[asterisk-bugs] [Asterisk 0016323]: confusing description in configs/sip.conf.sample

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Nov 25 16:14:00 CST 2009


The following issue has been SUBMITTED. 
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https://issues.asterisk.org/view.php?id=16323 
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Reported By:                klaus3000
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16323
Category:                   Channels/chan_sip/General
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.1.10 
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:             2009-11-25 16:14 CST
Last Modified:              2009-11-25 16:14 CST
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Summary:                    confusing description in configs/sip.conf.sample
Description: 
Hi! There is a confusing text in sip.conf when describing the NAT options:

externip/externhost/stunaddr

;   Note that at the moment all these mechanism work only for the SIP
socket.
;   The IP address discovered with externip/externhost/STUN is reused for
;   media sessions as well, but the port numbers are not remapped so you
;   may still experience problems.
;
; NOTE 1: in some cases, NAT boxes will use different port numbers in
; the internal<->external mapping. In these cases, the "externip" and
; "externhost" might not help you configure addresses properly, and you
; really need to use STUN.
;
; NOTE 2: when using "externip" or "externhost", the address part is
; also used as the external address for media sessions.
; If you use "stunaddr", STUN queries will be sent to the same server
; also from media sockets, and this should permit a correct mapping of
; the port numbers as well.

The first paragraph is contradictory to NOTE 2. Note 2 tells that
"stunaddr" also uses STUN on the media port to announce the proper socket
in the SDP, but the first paragraph says that ALL mechanisms do only take
care of SIP.
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Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-11-25 16:14 klaus3000      New Issue                                    
2009-11-25 16:14 klaus3000      Asterisk Version          => 1.6.1.10        
2009-11-25 16:14 klaus3000      Regression                => No              
2009-11-25 16:14 klaus3000      SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
======================================================================




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