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
--
Kai Militzer WESTEND GmbH | Internet-Business-Provider
Technik CISCO Systems Partner - Authorized Reseller
Lütticher Straße 10 Tel 0241/701333-14
[EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879
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