configured to do many things. The most frequent uses are as full blown PBX phone systems, but they can be used strictly as, a VM platform, an IVR application server, media gateways, etc. Mike Rachel Quin wrote: I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Have you looked into Metaswitch? Rachel Quin wrote: I'd actually like to reintroduce my question. I'll start
RE: [on-asterisk] Conference bridge
Jim and all: We are in the process of deploying one of these in our datacenter. http://www.supermicro.com/products/SuperBlade/datacenterblade/ we are using this blade with dual quad core xeons, 12GB ram and a 640GB RAID5 array. http://www.supermicro.com/products/SuperBlade/module/SBI-7425C-S3.cfm We have the hardware but its going to be another month or so before we have the staff resources to deploy. So if you want 1 blade for each of FreeSwitch, Asterisk and YATE, I can make available but it has to get done in the next month. Bill -Original Message- From: Jim Van Meggelen [mailto:j...@vanmeggelen.ca] Sent: Friday, February 27, 2009 4:17 PM To: Simon P. Ditner Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge You know what? That's a fantastic idea! We should set up a basic idea of what we want to test (keep it simple), and the get together and have a 'mix-off' between YATE, FS and Asterisk and see which one can handle the most channels before it starts having issues. We probably wouldn't be able to be too formal in our methodology, so some purists might scoff, but if we did the same sort of test on the same sort of system, it'd at the least give some sort of benchmark as to where the performance differences lie. I would love to test this on the following: - Atom (I have a lab box for that) - Core 2 Duo - Core 2 Quad - Xeon - Dual Xeon Sorting out performance differences between the various projects would be interesting enough, but what would also be neat to discover is whether, say a core2 quad is that much better than a core2 duo, and whether a xeon is far better, or only a little better. I have an intel Atom box that I can bring to the party. Can't currently help with the others, though. Jim Simon P. Ditner wrote: Would anyone like to get scientific about it? I'm now really curious to know how many G.711 channels a quad core Xeon could mix using Asterisk, FreeSWITCH, and YATE respectively. Does anyone have a spare available that we might run some automated testing against? And if we're feeling really sick, we could also do some MOS tests (Mean Opinion Score); the test is outlined in ITU P.800: http://www.itu.int/rec/T-REC-P.800-199608-I/en Cheers, spd -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | Network: http://www.linkedin.com/in/spditner | http://facebook.com/people/Simon-P-Ditner/776370031 | http://twitter.com/spditner On Fri, 27 Feb 2009, Rachel Quin wrote: No, that flexibility is exactly what I'm looking for, but you simply can't mix that many G.711 channels in Xeon cores. My question is, does anyone know of any open source software that will utilize DSP cards for the actual voice stream crunching of G.711 channels? All of the signalling and management function would be in the software running on the host hardware. Every Telco grade media mixer does this, every edge T3 or OC gateway. Can FreeSwitch or Asterisk do all the conferencing work, using DSP hardware offloading? Rachel From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 12:03 PM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, I think you may have a misconception of what Asterisk ad/or FreeSwitch are. They are really telephony/media software platforms that can be configured to do many things. The most frequent uses are as full blown PBX phone systems, but they can be used strictly as, a VM platform, an IVR application server, media gateways, etc. Mike Rachel Quin wrote: I think I'm not making myself clear, sorry. Our t3's and Megalink circuit from Bell come into AS5400's. Our VoIP infrastructure is entirely SIP. A conferencing server would only handle RTP streams, mixing channels for many large-ish volume conferences. The box I'm talking about would have 2 10gig nics, one or two DSP cards, and whatever software is needed to handle managing conferencing and directing RTP/G.711 content channels to and from the DSP card(s). I am not looking to build a stand alone phone system. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 11:21 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge asterisk does work well and you can stick 2 cards (8 pris worth of cards to a beefy server) but really no more. for larger scale conferencing on a single box you really need something larger. Rachel Quin wrote: Really all I'm looking at is media mixing for call conferencing, I have all the other puzzle pieces. Rachel -Original Message- From: Philip Mullis [mailto:philip.mul...@syx.ca] Sent: February 27, 2009 10:44 AM To: Rachel Quin Cc: asterisk@uc.org Subject: Re: [on-asterisk
Re: [on-asterisk] Conference bridge
Sure thing. For me it's not so much about this CPU vs that CPU, but more about giving people an idea about what sort of performance to expect from different platforms. So to qualify, what I think we'd want to do is come up with several categories of system. Five categories that come to my mind are: 1) A low end, low power, cheap and totally x86 compatible processor or board (such as Intel Atom) for use in small deployments. The key here would be low power consumption, low noise, simple Linux install (i.e. no mucking about with embedded OSes), and cheap sticker price. Typical uses would be a SOHO office. 2) An average system that'd be common in a PC-based deployment (such as Intel Core 2) for smaller SMEs 3) An average system that'd be common in a Server-based deployment (such as Dual Intel Xeon) for larger SMEs 4) The most powerful x86 system that money can buy. The idea here is in a clustered environment where you want to squeeze every last drop of horsepower out of every machine, and you have to support tens or even hundreds of thousands of users. (this is really beyond the scope of what we can reasonably test, I would think) 5) The best platform for running virtual machines (if there is such a thing). This is probably out of scope for the first lab day. There's just too many extra variables when dealing with VMs Doing a test using conferencing would not give any understanding of how it would handle other PBX tasks, or things like AGI or database work, but it'd sure be telling regardless. I would say if we were to look to do 1, 2 and 3, we'd have accomplished something worthwhile. 4 and 5 are probably a different project (also well worth doing), and would relate to each other in many ways. There is certainly no reason not to use AMD-based systems, as long as we categorize them in terms that would relate to similar Intel offerings. I'm not an Intel fanboy, but they do hold the market share, and I would think that people will tend to want to see benchmarks that they can relate back to Intel's offerings. Jim Philip Mullis wrote: What no amds in there? We should also toss an intel i7 in there for testing. Jim Van Meggelen wrote: You know what? That's a fantastic idea! We should set up a basic idea of what we want to test (keep it simple), and the get together and have a 'mix-off' between YATE, FS and Asterisk and see which one can handle the most channels before it starts having issues. We probably wouldn't be able to be too formal in our methodology, so some purists might scoff, but if we did the same sort of test on the same sort of system, it'd at the least give some sort of benchmark as to where the performance differences lie. I would love to test this on the following: - Atom (I have a lab box for that) - Core 2 Duo - Core 2 Quad - Xeon - Dual Xeon Sorting out performance differences between the various projects would be interesting enough, but what would also be neat to discover is whether, say a core2 quad is that much better than a core2 duo, and whether a xeon is far better, or only a little better. I have an intel Atom box that I can bring to the party. Can't currently help with the others, though. Jim Simon P. Ditner wrote: Would anyone like to get scientific about it? I'm now really curious to know how many G.711 channels a quad core Xeon could mix using Asterisk, FreeSWITCH, and YATE respectively. Does anyone have a spare available that we might run some automated testing against? And if we're feeling really sick, we could also do some MOS tests (Mean Opinion Score); the test is outlined in ITU P.800: http://www.itu.int/rec/T-REC-P.800-199608-I/en Cheers, spd -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | Network: http://www.linkedin.com/in/spditner | http://facebook.com/people/Simon-P-Ditner/776370031 | http://twitter.com/spditner On Fri, 27 Feb 2009, Rachel Quin wrote: No, that flexibility is exactly what I'm looking for, but you simply can't mix that many G.711 channels in Xeon cores. My question is, does anyone know of any open source software that will utilize DSP cards for the actual voice stream crunching of G.711 channels? All of the signalling and management function would be in the software running on the host hardware. Every Telco grade media mixer does this, every edge T3 or OC gateway. Can FreeSwitch or Asterisk do all the conferencing work, using DSP hardware offloading? Rachel From: Mike Ashton [mailto:mike.ash...@qualitytrack.com] Sent: February 27, 2009 12:03 PM Cc: asterisk@uc.org Subject: Re: [on-asterisk] Conference bridge Rachel, I think you may have a misconception of what Asterisk ad/or FreeSwitch are. They are really telephony/media software platforms that can be configured to do many things. The most frequent uses are as full blown PBX
