[asterisk-bugs] [Asterisk 0018833]: SIP over TCP and TLS does not appear to support NAT=yes
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Feb 18 02:29:42 CST 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18833
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Reported By: Cabel McCoy
Assigned To:
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Project: Asterisk
Issue ID: 18833
Category: Channels/chan_sip/TCP-TLS
Reproducibility: always
Severity: block
Priority: normal
Status: new
Asterisk Version: 1.8.2.3
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2011-02-17 16:55 CST
Last Modified: 2011-02-18 02:29 CST
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Summary: SIP over TCP and TLS does not appear to support
NAT=yes
Description:
I am using a soft client to test SIP over TCP and I am having problems
getting RTP to send traffic back to my NATed ip address. Nat works
perfectly with UDP but not with TCP or TCP/TLS.
Here is the output from my RTP Debug
TCP connection gives me this
Sent RTP packet to 192.168.49.16:10004 (type 09, seq 043932, ts
108160, len 000170)
Which never gets back to me because its my private IP.
UDP connection gives me this
Sent RTP packet to 99.137.230.28:10000 (type 09, seq 015177, ts
000320, len 000170)
Works great.
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(0132112) wdoekes (reporter) - 2011-02-18 02:29
https://issues.asterisk.org/view.php?id=18833#c132112
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I could be wrong, but it looks like the SDP in the UDP case is already
"fixed"/rewritten by an ALG/SBC/router.
I would expect to see the same 192-address in the UDP case, and the RTP
stream to be correctly matched first after traffic *from* your LAN arrives
on the asterisk.
Does any packet *from* your LAN ever arrive in the TLS case? (rtp set
debug on)
What happens with the UDP case if you disable SIP/SDP rewriting in your
router?
Issue History
Date Modified Username Field Change
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2011-02-18 02:29 wdoekes Note Added: 0132112
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