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
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