[asterisk-bugs] [Asterisk 0016299]: [patch] pedantic sip checking needed to generate valid messages (but broken)

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16299 
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Reported By:                wdoekes
Assigned To:                dvossel
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Project:                    Asterisk
Issue ID:                   16299
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   trivial
Priority:                   normal
Status:                     acknowledged
Target Version:             1.6.1.13
Asterisk Version:           SVN 
JIRA:                       SWP-451 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-11-21 15:03 CST
Last Modified:              2009-12-22 03:27 CST
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Summary:                    [patch] pedantic sip checking needed to generate
valid messages (but broken)
Description: 
In function 'initreqprep' in channels/chan_sip.c, the following code can be
found:

if (sip_cfg.pedanticsipchecking) {
  ast_uri_encode(n, tmp_n, sizeof(tmp_n), 0);
  n = tmp_n;
  ast_uri_encode(l, tmp_l, sizeof(tmp_l), 0);
  l = tmp_l;
}
<...snip...>
snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s>;tag=%s", n, l, d,
p->tag);


The function ast_uri_encode encodes chars < 32 and > 127 -- perhaps one
should replace that with ((signed char)*ptr < 32) ;-) -- as %HH hex
escapes.

A couple of problems (all minor):
- ast_uri_encode forgets to escape % and 0x7F (RFC2396 2.4.2 and 2.4.3)
- ast_uri_encode does not escape <, >, @ and some other characters that
'l' would've liked to be escaped
- 'n' is not supposed to be hex-escaped (RFC4475 3.1.1.5 writes """The
display name portion of the To and From header fields is "%Z%45". Note that
this is not the same as %ZE.""")
- 'n' does however like the double-quote to be escaped, by a backslash
- ast_uri_decode is called on entire messages, not on already broken up
parts


Browsing through chan_sip.c, I see pedanticsipchecking used in these
cases:
- allow blanks between the header key and the colon
- allow multiline sip headers
- compare the from-tag/to-tag/branches as well instead of only the
call-id
- check that a packet really is for us (handle_incoming)
- encode/decode reserved characters

In my humble opinion, I don't think creating valid output (correctly
encoding illegal characters) should be enabled only by a flag that is
reported as being 'slow'. And, not as relevant to me in this case, but
decoding valid hex-escapes from peers does not sound like too much to ask,
either.


What to do?
- I can easily write a patch that fixes my minor issue: always -- not
dependent on the pedanticsipchecking -- run a s/"/\\"/g (instead of
ast_uri_encode) on the name part in the From.
- I can also easily fix ast_uri_encode to escape %, 0x7f and the others as
mentioned in RFC2396 2.4.3.
- Fixing all ast_uri_decode to operate first after the data has been
broken up is a bit more tedious, so I can't promise I'll do that.


Regards,
Walter Doekes
OSSO B.V.
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---------------------------------------------------------------------- 
 (0115632) oej (manager) - 2009-12-22 03:27
 https://issues.asterisk.org/view.php?id=16299#c115632 
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Nick, IDNA doesn't change the actual character set in the domain name used
in protocols. It's for end-user applications. So no, International
character set in domains doesn't change anything in the SIP protocol. It
changes your requirements on end-user clients though.

We have to make a decision in regards to the character set for Caller ID
names internally in Asterisk. Right now, there's no definition and using
caller ID names in UTF8 will cause issues. We do have to respect the
multiprotocol architecture of asterisk. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-12-22 03:27 oej            Note Added: 0115632                          
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