[Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!

Fri Sep 2 12:07:00 MST 2005

So, can no one give me any suggestions? Perhaps I can elaborate upon 
further testing and attempts to debug this tremendously frustrating problem.

My softphone (typically IAXComm, but same results connecting via SIP on 
Xten Xlite)  is installed on a P4 1.6 w/ 256 megs of RAM and an 
integrated sound chipset (Intel/AC97). I've had some problems with this 
chipset in Linux, and it doesn't support hardware mixing, so I've had to 
attempt to get dmix and ALSA running in an acceptable fashion; needless 
to say, I still have problems, and I don't know if this is related.

I can record and playback my own voice and other audio (podcasts, Net 
radio, and music) fine with this headset (some cheap, Chinese $15 
headset). However, when it comes to receiving decoded audio on the other 
end of a VoIP conversation, it sounds "scratchy," distorted, crackly, 
whatever you want to call it. It's not the clarity so much as it is the 
other things I just mentioned. It's very hard ot put into words, but I'm 
hoping *someone* can associate with my problem.

To make sure it wasn't my Asterisk box that was mucking things up, I 
connected directly with my softphone to my outgoing VoIP terminators, 
voipjet and Voxee. Sure enough, it sounded *exactly* the same as it does 
going through my * box on the LAN and then out through the public Net to 
voipjet and/or Voxee. Thus, I know it is my side of the equation that is 
mucking things up, but I cannot for the life of me pinpoint exactly 
*WHERE* this is taking place.

Actually, I also unplugged my headphones and plugged back in the 
speakers, and it sounded roughly the same, but it's harder to tell 
because they're not surrounding your ears, quality isn't as clear, and 
ambient noise can be more easily heard and is distracting.

I'm thinking, at this point, that it's my sound card that's messing 
things up, or its configuration or something. However, can anyone 
explain *why* non-VoIP-conversations sound perfect on my speakers and 
headset, while VoIP calls sound very bad?

Or, am I totally overreacting, should I expect this as the standard for 
softphone calls with VoIP, and should I just get a hardphone and stop 
worrying about it? However, the geek in me really wants to pinpoint the 
source of this problem!

Please help, all, as this problem is occupying me for days on end and I 
can't get anything else done. :-) I really want to figure out why I hear 
scratchiness, skipping, and general lack of clarity on the other side. 
*Please note that I can call in to my PSTN number on the Asterisk system 
and hear the demo (Allison) pretty much perfectly, so it's definitely 
not an Asterisk problem!*

Regards,
Robert Geller

Robert Geller wrote:

> UPDATE:
> I've been advised by users on #asterisk (IRC) that this is standard 
> for softphones in general, and that if I were to use a hardphone, 
> quality would be significantly better. Is this the case? Are 
> softphones that much inferior to hardphones? That might make sense to 
> me, as they have to go through the sound card of your computer and 
> then out through -- and with -- all the other Net traffic from the 
> computer you're using the softphone on. Again, is this the case, or 
> should there be little difference between a soft- and hardphone?
>
> Also, regarding the Monitor() command, I wanted to see what the 
> quality would be like on a recorded and played-back conversation, as I 
> thought maybe that would clue me in on some of the problems, but it 
> sounded pretty similar to how it sounded to me on the headset when I 
> was talking (this was a conversation with Newegg.com's tech support). 
> Can anyone tell me why that is? I don't really know how the Monitor() 
> command works (I mean, I understand the concept, but not /how/ it 
> actually goes about recording the channel(s). Would it be expected 
> that you would hear the same quality from the other side if you listen 
> to a Monitor()ed conversation?
>
> Thanks a lot, all.
>
> Robert Geller wrote:
>
>> Hello all,
>>
>> I am using a headset and the X-lite softphone (sometimes I use 
>> IAXComm, but I'm having difficulties using OSS emulation with it) to 
>> connect via uLaw to my internal Asterisk server, which is a Pentium 
>> II 400 with 128 megs of RAM. After getting this headset, most or all 
>> of the echo people on the other line were complaining about is now 
>> gone, according to them. However, every five to ten seconds, I get 
>> quick "skipping" or lag on the other side, so that the person whom 
>> I'm talking to's voice sounds like it "skips a beat," analogous to 
>> when a CD you're listening to skips quickly.
>>
>> I don't think -- but am not positive -- that it is a question of 
>> insufficient bandwith, as I am on a Cox 5mbps/2mbps cable line that 
>> is very reliable and pretty stable. I believe I am using uLaw both to 
>> the Asterisk server /and/ from the Asterisk server to Voxee, my 
>> outgoing SIP provider/PSTN terminator.
>>
>> Is this a common problem? It doesn't seem like it should be, as it is 
>> a major detriment to having enjoyable, good-quality VoIP 
>> conversations and doesn't seem like it would be the "standard" for 
>> such conversations. Perhaps I shouldn't be using uLaw, but this 
>> really bugs me because I do have the bandwith to use uLaw, and its 
>> quality is unsurpassed.
>>
>> Could this be an insufficient RAM problem with my * server? As I have 
>> 128 megs of RAM on my PII, about 122 megs of it are constantly in 
>> use, and the CPU is, for the most part, pretty idle during single 
>> conversations. I'm not sure about incoming calls when music, etc., is 
>> played, but I'm not talking about that right now -- just the skipping 
>> I'm getting when I make outgoing SIP calls to Voxee (and, ultimately, 
>> to the PSTN). Would the problem be resolved with more RAM? This is an 
>> old Compaq Deskpro that I'm using as an Asterisk server (not much 
>> else is running, but I haven't specifically optimized it) on Debian, 
>> and I don't even know what type of RAM it takes and can find no 
>> documentation to tell me.
>>
>> The problem exists even when I make internal SIP calls, i.e. to 
>> voicemail (Comedienne mail, is it?) and other test extensions. 
>> Allison, the voice of Asterisk, asks for your mailbox, but it isn't a 
>> continuous flow; instead, it skips: "Maaa-aa-aailbox".
>>
>> Something is definitely wrong, and I eagerly await advice and the key 
>> to making crisp, clear VoIP/PSTN calls -- free of this extremely 
>> annoying skipping!
>>
>> Regards,
>> Robert Geller
>> _______________________________________________
>> --Bandwidth and Colocation sponsored by Easynews.com --
>>
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>