[asterisk-bugs] [Asterisk 0018833]: SIP over TCP and TLS does not appear to support NAT=yes

[Asterisk Bug Tracker]

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 09:51 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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 (0132143) Cabel McCoy (reporter) - 2011-02-18 09:51
 https://issues.asterisk.org/view.php?id=18833#c132143 
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Yes both phone and asterisk are behind NAT at different locations. I tried
to disable the ALG in my router and it appears the SDP is getting
re-written somewhere else, probably in my DSL router, which I have no
control over. Do you know if it would be possible to re-write the SDP with
the SIP_TRANSPORT_TCP address if NAT is set to yes? I think this would
solve my issue with double nat and TLS/TCP connections.
So what your saying is that in the UDP case the RTP stream waits for
packets from my client and if they arrive then it uses that source address?
If so how would this work for TCP since its not stateless? I will start a
packet capture on my client and on the asterisk box to see what happens to
the RTP packets on UDP and TCP.

Thanks for your help! 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-18 09:51 Cabel McCoy    Note Added: 0132143                          
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