[asterisk-dev] [Code Review] Fix RURI generation in chan_sip so that URIs are correct when SRV records are involved or when port 5060 is specified
Matthew Nicholson
mnicholson at digium.com
Tue Sep 22 13:49:47 CDT 2009
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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/369/
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(Updated 2009-09-22 13:49:47.100791)
Review request for Asterisk Developers and Olle E Johansson.
Changes
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Added a note about the modified behavior in the sip configuration file.
Summary
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This patch ensures that RURIs generated by asterisk are correct when a port is specified and in the presence of an SRV lookup. Without this patch, asterisk will include a port in an RUI when port != 5060 and will not include a port if port == 5060. This behavior is not correct. According to RFC 3261, the URIs "host.tld" and "host.tld:5060" are not the same. Additionally the URI "host.tld" indicates that an SRV lookup should be performed (which asterisk handles properly) BUT the SRV lookup should NOT change the RURI.
Note: This patch does not address asterisk generation REGISTER requests, which may also have RURI problems.
This addresses bug 14418.
https://issues.asterisk.org/view.php?id=14418
Diffs (updated)
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/branches/1.4/channels/chan_sip.c 219791
/branches/1.4/configs/sip.conf.sample 219791
Diff: https://reviewboard.asterisk.org/r/369/diff
Testing
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I tested the following cases where RURIs are generated. In all cases the proper behavior was observed.
* port specified in dialplan as follows: Dial(SIP/user at host:port). Testing was done with port = 5060, 5061 and with an SRV lookup that altered the host and port.
* port provided by SIP client during registration
* port provided in sip.conf peer definition. In this case the RURI is generated properly, but SRV lookups are done regardless if a port is specified in the peer definition or not. This is not correct (SRV lookups should only occur in the absence of a port specification). I consider that an unrelated issue.
* port specified in dialplan and specified in sip.conf or during peer registration: Dial(SIP/123 at peer:1234). This works properly with the same caveats as the previous case.
Thanks,
Matthew
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