[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 10:05:51 CST 2011
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=18833
======================================================================
Reported By: Cabel McCoy
Assigned To:
======================================================================
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:
======================================================================
Date Submitted: 2011-02-17 16:55 CST
Last Modified: 2011-02-18 10:05 CST
======================================================================
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.
======================================================================
----------------------------------------------------------------------
(0132145) Cabel McCoy (reporter) - 2011-02-18 10:05
https://issues.asterisk.org/view.php?id=18833#c132145
----------------------------------------------------------------------
I found my problem,
I needed to open the RTP ports on the router in front of asterisk. Looks
like the ALG was doing this for me on UDP but it cannot see into the
SIP/TLS packets to do it. When I manually opened these ports my problem
went away. Thanks for pointing me in the right direction with RTP.
Issue History
Date Modified Username Field Change
======================================================================
2011-02-18 10:05 Cabel McCoy Note Added: 0132145
======================================================================
More information about the asterisk-bugs
mailing list