[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:                     feedback
Target Version:             1.6.1.14
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:              2010-01-15 06:06 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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---------------------------------------------------------------------- 
 (0116714) Nick_Lewis (reporter) - 2010-01-15 06:06
 https://issues.asterisk.org/view.php?id=16299#c116714 
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Once there is a working get_calleridname function it will also need to be
used by the read_to_parts function which currently tries to get the caller
id name itself - sometimes unsuccessfully 

Issue History 
Date Modified    Username       Field                    Change               
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2010-01-15 06:06 Nick_Lewis     Note Added: 0116714                          
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