[asterisk-ss7] Version 0.8.3 of chan_ss7 for asterisk released

[Kai Militzer]

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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