jan.sarin@securia.se
2006-Jul-04 00:49 UTC
SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)
I should add that thease 25 calls where SIP (internal) to Zap (PSTN) calls. Mvh, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r jan.sarin@securia.se Skickat: den 4 juli 2006 09:41 Till: asterisk-users@lists.digium.com ?mne: [Asterisk-Users] Running 40 active calls (too much f?r CPU?) Hi, We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server connected to the PSTN through two E1 pipes to a TE405P. This has been running just fine for several months... But yesturday we connected a large number of softphone SIP clients (50) and 25 of these where running simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio "tracks" on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan _______________________________________________ --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
jan.sarin@securia.se
2006-Jul-04 02:20 UTC
SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)
Hello again, I read this interesting article about the TE405P card. How do I check what firmware version my card has? http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html ... And how do I update it if it's an old one? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r jan.sarin@securia.se Skickat: den 4 juli 2006 09:41 Till: asterisk-users@lists.digium.com ?mne: [Asterisk-Users] Running 40 active calls (too much f?r CPU?) Hi, We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server connected to the PSTN through two E1 pipes to a TE405P. This has been running just fine for several months... But yesturday we connected a large number of softphone SIP clients (50) and 25 of these where running simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio "tracks" on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan _______________________________________________ --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
broadbandvoice@comcast.net
2006-Jul-04 06:54 UTC
SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)
Are the phones behind a NAT? What is the processory memory size? Are the E1 channelized? -------------- Original message -------------- From: <jan.sarin@securia.se>> I should add that thease 25 calls where SIP (internal) to Zap (PSTN) calls. > > Mvh, > Jan > > -----Ursprungligt meddelande----- > Från: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] För jan.sarin@securia.se > Skickat: den 4 juli 2006 09:41 > Till: asterisk-users@lists.digium.com > Ämne: [Asterisk-Users] Running 40 active calls (too much för CPU?) > > Hi, > > We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server > connected to the PSTN through two E1 pipes to a TE405P. This has been running > just fine for several months... > > But yesturday we connected a large number of softphone SIP clients (50) and 25 > of these where running simultaneous active calls on the INTERNAL ethernet using > g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't > handle 25 calls (?!). > > I checked the CPU load and it never went over 55 % and memusage was low too. > > Does anyone know what could be the problem? Are there some kind of CPU spikes > that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor > handle 25 low-quality audio "tracks" on asterisk when I can run +50 cd-quality > audio tracks when producing music? > > ANY help and/or comments would be appreciated since this is quite an acute > problem. > > Regards, > Jan > _______________________________________________ > --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-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060704/4686ef4c/attachment.htm
jan.sarin@securia.se
2006-Jul-04 08:26 UTC
SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)
Phones are not behind NAT. Every client is on the same internal network as the asterisk pbx (nothing is sent through the internet). It's not the network since I tested this by calling asterisk from an outside phone (cell) and let asterisk play a message for me. Same "cutting" and "chopping" when many SIP-clients where active in a call at the same time. Computer RAM is 2 gb. If the E1 is channelized or not I don't actually know. How would I know this and why would it affect the call quality when many people are in a call at the same time (same lines work fine with an Ericsson BusinessPhone Exchange)? Thanks! Regards, Jan ________________________________ Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r broadbandvoice@comcast.net Skickat: den 4 juli 2006 15:55 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: Re: SV: [Asterisk-Users] Running 40 active calls (too much f?r CPU?) Are the phones behind a NAT? What is the processory memory size? Are the E1 channelized? -------------- Original message -------------- From: <jan.sarin@securia.se> > I should add that thease 25 calls where SIP (internal) to Zap (PSTN) calls. > > Mvh, > Jan > > -----Ursprungligt meddelande----- > Fr?n: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] F?r jan.sarin@securia.se > Skickat: den 4 juli 2006 09:41 > Till: asterisk-users@lists.digium.com > ?mne: [Asterisk-Users] Running 40 active calls (too much f?r CPU?) > > Hi, > > We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server > connected to the PSTN through two E1 pipes to a TE405P. This has been running > just fine for several months... > > But yesturday we connected a large number of softphone SIP clients (50) and 25 < BR>> ; of these where running simultaneous active calls on the INTERNAL ethernet using > g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't > handle 25 calls (?!). > > I checked the CPU load and it never went over 55 % and memusage was low too. > > Does anyone know what could be the problem? Are there some kind of CPU spikes > that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor > handle 25 low-quality audio "tracks" on asterisk when I can run +50 cd-quality > audio tracks when producing music? > > ANY help and/or comments would be appreciated since this is quite an acute > problem. > > Regards, > Jan > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > h ttp:// 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060704/92b0b3ea/attachment.htm