[asterisk-bugs] [Asterisk 0012902]: Video RTP is not sended to originating SIP extension when using IAX2 to interconnect both servers
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Jul 13 10:37:18 CDT 2009
The following issue has been UPDATED.
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https://issues.asterisk.org/view.php?id=12902
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Reported By: albersag
Assigned To:
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Project: Asterisk
Issue ID: 12902
Category: Channels/chan_iax2
Reproducibility: always
Severity: major
Priority: normal
Status: closed
Target Version: 1.4.27
Asterisk Version: 1.4.18
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: suspended
Fixed in Version:
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Date Submitted: 2008-06-20 02:46 CDT
Last Modified: 2009-07-13 10:37 CDT
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Summary: Video RTP is not sended to originating SIP extension
when using IAX2 to interconnect both servers
Description:
When i use IAX2 to interconnect two servers, Asterisk does not send RTP in
H.263,263+ or H.264 to the originating SIP extension of the call.
It´s always reproducible and it always happens in originating extension.
Other party could see you but you could not see him/her.
Analyzing RTP/SIP captures, i see Asterisk1 (from where i originate SIP
Video Call) does not receive RTP packets in video códec sended by client
SIP2 in other Asterisk2
It does not depend on Softhphone or Hardphone used. We had same problem
with Grandstream GXV3000, or instead Eyebeam Softphone.
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Relationships ID Summary
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related to 0008824 [patch] Remote (called) Party Identific...
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(0107677) lmadsen (administrator) - 2009-07-13 10:37
https://issues.asterisk.org/view.php?id=12902#c107677
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I'm closing this for now due to lack of response. If someone can provide
some additional information here, please feel free to reopen the issue.
Thanks!
Issue History
Date Modified Username Field Change
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2009-07-13 10:37 lmadsen Note Added: 0107677
2009-07-13 10:37 lmadsen Status feedback => closed
2009-07-13 10:37 lmadsen Resolution open => suspended
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