[asterisk-bugs] [Asterisk 0013405]: [patch] T38 gateway
Asterisk Bug Tracker
noreply at bugs.digium.com
Sun Mar 7 17:02:14 CST 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=13405
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Reported By: dafe_von_cetin
Assigned To:
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Project: Asterisk
Issue ID: 13405
Category: Applications/app_fax
Reproducibility: N/A
Severity: feature
Priority: normal
Status: confirmed
Asterisk Version: SVN
JIRA: SWP-115
Regression: No
Reviewboard Link: https://reviewboard.asterisk.org/r/459/
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 140548
Request Review:
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Date Submitted: 2008-08-30 16:44 CDT
Last Modified: 2010-03-07 17:01 CST
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Summary: [patch] T38 gateway
Description:
Hi all,
I'm sending you patch containing new application app_faxgateway.c
("FaxGateway") which is able to mediate T30 to T38 and vice versa.
Feature is using spands library (I used spandsp-0.0.4pre18 and
spandsp-0.0.5pre4).
Best regards
Daniel.
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(0119072) klaus3000 (reporter) - 2010-03-07 17:01
https://issues.asterisk.org/view.php?id=13405#c119072
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I now managed to have t38 gatewaying working for DAHDI->SIP. I also tried
SIP->SIP but it fails due to frame timing. Current sending of voice frames
is triggered by received voice frames - thus if the non-t38 client
(Asterisk using SendFAX) does not send RTP then the t38-gateway also does
not send voice frames.
I tried to modify the event loop to regularly check t38_gateway_tx() for
new voice data to send, but t38_gateway_tx() always responds with a full
voice-data bufer even it is called every ms.
So, does anybody know how t38_gateway_tx() works (no documentation :-()?
What is the preferred way to decouple frame generation from RTP reception
(I think the t38-gateway should send RTP every 20ms (or whatever the frame
rate is) also if it does not receive audio)?
Issue History
Date Modified Username Field Change
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2010-03-07 17:01 klaus3000 Note Added: 0119072
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