has anyone experienced a problem where RTP audio cuts out when doing 30-40 concurrent channels via sip? The box is freebsd 6, asterisk 1.2.4, voip only (no libpri, no zaptel - not even a timing source) The box has plenty of bandwidth, when a call to the same box is iax2 it works, but when its sip a call gets connected a few frames of audio are passed and then silence. When the box is completly idle sip does not experience this problem, it is only when there are a few concurrent calls. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060214/df7ac2eb/attachment.pgp
Hey All,
I've been experiencing a problem for a bit. During a call to the PSTN, audio
will cut out for 2-5 seconds. It's completely random and may or may not
happen during a call.
Our setup is 79XX phones -> asterisk -> 2811 router -> PRI to the PSTN.
Everything is talking SIP. The asterisk box is a dual core system.
/proc/interrupts looks like:
cat /proc/interrupts
CPU0 CPU1
0: 733669449 732813122 IO-APIC-edge timer
8: 1 0 IO-APIC-edge rtc
9: 0 0 IO-APIC-level acpi
14: 6598410 6589174 IO-APIC-edge ide0
169: 0 0 IO-APIC-level uhci_hcd
185: 0 0 IO-APIC-level ehci_hcd, uhci_hcd
193: 0 0 IO-APIC-level uhci_hcd
201: 0 0 IO-APIC-level uhci_hcd
209: 11404158 10762030 IO-APIC-level 3w-9xxx
225: 100440701 136 PCI-MSI eth0
233: 14 10512166 PCI-MSI eth1
NMI: 0 0
LOC: 1466464719 1466464718
ERR: 0
MIS: 0
Can-Reinvite is enabled, but I do have it configured to allow call recording
on outbound calls, so I think the audio streams all go through asterisk.
There are no G.729 licenses involved and everything should be talking G.711.
Oh, and this is an 1.2.7.1 install. ztdummy is loaded.
Does anyone have any insite into this problem?
Thanks.
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We battled this same issue for a couple weeks, at about 30-50 simultaneous recordings the audio would get all screwy. I looked at that solution but opted for something a little more passive. I use orkaudio to sniff rtp streams and mux them. I have it to perfect quality, the same as the monitor app in asterisk. I think it is a much better solution than ramdisk since it is so passive and puts no strain or need for additional RAM on the asterisk machine itself. Let the phone system be a phone system and not a recording device I say. Best part about the orkaudio project is Henri. I had audio issues with orkaudio in the beginning but Henri re-worked his software to eliminate my problems in a matter of days. A true credit and great contribution to opensource software. Thanks, Steve Totaro Gary Richardson wrote:> That could be an issue. Would recording to a ram drive solve the problem? > > Thanks. > > On 6/12/06, *Steve Totaro* < stotaro@asteriskhelpdesk.com > <mailto:stotaro@asteriskhelpdesk.com>> wrote: > > Recording many simultaneous calls can cause this behavior too. > > Thanks, > Steve Totaro > > > ------------------------------------------------------------------------ > > _______________________________________________ > --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 >