[asterisk-bugs] [Asterisk 0012670]: some RTP packets sent to NAT IP instead of public IP; breaks built-in jitterbuffer on some phones
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon Jun 2 17:55:13 CDT 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=12670
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Reported By: jolan
Assigned To:
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Project: Asterisk
Issue ID: 12670
Category: Core/RTP
Reproducibility: sometimes
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.4.19-rc3
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 05-16-2008 15:55 CDT
Last Modified: 06-02-2008 17:55 CDT
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Summary: some RTP packets sent to NAT IP instead of public
IP; breaks built-in jitterbuffer on some phones
Description:
I am experiencing the same symptoms as
http://bugs.digium.com/view.php?id=0012566 but with 1.4.20-rc3, not
trunk. The bug was fixed in trunk, but not in 1.4.x.
To reiterate the problem:
Phone 100 calls phone 102. Phone 102 answers and starts counting "1 2 3 4
5". Phone 100 doesn't hear anything until "3" or "4".
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jolan - 06-02-08 17:55
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Looking at packet number 975 in the badnat.pcap again:
Ethernet II, Src: Cisco_c2:f9:4e
f: "7940 g711" <sip:100 at 209.242.35.6>;tag=as4f924dfb
t: <sip:102 at 10.0.0.96:5060>;tag=429ED0C-8F026405
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.1.0313
Am I reading that right? A packet from the Cisco has the User-Agent set
to Polycom? I also don't understand why there are SIP compact headers.
Neither the Cisco, the Polycom, or Asterisk is configured to use compact
headers.
Issue History
Date Modified Username Field Change
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06-02-08 17:55 jolan Note Added: 0087688
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