[asterisk-bugs] [Asterisk 0012570]: Changing of RTP SSRC between early-session and actual call session
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon May 5 09:04:19 CDT 2008
The following issue has been RESOLVED.
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http://bugs.digium.com/view.php?id=12570
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Reported By: fcastellano
Assigned To: file
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Project: Asterisk
Issue ID: 12570
Category: Core/RTP
Reproducibility: always
Severity: minor
Priority: normal
Status: resolved
Asterisk Version: 1.4.19
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
Resolution: no change required
Fixed in Version:
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Date Submitted: 05-02-2008 05:52 CDT
Last Modified: 05-05-2008 09:04 CDT
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Summary: Changing of RTP SSRC between early-session and
actual call session
Description:
Upgrading Asterisk from 1.4.17 to 1.4.19 I noticed some SIP clients (for
example Siemens Gigaset C450IP), experienced a silence period (4/5 secs)
after the remote party (on PSTN) answered. Looking at the RTP packets, I
concluded this is relate to the change of SSRC between the early session
established with 183 Session Progress, and the actual call session
following the 200 Ok message.
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file - 05-05-08 09:04
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This has already been changed.
Issue History
Date Modified Username Field Change
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05-05-08 09:04 file Assigned To => file
05-05-08 09:04 file Note Added: 0086403
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