*however*, if I use some SIP hardware, such as a Grandstream 236 or an IP phone (still use alaw just like Xten and SJ), the quality is great, even from halfway around the world. Literally.
This leads me to think that the softphones are doing something not as well as the hardware SIP devices. Anyone have any thoughts on that? I've seen this behavior with multiple client computers, so I don't think it's just the computer that's using the softphone that's to blame...
Paul
----- Original Message ----- From: "Bruno Hertz" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]>
Sent: Saturday, December 18, 2004 4:37 PM
Subject: Re: [Asterisk-Users] Asterisk Crackly Bad quality
On Sat, 2004-12-18 at 14:55 -0600, Steven Critchfield wrote:
I highly suggest you work on getting either the RTC or USB driver loaded to provide timing if you don't already have a PSTN card for that job.
OK, this is all softphones and one AVM passive BRI card here, so no digium hardware. And frankly, I'd be rather surprised if asterisk, apart from the standard kernel rtc timer, needs a special timer just to play back the demo voice and send it over the LAN. Remember, it's the initial setup we're talking about, and only the demo playback.
To make sure, I compiled and loaded the ztdummy driver (from zaptel dir for 2.6 kernel). No difference.
Also, if it really was the timer, that would hardly explain why e.g. FC3 and Debian Sarge behave so (wildly) different. I admit though that strange things happen sometimes :)
So no, the dummy driver didn't do it.
Thanks for your hints, Bruno.
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