[Freeswitch-users] Using 16 KHz sounds

Wed Feb 23 20:54:22 MSK 2011

It sounds very much like you have standard 8k calls. Increasing the sampling
rate won't help since FS would have to downsample to 8k on the call leg
anyway.

It's time to go back to the original issue. You are having sound quality
issues, correct? It's time to roll up your sleeves and do some detective
work:

What kind of network are you running? What routers, switches, NAT devices,
and other computers are using the network? What kind of system is FS running
on? Any virtualization being used? What OS?

Are you running the latest git of FS?

What SIP clients have you tried? Can you reproduce the sound quality issues
on all of your SIP clients? Can you reproduce with different SIP clients on
a different computer? Do you have a hard phone and do the symptoms persist
there?

Are you having sound quality issues in one direction or both directions?
Have you done a tcpdump of the traffic and analyzed in wireshark?

Those are all questions worth pursuing. The idea is to narrow the symptoms
down as much as possible. I know it's not fun, but then again, this is
telephony. :)

-MC

On Wed, Feb 23, 2011 at 9:31 AM, Malay Thakershi <mthakershi at gmail.com>wrote:

> I don't use Sipura. I use FS to make / receive calls from mobile phones /
> regular land line phones.
>
> Unlike what I said in my previous email, I am still using Allison-8kHz
> voice. Somehow in managed code I had Allison-16kHz specified.
>
> I created three WAV files using Cepstral SWIFT command with 8000, 16000,
> 22000 Hz.  When I play each file, the later two give me message at the FS
> console "Sample rates don't match".
>
> Is there a setting where I can ask FS to sample at a higher rate that would
> help me with sound quality issues? Is having a good sound card on the server
> a good practice?
>
> Thank you for replies.
>
> Malay
>
>
> On Wed, Feb 23, 2011 at 10:55 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Depends, are you using a sipura? if so, try it, the setting is on the
>> web ui of the phone/device not in FS.
>>
>>
>> On Tue, Feb 22, 2011 at 12:25 PM, Malay Thakershi <mthakershi at gmail.com>
>> wrote:
>> > I found I am already using 16 KHz profile. .SetTtsParameters("cepstral",
>> > "Allison-16kHz");
>> > I read this under FS wiki on Cepstral under 'Gotchas':
>> > -------------
>> > Using a 16khz voice and 0.03 for RTP Packet Size (Sipura Setting), it
>> will
>> > sound horrible. Workaround: modify RTP Packet Size to 0.02 in Sipura
>> config,
>> > under Advanced/SIP section.
>> > -------------
>> > Do you think that is my problem? Is this to be done in FS configuration?
>> > Malay
>> > On Mon, Feb 21, 2011 at 2:35 PM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> >>
>> >> It depends on why there is choppy audio. My guess is that going to 16k
>> >> won't help. You should update to latest git and re-test, preferably on
>> a
>> >> system that is not in production. See if you can narrow down the
>> conditions
>> >> under which the audio is not good. Does it happen when the system is
>> under
>> >> load? Does it happen on every call, or only on certain calls? Things
>> like
>> >> that.
>> >> -MC
>> >>
>> >> On Sun, Feb 20, 2011 at 12:51 PM, Malay Thakershi <
>> mthakershi at gmail.com>
>> >> wrote:
>> >>>
>> >>> Hello,
>> >>> I use Cepstral in my mod_managed FS application. I mainly use
>> >>> Session.Speak and PlayAndGetDigits in my code to play WAV / audio
>> text.
>> >>> When I started using FS and got a stable program running, I used
>> Cepstral
>> >>> Allison 8 KHz voice. But frequently I get choppy type of sound.
>> Earlier it
>> >>> was acceptable but now some callers seem to have difficulty
>> understanding
>> >>> the call audio.
>> >>> Would it help if I get 16 KHz sounds / Cepstral license? What are
>> changes
>> >>> I would need to make?
>> >>> Thank you for any help.
>> >>> Malay Thakershi
>> >>> _______________________________________________
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>> >>>
>> >>
>> >>
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>> >
>> >
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>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
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