[Freeswitch-users] Using 16 KHz sounds

Fri Feb 25 20:17:51 MSK 2011

OK. I am on my way to find out what you've suggested to isolate my issues.

I already tried latest GIT version. I will be deploying it shortly.

I should get back to this post with my findings.

Thank you.

On Wed, Feb 23, 2011 at 11:54 AM, Michael Collins <msc at freeswitch.org>wrote:

> 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
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110225/4cc5fad8/attachment.html 