Re: [Asterisk-Users] Asterisk Crackly Bad quality
I'm interested in this, too. I find that when I use Xten or SjPhone software locally the quality is quite good, but when I use it remotely across the internet, I get quite a crackly response. *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. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Crackly Bad quality
On 17 Dec 2004, at 23:45, Mark Farver wrote: On Fri, 2004-12-17 at 12:01 -0700, Nihal wrote: I've freshly installed Asterisk on a Fedora C2 machine. Dual P4's, 2GB RAM. 15KRPM Drive. Using the default configs and added one Soft Sip phone. While listening to the demo the quality isnt very good. It's kind of crackly and skips a bit. Should the sound be better or is that just what you get using IP phones/Asterisk? If nothing else works, try : 1) disabling one of the P4's 2) loading a 'stock' kernel in place of the FC2 one I run * on a PIII 800 with 256Mb on 100Mb/s and it sounds in-distinguishable from a normal phone. T. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Crackly Bad quality
On Fri, 2004-12-17 at 12:01 -0700, Nihal wrote: I've freshly installed Asterisk on a Fedora C2 machine. Dual P4's, 2GB RAM. 15KRPM Drive. Using the default configs and added one Soft Sip phone. While listening to the demo the quality isnt very good. It's kind of crackly and skips a bit. Should the sound be better or is that just what you get using IP phones/Asterisk? (I ran the X-Lite phone). Hi Nihal don't know about FC2, but I can confirm that sound quality depends on the distribution/kernel used, where I didn't figure out yet why. I myself have FC3 and Debian Sarge with 2.6.8 kernel on the same machine, and under both OS I compiled and installed an identical asterix cvs version. While the FC3 installation works alright regarding sound, I see the same problems you described on Debian. I.e. the demo voice is crackling and has delays (short moments of silence). A tcpdump showed me that during those 'skips' actually no packets are transferred from the asterix server to the client machine. So somewhere on the server that voice data is buffered resp. delayed, god knows why. As said, the same asterisk version works alright with FC3 on the same machine. I even logged and compared the make runs on both OS, but there are no notable differences. Might be kernel, some external library, I have no idea actually at this point. Regards, Bruno. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Crackly Bad quality
On Sat, 2004-12-18 at 18:34 +0100, Bruno Hertz wrote: On Fri, 2004-12-17 at 12:01 -0700, Nihal wrote: I've freshly installed Asterisk on a Fedora C2 machine. Dual P4's, 2GB RAM. 15KRPM Drive. Using the default configs and added one Soft Sip phone. While listening to the demo the quality isnt very good. It's kind of crackly and skips a bit. Should the sound be better or is that just what you get using IP phones/Asterisk? (I ran the X-Lite phone). Hi Nihal don't know about FC2, but I can confirm that sound quality depends on the distribution/kernel used, where I didn't figure out yet why. I myself have FC3 and Debian Sarge with 2.6.8 kernel on the same machine, and under both OS I compiled and installed an identical asterix cvs version. While the FC3 installation works alright regarding sound, I see the same problems you described on Debian. I.e. the demo voice is crackling and has delays (short moments of silence). A tcpdump showed me that during those 'skips' actually no packets are transferred from the asterix server to the client machine. So somewhere on the server that voice data is buffered resp. delayed, god knows why. As said, the same asterisk version works alright with FC3 on the same machine. I even logged and compared the make runs on both OS, but there are no notable differences. Might be kernel, some external library, I have no idea actually at this point. I haven't seen it mentioned here yet, but do you have a timing device installed? Without a timing device such as a Digium PSTN card or a dummy interface to drive the interupts, asterisk relies on the incoming data stream. If your phone or soft phone is generating silence and uses VAD, then there will be gaps where the audio won't be sent from your phone and asterisk won't know it is time to send you more audio. 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. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Crackly Bad quality
I've freshly installed Asterisk on a Fedora C2 machine. Dual P4's, 2GB RAM. 15KRPM Drive. Using the default configs and added one Soft Sip phone. While listening to the demo the quality isnt very good. It's kind of crackly and skips a bit. Should the sound be better or is that just what you get using IP phones/Asterisk? (I ran the X-Lite phone). Nihal -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.2.3 (GNU/Linux) mQGiBD98Z1oRBACjNojsbWp/jQSa9TUD3nCwiksb1WuwM8pKZkkiDUzLTo5ase/0 l2aLs+r3pwWZ+79BDBSH7jmpM3DWeLzRDb3dunkg2Yte3YQ+nZ2MQJP7TqsAvZ7H IjXYeqT4ZosVdnj1yrKKxkvpVdnLURv5Yiolo47rhkVH5LWkxmbxYhEwRwCgiJsB jiCxw+rWpY6dDVk2BHg9WdsD/RZR6p9WM5L22lJN+71laFVJa3GYmxHcaWJ+MkeM YodQmQUgIY4lMphtMPfDXhfozGO/PPzLtC85Yhr1909dW9vxl4KYRjYpzlOgaP4G KhsFj6zMhixdcTn+FqtlnagWK4cse25ZS9MWt/5lXAum+bQjLSN+uWA1mquXvhat a6h4BACKrg2V506z2akyhx/e4/6Elg4C/bLAtxtFR/INcA22oKpQuhgm4ADa9x1p 5Wl2YCcAzpr+syiCM/xLHHJYLw5jn66qng4gE30An77U/1uxJD2udcyIOd4h/frP DdT7/oX+Br1WYjL8SkK+tOx+dvc8R1NWaGB76/DP2Ewez1/K9LQhTmloYWwgKGNs YWltLm1kKSA8bmloYWxAY2xhaW0ubWQ+iFsEExECABsFAj98Z1oGCwkIBwMCAxUC AwMWAgECHgECF4AACgkQit4xDY9sohtBkQCffXjYFKz/HyMICCbO9J7WxO5LZg4A n23l3VyERElrjW9jz4+ikUmOF6x3uQQNBD98Z8IQEACwwuStFT1lr+X/nfcyTbTE uOP6TJbuzWqnIdlo8awzjQrjXrnen+kSIGKFyjvLreueWXUFYMLXm8nrMGi+765A +/xpYHOFy5rL2ONBYiFO9SCrRKf2Pd4yZ0XTL0P808X4OKI2v20sOHFkM4xvD3em yVt8B+OigSBWR+l0NPc6POfBQH7nfNkn2tw//8pVMbSvTNaswRpNwETRRtXbXukd eONwmLPMXBooSl2zpztSrl+YXQMeQnnpelXpqah4Am0K06HIvSniY0Ne2jOy3AJ9 NQIIUzOt9yKdjFlkPWIdD3G4br2drSPP53WZiAGGhA0DCBu5sEIsQLp98qJbshC5 MG4/n97tDiFe9JhS1xaeyiBZFof+PnraVX0PI/+/Dc3SBbOm11PiWBb0w+x4fQ+e ImIErpaEEwUkmOfbVpE4vgNHnTcKc63dFyS+yHf+Z/rN1xZWFv7q/FZGHhhb7M61 9OB3BG4GhKRZ31WLj+wMikiFEoYLGxLNtRHK1a6LnbwrIBy4/GdIEoO9vURw5aut tO5pIbYrqFoHqTXmsfwtCVlljZstgqcWUtmsTTLT6acXBqWi8gR1KF98ZHRoNzlO +Wr86OkqvffpY0e4KikHqxl+f87dIaC8r8LCgdwtSiZLI/YvkL+CBkAStl8XQo84 ebzcjZrbipHnxk2GheTMQwADBQ//W7G7BPorzEfK6EQv7iiooVZsaEW4XC+mHxaX CeWyVgJnNq1TI1AF0pHS8jTcTuNj51dJtToobTA3jIjtk+/XWEu/tTpST3ADmAI7 8B9wwh4UeFfZkxL0fT/FSmSadRkIoi9RIIUc3NF9gqICeC5aiUTLc4jg0V6xQo89 yfQF6QXi1SuCoQ1vLWSmG6RtYA78KkHcDCmCx2OM45WNT8KiGr8tZgM9s5UIrusv qhRxHRlS/JzBfgpGIAvMuTbBb7XTvodQGPss9DHUzQw0kI/9Y+evtSlrEtDZNqxT GrXcYe6hNqDZCZGifHE0mXovhF4aLoYdBe37Sf5u6tdODHrlISnKbyopQWQ0OvD1 W4riKSjE9Qk8a6nuB2It2SKqk8RoYYYqe59w22G9voQBU8Y/iILOAvmMYhxHz2gr IQXa5A2LCwl8JDNJzIw9Ptc9wpxABqEq+lkNVmM2ZGqdP9GLKFyirJVP7n8YWbAT 0jUeO1WMbI8WGyZiSSgn9b+2BqZDQJF1XDYUQBUzrfex/SxwkGvNiZFe+oq5KRnU WSum9/XDx7E/k7MtrAaKOh+cIEAl3QnSbFhM3R034gK1pBr04rL8lbSH/doscW0c jnWhzjR6nX+jyKVNnBOgwuPViAdAc4VFLX8Md29142McYmiU2UaUykGaPtPCuwyu h4FjJkSIRgQYEQIABgUCP3xnwgAKCRCK3jENj2yiGwqGAJ4p99bRgKlwj3HOneUj d5iBxiVU3QCgiJgiRIJW3KHDAoU+NVaxprvP7CU= =4aZ2 -END PGP PUBLIC KEY BLOCK- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Crackly Bad quality
On Fri, 2004-12-17 at 12:01 -0700, Nihal wrote: I've freshly installed Asterisk on a Fedora C2 machine. Dual P4's, 2GB RAM. 15KRPM Drive. Using the default configs and added one Soft Sip phone. While listening to the demo the quality isnt very good. It's kind of crackly and skips a bit. Should the sound be better or is that just what you get using IP phones/Asterisk? If you're on the same LAN the quality should be pretty good, at least during the congratulatory message part. Extension 500, the call to Digium might not be very good, depending on your Internet connection. Otherwise.. you might want to check the client PC for sound quality... a lot of audio cards these days have miserable distortion if the volume on the PCM channels is turned up. You might want to try isolating the Asterisk box and client box on a separate network switch for testing purposes, to eliminate outside network problems. (Streaming audio and VOIP can be very sensitive to network conditions that do not bother less time sensitive protocols.) Its amazing how many network issues can be traced to a forgotten 10mb hub buried in a wiring closet. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
