[Asterisk-Users] Skips and Pops in Call Recordings

Matt Florell astmattf at gmail.com
Tue Dec 13 12:19:31 MST 2005


Hello,

To see if it's somehow the recording throughput that's the problem I'd
suggest trying recording in GSM just as a test and see how that is.

As for the hardware, just try a machine with no Dell parts in it. I've
talked to many Asterisk users who's problems went away when they
switched to something that wasn't a Dell.

MegaRAID2 might help just because it's another reduction in the
overall data that flows over the PCI bus. It's faster and more
streamlined than the original megaraid driver and it can't hurt to try
it.

MATT---


On 12/13/05, Matt Roth <mroth at imminc.com> wrote:
>  Matt,
>
>  The calls are u-Law.  The format of the recordings is PCM.  Is this correct
> to prevent transcoding the recording?  We've noloaded all other codecs, so I
> don't believe that transcoding is occurring.  I've only ever seen "show
> translation" generate the following output:
>
>  immlx15*CLI> show translation
>           Translation times between formats (in milliseconds)
>            Source Format (Rows) Destination Format(Columns)
>
>           g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
>     g723     -     -     -     -     -     -     -     -     -     -     -
>      gsm     -     -     -     -     -     -     -     -     -     -     -
>     ulaw     -     -     -     -     -     -     1     -     -     -     -
>     alaw     -     -     -     -     -     -     -     -     -     -     -
>     g726     -     -     -     -     -     -     -     -     -     -     -
>    adpcm     -     -     -     -     -     -     -     -     -     -     -
>     slin     -     -     1     -     -     -     -     -     -     -     -
>    lpc10     -     -     -     -     -     -     -     -     -     -     -
>     g729     -     -     -     -     -     -     -     -     -     -     -
>    speex     -     -     -     -     -     -     -     -     -     -     -
>     ilbc     -     -     -     -     -     -     -     -     -     -     -
>
>
>  Any suggestions on hardware?  Are you talking the entire server or
> components?
>
>  I'll look into the megaraid2 drivers, but I'm interested in knowing how
> they come into play when recording to a RAM disk.
>
>  Matthew Roth
>  InterMedia Marketing Solutions
>  Software Engineer and Systems Developer
>
>  Matt Florell wrote:
>  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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