[asterisk-dev] [Code Review] Truncate host:port when checking SIP domains and make the URI parsing API more explicit about returning host:port and not a domain
Terry Wilson
reviewboard at asterisk.org
Wed Nov 9 00:04:44 CST 2011
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1574/
-----------------------------------------------------------
(Updated Nov. 9, 2011, 12:04 a.m.)
Review request for Asterisk Developers and wdoekes.
Changes
-------
Address Simon's comment about IPv6 breakage by using the ast_sockaddr_split_hostport function. I wrap it with the truncate_hostport function just so we don't have to declare a throwaway variable everywhere we use it.
Summary
-------
In reviewing https://reviewboard.asterisk.org/r/1533/ I noticed that going from 1.4 to 1.8, setting domain=x.x.x.x stopped matching when receiving a request to x.x.x.x. The reason is that the parse_uri function returns host:port in what was labeled the "domain" variable in 1.8 when in 1.4 the parsing was done in each individual function and the port truncated manually. For example, peer->fromdomain could be set to a host:port, even though it is specifically split into peer->fromdomain and peer->fromport other places in the code.
We can't change the parsing function because sometimes we actually use the host:port form of the result (as in when passing it to ast_sockaddr_resolve_first), so we need to do the truncation manually when appropriate. This patch does two things: 1) it renames the "domain" variables in the API functions to "hostport" to more accurately reflect what we are really returning and 2) implements a truncate_hostport function and uses it where appropriate.
Diffs (updated)
-----
/branches/1.8/channels/chan_sip.c 343673
/branches/1.8/channels/sip/include/reqresp_parser.h 343673
/branches/1.8/channels/sip/include/sip.h 343673
/branches/1.8/channels/sip/reqresp_parser.c 343673
Diff: https://reviewboard.asterisk.org/r/1574/diff
Testing
-------
Ran the SIP unit tests and SIP-related testsuite tests and saw no difference in behavior. Set domain=127.0.0.1 and verified register requests started working.
Thanks,
Terry
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20111109/d4185dad/attachment.htm>
More information about the asterisk-dev
mailing list