Less Good Call Quality Using Asterisk

Stefan at July 21, 2012 Asterisk Users 8 Comments

Hello,

I am currently using the following setup:
Snom 300 – Asterisk 1.8.13.0 running on Raspberry Pi – Sipgate SIP Provider When I am using this setup, the call quality isn’t as good as when using a direct connection like Snom 300 – Sipgate SIP Provider to my SIP Provider (Sipgate).

Sipgate supports

0×08 (G.711), so the best one possible. Still the quality isn’t as good as on a direct connection between my Snom phone and Asterisk. Using Asterisk it feels a little bit like there is some more background noise or so. Any hints why the quality using Asterisk is less good than on a direct connection? Does Asterisk by default change anything on the signal or is it just passed through? Thanks :-)

Best regards Stefan

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8 Responses to “Less Good Call Quality Using Asterisk”

  1. Doug Lytle said:

    Jul 21, 12 at 2:13 pm

    Stefan at WPF wrote:

    Not that I can help, but I’m sorta shocked that you have Asterisk running on a Raspberry Pi!

    Very cool!

    Doug

  2. Mike said:

    Jul 21, 12 at 2:19 pm

    I’ve had it running on a Guruplug without any problems -
    http://www.globalscaletechnologies.com/p-49-guruplug-server-standard.aspx#summary
    - handly little device if you need something small.

  3. Stefan at said:

    Jul 23, 12 at 10:57 am

    –14dae9340afb9964e204c5814ace Content-Type: text/plain; charset=ISO-8859-1

    For private use the RPI is really cool for that, though I am not yet sure if it works 100% without problems – at least it did in my latest tests. Anyone has any hint on the call quality or if Asterisk does any kind of transcoding of the audio?

    2012/7/21 Mike

    –14dae9340afb9964e204c5814ace Content-Type: text/html; charset=ISO-8859-1
    Content-Transfer-Encoding: quoted-printable

    For private use the RPI is really cool for that, though I am not yet sure i=
    f it works 100% without problems – at least it did in my latest tests.

    =
    Anyone has any hint on the call quality or if Asterisk does any kind of tra=
    nscoding of the audio?

    2012/7/21 Mike < =3D"mailto:ispbuilder@gmail.com" target=3D"_blank">ispbuilder@gmail.com=
    >
    ;border-left:1px #ccc solid;padding-left:1ex">
    On 12-07-21 04:13 PM, Doug Lytle wrote:

    x #ccc solid;padding-left:1ex">

    Stefan at WPF wrote:

    x #ccc solid;padding-left:1ex">

    Snom 300 – Asterisk 1.8.13.0 running on Raspberry Pi

    Not that I can help, but I'm sorta shocked that you have Asterisk runni=
    ng on a Raspberry Pi!

    I've had it running on a Guruplug without any problems – p://www.globalscaletechnologies.com/p-49-guruplug-server-standard.aspx#summ=
    ary" target=3D"_blank">    http://www.globalscaletechnologies.com/p-< =
    /u>49-guruplug-server-standard.    aspx#summary – handly little devi=
    ce if you need something small. 8">

    –
    Looking for (employment|contract) work in the
    Internet industry, preferrably working remotely.
    Building / Supporting the net since 2400 baud was
    the hot thing. Ask for a resume! rget=3D"_blank">ispbuilder@gmail.com


    –
    _____________________________________________________________=
    ________
    – Bandwidth and Colocation Provided by om" target=3D"_blank">http://www.api-digital.com –
    New to Asterisk? Join us for a live introductory webinar every Thurs:
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 t=3D"_blank">http://www.asterisk.org/hello

    asterisk-users mailing list
    To UNSUBSCRIBE or update options visit:
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    users

    –14dae9340afb9964e204c5814ace

  4. Kevin P. said:

    Jul 23, 12 at 11:07 am

    If both legs of the call are using the same codec, then normally Asterisk would not modify the audio in any way at all.

  5. Bakko said:

    Jul 23, 12 at 11:11 am

    Hello,

    I tried Asterisk Confbridge with raspberry pi without audio issue.

    Asterisk was compiled from sources.

    http://www.voztovoice.org/?q=node/553

    Regards

  6. Stefan at said:

    Jul 25, 12 at 6:42 am

    Hmm is it possible, that the monitor command changes the quality? If not I
    guess I also once have to try compiling it from source, though I wanted to avoid that.

    2012/7/23 Bakko

  7. Kevin P. said:

    Jul 25, 12 at 9:39 am

    It certainly can, since recording the call causes disk I/O as the audio is written out. In addition, Monitor is more prone to this problem than MixMonitor is, because Monitor’s call recording is done in the same thread that handles the call’s audio normally. If you switch to MixMonitor, you’ll probably have better results, unless your system just can’t handle recording the call without overloading its CPU.

  8. Stefan at said:

    Aug 01, 12 at 7:21 am

    Thank you Kevin, I have the impression it got better, but I yet didn’t have the chance to test it much. If there are I/O problems, are there any logs for this / are there logs in general for monitor and MixMonitor? Thanks :-)

    2012/7/25 Kevin P. Fleming