Hello Anton,
Anton wrote:
> When I do connect from the chan_ss7 box to the endpoint over
> g711 codec - everything is fine.
Well, that's what I do. I only use G711 Alaw as allowed codec.
> When I do connect from the chan_ss7 box with g711 codec to
> ANOTHER asterisk box via SIP, which THAN transcodes g711 to
> IPP g729 or g723 codec with remote endpoint (over satellite
> link, 600+ms) - sound is glitchy and there is an audio
> lost!!!
>
> When I do connect from the chan_ss7 box with g711 codec to a
> COMMERCIAL MVTS softswitch via SIP, which THAN transcodes
> g711 to it's own builtin g729 or g723 codec with remote
> endpoint (over satellite link, 600+ms) - sound is OK and
> there is NO AUDIOLOST!
That sounds for me more like a problem of the asterisk transcoding then
a problem with chan_ss7. I sometimes (especially with a lot of channels
open) get messages like
Mar 20 15:48:39 NOTICE[22448]: chan_ss7.c:1880 ss7_write: Write buffer
full on CIC=38 (wrote only 0 of 160), audio lost.
But this seems to have no real impact on the voice quality as there are
no glitches hearable.
> The error happens in the code which uses write() to the
> zaptel fd. Than write() returns EAGAIN and resource
> temporarily unavalable and that error happens. But
> considering the conditions given above - that is strange
> and I could only guess that there is some global
> desyncronization...
What lets you come to the conclusion that the problem lies at write()
function? Did you do debugging? If that's the case, than it is really
strange.
Best regards,
Kai
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