[Asterisk-Users] Skips and Pops in Call Recordings

Matt Florell astmattf at gmail.com
Tue Dec 13 10:17:59 MST 2005


What codec are the calls? What codec are you recording in?

I would try some non-Dell hardware, I would also try a less bloated
Linux Distro, something like Slackware, just to see if that had any
effect. And make sure you use the megaraid2 linux drivers.

MATT---

On 12/13/05, Matt Roth <mroth at imminc.com> wrote:
> Matt Florell wrote:
>
>  >Hello,
>  >
>  >Need some more information here:
>  >- hardware specs (including what kind of hard drives?)
> The Asterisk server is a Dell PowerEdge 6850 with the following specs.
> Please note that we are NOT recording to the hard drive.  We are
> recording to a RAM disk as detailed here
> (http://voip-info.org/wiki/view/Asterisk+Dimensioning) under the heading
> "512 simultaneous SIP-to-SIP calls with Digital Recording".
> Unfortunately, the scalability tests we did at that time assumed that if
> call quality was good, so was the quality of the recording.
>
> Processor:  Quad Intel Xeon 3.16GHz/1MB Cache
> Memory: 20 GB DDR2 400MHZ Single Ranked DIMMs (4 GB System / 16 GB RAM Disk)
> Hard Drive:  Two 73GB, U320, SCSI, 1IN 15K Configured in a RAID 1 (Mirrored)
> Hard Drive Controller:  Embedded RAID - PERC4 Integrated (Driver:
> megaraid_mm, megaraid_mbox)
> Everything else:
> http://www.dell.com/downloads/global/products/pedge/en/PE6850_specs.pdf
>
>  >- Linux kernel version
> 2.6.12-1.1376_FC3smp (Fedora Core 3).
>
>  >- running Xwindows?
> No.
>
>  >- Asterisk version
> ABE-A.2-beta (Asterisk Business Edition A.2 beta).
>
>  >- kind of calls you are recording (Zap, SIP, IAX, Meetme, ...)
> Calls originate on the PSTN and are handled by a Cisco AS5400 Universal
> Gateway that is a SIP peer of Asterisk. The AS5400 converts the calls
> from TDM channels to VoIP (SIP) channels before sending them to
> Asterisk.  The Asterisk dialplan then routes them to one of our agents,
> who are using SNOM 320 VoIP (SIP) phones.  Essentially all of our calls
> are SIP-to-SIP, with absolutely no protocol bridging or transcoding
> occurring on the Asterisk server.
>
> The Asterisk server handles the following major tasks:
>
> - Routing calls through the dialplan to (dynamic) agents in the
> appropriate queues.
> - Adding/removing agents to/from queues via AddQueueMember and
> RemoveQueueMember (NO static agents!).
> - Recording calls via the Monitor application directly to RAM disk.
> Calls are moved to a remote machine for mixing.
> - ChanSpy-based quality assurance of calls.  Neither ChanSpy nor the
> quality of the calls themselves is affected by the problem.
>
>  >- how many recordings at once
> Anywhere from 5 to 30 concurrent recordings.  This is not our planned
> peak, but it's where we've experienced the problem so far.  We have not
> yet determined if the number of concurrent recordings is an issue, but
> we are considering it.  We also haven't determined if the problem gets
> worse as the number of recordings increases, but it definitely exists
> throughout that entire range.
>
>  >In my experience, HyperThreading does not cause recording problems,
>  >it's usually a disk issue. When we had issues, switching to fast SCSI
>  >drives on a MegaRAID 320-1 with the megaraid2 linux driver solved all
>  >of our problems(skips and clicks/pops)
>
> The disk issues also directly interfere with call quality, as our
> previous scalability tests showed.  Digium seems to think that the issue
> is scaling (some resource contention that causes a bit of audio to be
> unavailable when the write occurs).  I see their point, but given our
> hardware and the current call volume I'm not completely sold on it.
> Could it be a configuration issue (file handles, interrupts, etc...)?
>
>  >MATT---
>
> Colin Anderson wrote:
>
>  >Matt Roth: FWIW, I am recording 1000-1500 calls a day (as of 8:52PM today,
>  >1482 calls!) of various length on my Netfinity with the onboard IBM RAID
>  >controller in RAID 5 Ultra 320 SCSI with suprisingly good quality. As the
>  >other Matt indicated, maybe what is needed here is an intelligent
> controller
>  >to offload some of the chore.
>  >
>  >No definite solution here, but at least it's another data point to
> compare.
>
> I appreciate any information contributed by list users. It's by far the
> most valuable resource available to me.
>
>  >On 12/12/05, Matt Roth <mroth at imminc.com> wrote:
>  >
>  >>List users,
>  >>
>  >>I'm using the Monitor application to record calls.  Most of the
>  >>recordings are audible, but contain skips accompanied by a popping
>  >>sound.  Sometimes they are isolated, sometimes they appear in groups.
>  >>Call quality is excellent and seems unaffected by whatever is causing
>  >>this problem.
>  >>
>  >>If anyone has experienced this problem before, I'd appreciate if you'd
>  >>share what the source was and any tips on eliminating it.  I contacted
>  >>Digium tech support and they suggested turning off hyperthreading.  I
>  >>have done that, but I won't know if it improved things until tomorrow.
>  >>
>  >>The machine is running at a moderate call volume and is always at least
>  >>90% idle.  I'm not seeing any "Avoided deadlock" messages in the logs.
>  >>If you need any more information, I'd be happy to provide it.
>
> Thank you,
>
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
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