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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14230 
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Reported By:                Nick_Lewis
Assigned To:                putnopvut
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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:              
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Date Submitted:             2009-01-13 10:19 CST
Last Modified:              2009-01-15 05:21 CST
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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. 
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 (0097872) Nick_Lewis (reporter) - 2009-01-15 05:21
 http://bugs.digium.com/view.php?id=14230#c97872 
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Regarding authentication - ITSP voip lines do not provide this facility.
They expect the UAC to authenticate when registering with them or making
outgoing calls to them but they cannot handle any authentication challenge
when delivering incoming calls. The best that can be done with ITSP voip
lines is checking that the call came from the same IP address that was
authenticated during registration.

Regarding an incoming peer for each outgoing peer's outboundproxy - What
is meant by separate on the To: header? It seems to me that this would
require an entry in sip.conf something like:

register=myaccount at sip.myitsp.com@nat.myitsp.net/line1
[line1]
type=peer
host=nat.myitsp.net
[line1_out]
type=peer
username=myaccount
host=sip.myitsp.com
outboundproxy=nat.myitsp.net

which is clumsy and there would also need to be extra dialplan stuff to
associate line1 and line1_out. I dont think this complexity of config is
something that a typical ITSP sip phone user would tolerate. It was a great
step forwards when peers became bidirectional and this benefit needs to be
maintained. 

Regarding DNS resolution - if I have understood your proposal correctly
then it suffers the same potential problem. When the peer->host for
incoming calls resolves nat.myitsp.net it could theoretically be a
different IP from when reg->host resolves nat.myitsp.net. This is no
different from the bidirectional peer case when peer->outboundproxy
resolves nat.myitsp.net. In practice though this seems not to be a problem
because ITSPs somehow get whole load-balanced natproxy clusters to project
a single IP address 

Issue History 
Date Modified    Username       Field                    Change               
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2009-01-15 05:21 Nick_Lewis     Note Added: 0097872                          
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