OK. I am on my way to find out what you've suggested to isolate my issues.<div><br></div><div>I already tried latest GIT version. I will be deploying it shortly.</div><div><br></div><div>I should get back to this post with my findings.</div>
<div><br></div><div>Thank you.<br><br><div class="gmail_quote">On Wed, Feb 23, 2011 at 11:54 AM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
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.<div><br></div><div>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:</div>
<div><br></div><div>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? </div>
<div><br></div><div>Are you running the latest git of FS?</div><div><br></div><div>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?</div>
<div><br></div><div>Are you having sound quality issues in one direction or both directions? Have you done a tcpdump of the traffic and analyzed in wireshark?</div><div><br></div><div>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. :)</div>
<div><br></div><font color="#888888"><div>-MC</div></font><div><div></div><div class="h5"><div><br><div class="gmail_quote">On Wed, Feb 23, 2011 at 9:31 AM, Malay Thakershi <span dir="ltr"><<a href="mailto:mthakershi@gmail.com" target="_blank">mthakershi@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I don't use Sipura. I use FS to make / receive calls from mobile phones / regular land line phones.<div><br></div><div>Unlike what I said in my previous email, I am still using Allison-8kHz voice. Somehow in managed code I had Allison-16kHz specified.</div>
<div><br></div><div>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".</div>
<div><br></div><div>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?</div><div><br></div><div>Thank you for replies.</div>
<div><br></div><font color="#888888"><div>Malay</div></font><div><div></div><div><div><br></div><div><div><br><div class="gmail_quote">On Wed, Feb 23, 2011 at 10:55 AM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Depends, are you using a sipura? if so, try it, the setting is on the<br>
web ui of the phone/device not in FS.<br>
<div><div></div><div><br>
<br>
On Tue, Feb 22, 2011 at 12:25 PM, Malay Thakershi <<a href="mailto:mthakershi@gmail.com" target="_blank">mthakershi@gmail.com</a>> wrote:<br>
> I found I am already using 16 KHz profile. .SetTtsParameters("cepstral",<br>
> "Allison-16kHz");<br>
> I read this under FS wiki on Cepstral under 'Gotchas':<br>
> -------------<br>
> Using a 16khz voice and 0.03 for RTP Packet Size (Sipura Setting), it will<br>
> sound horrible. Workaround: modify RTP Packet Size to 0.02 in Sipura config,<br>
> under Advanced/SIP section.<br>
> -------------<br>
> Do you think that is my problem? Is this to be done in FS configuration?<br>
> Malay<br>
> On Mon, Feb 21, 2011 at 2:35 PM, Michael Collins <<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>> wrote:<br>
>><br>
>> It depends on why there is choppy audio. My guess is that going to 16k<br>
>> won't help. You should update to latest git and re-test, preferably on a<br>
>> system that is not in production. See if you can narrow down the conditions<br>
>> under which the audio is not good. Does it happen when the system is under<br>
>> load? Does it happen on every call, or only on certain calls? Things like<br>
>> that.<br>
>> -MC<br>
>><br>
>> On Sun, Feb 20, 2011 at 12:51 PM, Malay Thakershi <<a href="mailto:mthakershi@gmail.com" target="_blank">mthakershi@gmail.com</a>><br>
>> wrote:<br>
>>><br>
>>> Hello,<br>
>>> I use Cepstral in my mod_managed FS application. I mainly use<br>
>>> Session.Speak and PlayAndGetDigits in my code to play WAV / audio text.<br>
>>> When I started using FS and got a stable program running, I used Cepstral<br>
>>> Allison 8 KHz voice. But frequently I get choppy type of sound. Earlier it<br>
>>> was acceptable but now some callers seem to have difficulty understanding<br>
>>> the call audio.<br>
>>> Would it help if I get 16 KHz sounds / Cepstral license? What are changes<br>
>>> I would need to make?<br>
>>> Thank you for any help.<br>
>>> Malay Thakershi<br>
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