In our experience, it's not a bandwidth limitation. If you do nothing special, interrupt servicing for a single NIC on our high throughput hardware maxes at something in excess of 1,000 calls when you are keeping the streams. I don't believe you can get even that far on a PC server, but we haven't tried.
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Tuesday, January 31, 2006 5:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] 5,000 concurrent calls system rollout question > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of C F > Sent: Tuesday, January 31, 2006 4:03 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout > question > > I don't know how much 1+1 by you is, but lets recalculate this for a > moment: > First the bandwidth per channel: > http://www.airewaves.com/aire/support/bandwidth_explain.php > 1.5mbps (mega *BITS* not BYTES per second) to a full T1, which equals > 1536 Kbits, each channel then takes 64kbps. > 64*5,000=320,000kbps. > 32,000/1,024=312.5 Mbps (round off to Mbps), no where close to a Gb. > Every single PC made in the last 4 years I came across, can handle > this type of bandwidth. > BTW, this all amounts to just over 39 MBYTES per second. 312.5/8=39.0625 > Not that I disagree with your point, the bandwidth is not huge, but the math is a little fuzzy; First of all, a g.711u stream over UDP is closer 80k than 64k, the payload is 64k + udp overhead + IP overhead. Now consider that the call is originated as SIP (llok back a few days in the thread), and lets assume the call goes to an external hard or softphone, and lets also assume that there is a reason to keep the RTP stream running through asterisk (monitoring, recording, transferring, dtmf, ability to re-enter IVR, etc). I make all the assumptions safely since the thread was started by someone looking to set up a large call center and I have followed thread out of curiosity. So a 80k full duplex RTP stream originates on media gateway somewhere, hits the asterisk box, is internally bridged, and is sent back out to a phone somewhere. My math says this puts a 160kbps full duplex load in the NIC. Ok, now lets go for 5000 of them. 160kbps*5000=800000kbps or 800mbps - full duplex. Have you ever seen a NIC or switch that can run GigE full duplex at 80% utilization and not at least start to fall apart? To get to a comfortable load you would need 2x GigE NICs (for ~40% utilization), of course now we are adding additional overhead for the bonded NIC trunking protocol. Is still contend this is not practical without multiple very high end servers and round robin call origination from the upstream provider delivered over something like GigE or OCx. Maybe someone will step up and post some real-world application limits based on experience... _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
