[asterisk-bugs] [Asterisk 0014230]: [patch] Calls are not accepted from an outbound proxy

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14230 
====================================================================== 
Reported By:                Nick_Lewis
Assigned To:                putnopvut
====================================================================== 
Project:                    Asterisk
Issue ID:                   14230
Category:                   Channels/chan_sip/General
Reproducibility:            have not tried
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.6.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-01-13 10:19 CST
Last Modified:              2009-01-15 03:28 CST
====================================================================== 
Summary:                    [patch] Calls are not accepted from an outbound
proxy
Description: 
find_peer does not make a match between a received invite and a peer if the
invite comes from the outboundproxy rather than directly from the host. 

For example if a peer has the following sip.conf:

[mytrunk]
host=sip.myitsp.net
port=5060
outboundproxy=nat.myitsp.net:5082

an invite from nat.myitsp.net:5082 is not matched to peer mytrunk

This is a problem for applications in which a natproxy is used for both
incoming and outgoing calls. 
====================================================================== 

---------------------------------------------------------------------- 
 (0097866) Nick_Lewis (reporter) - 2009-01-15 03:28
 http://bugs.digium.com/view.php?id=14230#c97866 
---------------------------------------------------------------------- 
Users are authenticated by the From: header which is easy for anyone
anywhere to forge in a call. Peers are authenticated by the actual IP
source of the call. This ensures that the call is really coming from your
ITSP.

For bog standard voip lines it needs to be as simple to setup an asterisk
PBX as it is a sip phone. In most cases the following sip.conf does the job
to provide a two way voip line:

register=myaccount at sip.myitsp.com/line1
[line1]
type=peer
username=myaccount
host=sip.myitsp.com

Some ITSP use a separate unit to provide nat traversal. These ITSP state
that the address of this unit should be added to the client config as the
outbound proxy for this voip line. For all sip phones this works fine. The
equivalent on asterisk is:

register=myaccount at sip.myitsp.com@nat.myitsp.net/line1
[line1]
type=peer
username=myaccount
host=sip.myitsp.com
outboundproxy=nat.myitsp.net

Hence why I regard there to be a bug on asterisk.


The consumer will also feel there is a bug when he tries to upgrade the
sip phone on his voip line to an asterisk pbx and finds that it sometimes
fails because of the different behaviour of asterisk to all other UAC
devices that are connected to ITSP voip services.

For example a two line sip phone may have one voip line from an ITSP that
has integrated nat traversal and another that uses a separate unit for nat
traversal. The consumer should be able to upgrade from a sip phone to
asterisk configured thus:

register=myitsp1account at sip.myitsp1.com/line1
register=myitsp2account at sip.myitsp2.com@nat.myitsp2.net/line2
[line1]
type=peer
username=myitsp1account
host=sip.myitsp1.com
[line2]
type=peer
username=myitsp2account
host=sip.myitsp2.com
outboundproxy=nat.myitsp2.net 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-15 03:28 Nick_Lewis     Note Added: 0097866                          
======================================================================