[Asterisk-Dev] libiax2 OK for production?

Michael Van Donselaar michael-list at vandonselaar.org
Tue Feb 1 13:34:39 MST 2005


On Mon, 31 Jan 2005 20:00:11 -0500, Steve Kann <stevek at stevek.com> wrote:

>Steve Underwood wrote:
>
>> Bruno Hertz wrote:
>>
>>> On Mon, 2005-01-31 at 18:41 -0500, Steve Kann wrote:
>>>
>>>  
>>>
>>>> What kind of latency are you talking about here?  There's about 60ms 
>>>> of latency, on the output side (i.e. to the speakers) in iaxclient 
>>>> proper that could be removed if we changed the architecture a bit 
>>>> [removing the buffers in-between decoding and the audio layer], but 
>>>> if you're seeing a _lot_ of latency, then there's probably some 
>>>> particular issue that can be resolved.
>>>>   
>>>
>>>
>>> About one to two seconds on Fedora Core 3. On my system, it just can't
>>> compete. Of all clients I tried, linphone, sjphone, cornfed,
>>> gnomemeeting, ... iaxclient is worst latency wise. So it might be that
>>> particular issue, but I myself see no easy way pinpointing it.
>>>
>>> Reinventing the wheel, not really I guess. Portability is not my
>>> concern, and as said miniphone already does what I want at a basic
>>> level. The rest, like codec selection and comfort stuff, is just
>>> fun hacking I'd say.
>>>
>>> Thanks, Bruno.
>>>  
>>>
>> This came up before. iaxcomm itself doesn't give such long delays, but 
>> some sound card drivers do. My desktop machine give little latency. My 
>> notebook gives a second or more. I think your sound card is most 
>> likely the source of this latency. However, the fact you get better 
>> results with miniphone means it is not a fundamental problem with the 
>> card and driver (I assume you tried these tests on the same machine, 
>> or they are meaningless). I believe miniphone and iaxcomm use 
>> different ways to drive the sound card. Maybe iaxcomm is doing 
>> something wrong. Sound cards normally work in a high latency mode, 
>> which provides lots of buffering, and have to be switched to low 
>> latency by the application. Maybe it isn't done correctly in iaxcomm 
>> for all cards, or maybe there is some OSS/ALSA conflict. Do you know 
>> what sound card hardware you have, and which driver?
>
>
>I haven't examined all the code, but I'm pretty sure the major 
>parameters of what they're doing should be similar:
>
>They both use OSS
>They both set the card to 8khz mono
>They both set the card to full-duplex.
>
>There are other things that might be different, though..

Portaudio?

>-SteveK
>
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