Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500 near-simultaneous INVITEs being sent. Can the SIP channel handle that? Any suggestions or war stories are appreciated. Mark Willis Cartama Consulting LLC 210 698 5097
Mark Willis wrote: This could potentially create a very weird audio situation where the delay between adjacent phones is audible so instead of acting like loudspeakers in parallel on a conventional system, it just sounds like a bunch of people talking at once and is not understandable.> Has anyone done any large scale intercom deployments with Asterisk? I've > been asked about building a system to one-way page 500 phones > simultaneously from a single server. > > My concerns are: > > - My limited math capabilities suggest 41 Mbps of RTP traffic, which > seems like a lot, plus asterisk would be taking a single input stream > and exploding it out to 500 endpoints. > - There are 500 near-simultaneous INVITEs being sent. Can the SIP > channel handle that? > > Any suggestions or war stories are appreciated. > > Mark Willis > Cartama Consulting LLC > 210 698 5097 > > >
Mark Willis schrieb:> Has anyone done any large scale intercom deployments with Asterisk? I've > been asked about building a system to one-way page 500 phones > simultaneously from a single server. > > My concerns are: > > - My limited math capabilities suggest 41 Mbps of RTP traffic, which > seems like a lot, plus asterisk would be taking a single input stream > and exploding it out to 500 endpoints. > - There are 500 near-simultaneous INVITEs being sent. Can the SIP > channel handle that? > > Any suggestions or war stories are appreciated.Multicast RTP might be the solution. http://wiki.snom.com/Settings/multicast_listen http://wiki.snom.com/Settings/multicast_address http://forum.snom.com/index.php?showtopic=1905 Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de --
On Fri, Feb 5, 2010 at 4:50 PM, Mark Willis <markslists at markwillis.net> wrote:> Has anyone done any large scale intercom deployments with Asterisk? I've > been asked about building a system to one-way page 500 phones > simultaneously from a single server. > > My concerns are: > > - My limited math capabilities suggest 41 Mbps of RTP traffic, which > seems like a lot, plus asterisk would be taking a single input stream > and exploding it out to 500 endpoints. > - There are 500 near-simultaneous INVITEs being sent. Can the SIP > channel handle that? > > Any suggestions or war stories are appreciated. > > Mark Willis > Cartama Consulting LLC > 210 698 5097What you really want is multicast RTP, preferably as implemented in SNOM phones: https://issues.asterisk.org/view.php?id=11797 http://wiki.snom.com/Settings/multicast_address One RTP stream, any number of receivers, no SIP session. Doing this with unicast RTP and individual INVITEs would be tough. If your system can't do 500 call setups per second (or better) you'll introduce massive delays in call setup to the recipients, not to mention serious RTP burden with that many streams. I hope you haven't bought phones yet (or bought Snom) ;). -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
On Fri, 5 Feb 2010, Mark Willis wrote:> Has anyone done any large scale intercom deployments with Asterisk? I've > been asked about building a system to one-way page 500 phones > simultaneously from a single server. > > My concerns are: > > - My limited math capabilities suggest 41 Mbps of RTP traffic, which > seems like a lot, plus asterisk would be taking a single input stream > and exploding it out to 500 endpoints.How did you get that number? Even with ulaw @ 64Kbps you theoretically get 32Mbps. If you used G.729 you would cut that down to 4 or 5Mbps. Totally oversimplified, but that seems a lot more doable.> - There are 500 near-simultaneous INVITEs being sent. Can the SIP > channel handle that? >I can't say I have ever pushed that hard, but that doesn't sound like it would be difficult to handle. There are plenty claiming they have 400 simultaneous two way conversations going on a single box. j> Any suggestions or war stories are appreciated. > > Mark Willis > Cartama Consulting LLC > 210 698 5097 > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Hi!> Has anyone done any large scale intercom deployments with Asterisk? > I've been asked about building a system to one-way page 500 phones > simultaneously from a single server. > > My concerns are: > > - My limited math capabilities suggest 41 Mbps of RTP traffic, which > seems like a lotUse multi-cast: Read the "See also" section at the bottom of this page and look at "MAST" and/or "app_rtppage". A couple of phone vendors have multi-cast support in their models. http://www.voip-info.org/wiki/view/Asterisk+cmd+Page http://www.voip- info.org/wiki/index.php?page=Asterisk+phone+snom#RelatedMulticastapp_rtppa geAsterisk16orl Philipp
For a case like this I would go with overhead paging. On Fri, Feb 5, 2010 at 4:50 PM, Mark Willis <markslists at markwillis.net> wrote:> Has anyone done any large scale intercom deployments with Asterisk? I've > been asked about building a system to one-way page 500 phones > simultaneously from a single server. > > My concerns are: > > - My limited math capabilities suggest 41 Mbps of RTP traffic, which > seems like a lot, plus asterisk would be taking a single input stream > and exploding it out to 500 endpoints. > - There are 500 near-simultaneous INVITEs being sent. Can the SIP > channel handle that? > > Any suggestions or war stories are appreciated. > > Mark Willis > Cartama Consulting LLC > 210 698 5097 > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-users >